prymula/com: DPF-Prymula-audioplugins/dpf/distrho/src/jackbridge/rtaudio/RtAudio.cpp comparison

comparison DPF-Prymula-audioplugins/dpf/distrho/src/jackbridge/rtaudio/RtAudio.cpp @ 3:84e66ea83026

DPF-Prymula-audioplugins-0.231015-2

author	prymula <prymula76@outlook.com>
date	Mon, 16 Oct 2023 21:53:34 +0200
parents
children

comparison

equal deleted inserted replaced

-:cf2cb71d31dd
+:84e66ea83026
+/************************************************************************/
+/*! \class RtAudio
+\brief Realtime audio i/o C++ classes.
+RtAudio provides a common API (Application Programming Interface)
+for realtime audio input/output across Linux (native ALSA, Jack,
+and OSS), Macintosh OS X (CoreAudio and Jack), and Windows
+(DirectSound, ASIO and WASAPI) operating systems.
+RtAudio GitHub site: https://github.com/thestk/rtaudio
+RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/
+RtAudio: realtime audio i/o C++ classes
+Copyright (c) 2001-2019 Gary P. Scavone
+Permission is hereby granted, free of charge, to any person
+obtaining a copy of this software and associated documentation files
+(the "Software"), to deal in the Software without restriction,
+including without limitation the rights to use, copy, modify, merge,
+publish, distribute, sublicense, and/or sell copies of the Software,
+and to permit persons to whom the Software is furnished to do so,
+subject to the following conditions:
+The above copyright notice and this permission notice shall be
+included in all copies or substantial portions of the Software.
+Any person wishing to distribute modifications to the Software is
+asked to send the modifications to the original developer so that
+they can be incorporated into the canonical version.  This is,
+however, not a binding provision of this license.
+THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
+EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
+MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
+IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
+ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
+CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
+WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
+*/
+/************************************************************************/
+// RtAudio: Version 5.1.0
+#include "RtAudio.h"
+#include <iostream>
+#include <cstdlib>
+#include <cstring>
+#include <climits>
+#include <cmath>
+#include <algorithm>
+// Static variable definitions.
+const unsigned int RtApi::MAX_SAMPLE_RATES = 14;
+const unsigned int RtApi::SAMPLE_RATES[] = {
+4000, 5512, 8000, 9600, 11025, 16000, 22050,
+32000, 44100, 48000, 88200, 96000, 176400, 192000
+};
+#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) || defined(__WINDOWS_WASAPI__)
+#define MUTEX_INITIALIZE(A) InitializeCriticalSection(A)
+#define MUTEX_DESTROY(A)    DeleteCriticalSection(A)
+#define MUTEX_LOCK(A)       EnterCriticalSection(A)
+#define MUTEX_UNLOCK(A)     LeaveCriticalSection(A)
+#include "tchar.h"
+template<typename T> inline
+std::string convertCharPointerToStdString(const T *text);
+template<> inline
+std::string convertCharPointerToStdString(const char *text)
+{
+return std::string(text);
+}
+template<> inline
+std::string convertCharPointerToStdString(const wchar_t *text)
+{
+int length = WideCharToMultiByte(CP_UTF8, 0, text, -1, NULL, 0, NULL, NULL);
+std::string s( length-1, '\0' );
+WideCharToMultiByte(CP_UTF8, 0, text, -1, &s[0], length, NULL, NULL);
+return s;
+}
+#elif defined(__LINUX_ALSA__) || defined(__LINUX_PULSE__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__)
+// pthread API
+#define MUTEX_INITIALIZE(A) pthread_mutex_init(A, NULL)
+#define MUTEX_DESTROY(A)    pthread_mutex_destroy(A)
+#define MUTEX_LOCK(A)       pthread_mutex_lock(A)
+#define MUTEX_UNLOCK(A)     pthread_mutex_unlock(A)
+#else
+#define MUTEX_INITIALIZE(A) abs(*A) // dummy definitions
+#define MUTEX_DESTROY(A)    abs(*A) // dummy definitions
+#endif
+// *************************************************** //
+//
+// RtAudio definitions.
+//
+// *************************************************** //
+std::string RtAudio :: getVersion( void )
+{
+return RTAUDIO_VERSION;
+}
+// Define API names and display names.
+// Must be in same order as API enum.
+extern "C" {
+const char* rtaudio_api_names[][2] = {
+{ "unspecified" , "Unknown" },
+{ "alsa"        , "ALSA" },
+{ "pulse"       , "Pulse" },
+{ "oss"         , "OpenSoundSystem" },
+{ "jack"        , "Jack" },
+{ "core"        , "CoreAudio" },
+{ "wasapi"      , "WASAPI" },
+{ "asio"        , "ASIO" },
+{ "ds"          , "DirectSound" },
+{ "dummy"       , "Dummy" },
+};
+const unsigned int rtaudio_num_api_names =
+sizeof(rtaudio_api_names)/sizeof(rtaudio_api_names[0]);
+// The order here will control the order of RtAudio's API search in
+// the constructor.
+extern "C" const RtAudio::Api rtaudio_compiled_apis[] = {
+#if defined(__UNIX_JACK__)
+RtAudio::UNIX_JACK,
+#endif
+#if defined(__LINUX_PULSE__)
+RtAudio::LINUX_PULSE,
+#endif
+#if defined(__LINUX_ALSA__)
+RtAudio::LINUX_ALSA,
+#endif
+#if defined(__LINUX_OSS__)
+RtAudio::LINUX_OSS,
+#endif
+#if defined(__WINDOWS_ASIO__)
+RtAudio::WINDOWS_ASIO,
+#endif
+#if defined(__WINDOWS_WASAPI__)
+RtAudio::WINDOWS_WASAPI,
+#endif
+#if defined(__WINDOWS_DS__)
+RtAudio::WINDOWS_DS,
+#endif
+#if defined(__MACOSX_CORE__)
+RtAudio::MACOSX_CORE,
+#endif
+#if defined(__RTAUDIO_DUMMY__)
+RtAudio::RTAUDIO_DUMMY,
+#endif
+RtAudio::UNSPECIFIED,
+};
+extern "C" const unsigned int rtaudio_num_compiled_apis =
+sizeof(rtaudio_compiled_apis)/sizeof(rtaudio_compiled_apis[0])-1;
+}
+// This is a compile-time check that rtaudio_num_api_names == RtAudio::NUM_APIS.
+// If the build breaks here, check that they match.
+template<bool b> class StaticAssert { private: StaticAssert() {} };
+template<> class StaticAssert<true>{ public: StaticAssert() {} };
+class StaticAssertions { StaticAssertions() {
+StaticAssert<rtaudio_num_api_names == RtAudio::NUM_APIS>();
+}};
+void RtAudio :: getCompiledApi( std::vector<RtAudio::Api> &apis )
+{
+apis = std::vector<RtAudio::Api>(rtaudio_compiled_apis,
+rtaudio_compiled_apis + rtaudio_num_compiled_apis);
+}
+std::string RtAudio :: getApiName( RtAudio::Api api )
+{
+if (api < 0 || api >= RtAudio::NUM_APIS)
+return "";
+return rtaudio_api_names[api][0];
+}
+std::string RtAudio :: getApiDisplayName( RtAudio::Api api )
+{
+if (api < 0 || api >= RtAudio::NUM_APIS)
+return "Unknown";
+return rtaudio_api_names[api][1];
+}
+RtAudio::Api RtAudio :: getCompiledApiByName( const std::string &name )
+{
+unsigned int i=0;
+for (i = 0; i < rtaudio_num_compiled_apis; ++i)
+if (name == rtaudio_api_names[rtaudio_compiled_apis[i]][0])
+return rtaudio_compiled_apis[i];
+return RtAudio::UNSPECIFIED;
+}
+void RtAudio :: openRtApi( RtAudio::Api api )
+{
+if ( rtapi_ )
+delete rtapi_;
+rtapi_ = 0;
+#if defined(__UNIX_JACK__)
+if ( api == UNIX_JACK )
+rtapi_ = new RtApiJack();
+#endif
+#if defined(__LINUX_ALSA__)
+if ( api == LINUX_ALSA )
+rtapi_ = new RtApiAlsa();
+#endif
+#if defined(__LINUX_PULSE__)
+if ( api == LINUX_PULSE )
+rtapi_ = new RtApiPulse();
+#endif
+#if defined(__LINUX_OSS__)
+if ( api == LINUX_OSS )
+rtapi_ = new RtApiOss();
+#endif
+#if defined(__WINDOWS_ASIO__)
+if ( api == WINDOWS_ASIO )
+rtapi_ = new RtApiAsio();
+#endif
+#if defined(__WINDOWS_WASAPI__)
+if ( api == WINDOWS_WASAPI )
+rtapi_ = new RtApiWasapi();
+#endif
+#if defined(__WINDOWS_DS__)
+if ( api == WINDOWS_DS )
+rtapi_ = new RtApiDs();
+#endif
+#if defined(__MACOSX_CORE__)
+if ( api == MACOSX_CORE )
+rtapi_ = new RtApiCore();
+#endif
+#if defined(__RTAUDIO_DUMMY__)
+if ( api == RTAUDIO_DUMMY )
+rtapi_ = new RtApiDummy();
+#endif
+}
+RtAudio :: RtAudio( RtAudio::Api api )
+{
+rtapi_ = 0;
+if ( api != UNSPECIFIED ) {
+// Attempt to open the specified API.
+openRtApi( api );
+if ( rtapi_ ) return;
+// No compiled support for specified API value.  Issue a debug
+// warning and continue as if no API was specified.
+std::cerr << "\nRtAudio: no compiled support for specified API argument!\n" << std::endl;
+}
+// Iterate through the compiled APIs and return as soon as we find
+// one with at least one device or we reach the end of the list.
+std::vector< RtAudio::Api > apis;
+getCompiledApi( apis );
+for ( unsigned int i=0; i<apis.size(); i++ ) {
+openRtApi( apis[i] );
+if ( rtapi_ && rtapi_->getDeviceCount() ) break;
+}
+if ( rtapi_ ) return;
+// It should not be possible to get here because the preprocessor
+// definition __RTAUDIO_DUMMY__ is automatically defined if no
+// API-specific definitions are passed to the compiler. But just in
+// case something weird happens, we'll thow an error.
+std::string errorText = "\nRtAudio: no compiled API support found ... critical error!!\n\n";
+throw( RtAudioError( errorText, RtAudioError::UNSPECIFIED ) );
+}
+RtAudio :: ~RtAudio()
+{
+if ( rtapi_ )
+delete rtapi_;
+}
+void RtAudio :: openStream( RtAudio::StreamParameters *outputParameters,
+RtAudio::StreamParameters *inputParameters,
+RtAudioFormat format, unsigned int sampleRate,
+unsigned int *bufferFrames,
+RtAudioCallback callback, void *userData,
+RtAudio::StreamOptions *options,
+RtAudioErrorCallback errorCallback )
+{
+return rtapi_->openStream( outputParameters, inputParameters, format,
+sampleRate, bufferFrames, callback,
+userData, options, errorCallback );
+}
+// *************************************************** //
+//
+// Public RtApi definitions (see end of file for
+// private or protected utility functions).
+//
+// *************************************************** //
+RtApi :: RtApi()
+{
+stream_.state = STREAM_CLOSED;
+stream_.mode = UNINITIALIZED;
+stream_.apiHandle = 0;
+stream_.userBuffer[0] = 0;
+stream_.userBuffer[1] = 0;
+MUTEX_INITIALIZE( &stream_.mutex );
+showWarnings_ = true;
+firstErrorOccurred_ = false;
+}
+RtApi :: ~RtApi()
+{
+MUTEX_DESTROY( &stream_.mutex );
+}
+void RtApi :: openStream( RtAudio::StreamParameters *oParams,
+RtAudio::StreamParameters *iParams,
+RtAudioFormat format, unsigned int sampleRate,
+unsigned int *bufferFrames,
+RtAudioCallback callback, void *userData,
+RtAudio::StreamOptions *options,
+RtAudioErrorCallback errorCallback )
+{
+if ( stream_.state != STREAM_CLOSED ) {
+errorText_ = "RtApi::openStream: a stream is already open!";
+error( RtAudioError::INVALID_USE );
+return;
+}
+// Clear stream information potentially left from a previously open stream.
+clearStreamInfo();
+if ( oParams && oParams->nChannels < 1 ) {
+errorText_ = "RtApi::openStream: a non-NULL output StreamParameters structure cannot have an nChannels value less than one.";
+error( RtAudioError::INVALID_USE );
+return;
+}
+if ( iParams && iParams->nChannels < 1 ) {
+errorText_ = "RtApi::openStream: a non-NULL input StreamParameters structure cannot have an nChannels value less than one.";
+error( RtAudioError::INVALID_USE );
+return;
+}
+if ( oParams == NULL && iParams == NULL ) {
+errorText_ = "RtApi::openStream: input and output StreamParameters structures are both NULL!";
+error( RtAudioError::INVALID_USE );
+return;
+}
+if ( formatBytes(format) == 0 ) {
+errorText_ = "RtApi::openStream: 'format' parameter value is undefined.";
+error( RtAudioError::INVALID_USE );
+return;
+}
+unsigned int nDevices = getDeviceCount();
+unsigned int oChannels = 0;
+if ( oParams ) {
+oChannels = oParams->nChannels;
+if ( oParams->deviceId >= nDevices ) {
+errorText_ = "RtApi::openStream: output device parameter value is invalid.";
+error( RtAudioError::INVALID_USE );
+return;
+}
+}
+unsigned int iChannels = 0;
+if ( iParams ) {
+iChannels = iParams->nChannels;
+if ( iParams->deviceId >= nDevices ) {
+errorText_ = "RtApi::openStream: input device parameter value is invalid.";
+error( RtAudioError::INVALID_USE );
+return;
+}
+}
+bool result;
+if ( oChannels > 0 ) {
+result = probeDeviceOpen( oParams->deviceId, OUTPUT, oChannels, oParams->firstChannel,
+sampleRate, format, bufferFrames, options );
+if ( result == false ) {
+error( RtAudioError::SYSTEM_ERROR );
+return;
+}
+}
+if ( iChannels > 0 ) {
+result = probeDeviceOpen( iParams->deviceId, INPUT, iChannels, iParams->firstChannel,
+sampleRate, format, bufferFrames, options );
+if ( result == false ) {
+if ( oChannels > 0 ) closeStream();
+error( RtAudioError::SYSTEM_ERROR );
+return;
+}
+}
+stream_.callbackInfo.callback = (void *) callback;
+stream_.callbackInfo.userData = userData;
+stream_.callbackInfo.errorCallback = (void *) errorCallback;
+if ( options ) options->numberOfBuffers = stream_.nBuffers;
+stream_.state = STREAM_STOPPED;
+}
+unsigned int RtApi :: getDefaultInputDevice( void )
+{
+// Should be implemented in subclasses if possible.
+return 0;
+}
+unsigned int RtApi :: getDefaultOutputDevice( void )
+{
+// Should be implemented in subclasses if possible.
+return 0;
+}
+void RtApi :: closeStream( void )
+{
+// MUST be implemented in subclasses!
+return;
+}
+bool RtApi :: probeDeviceOpen( unsigned int /*device*/, StreamMode /*mode*/, unsigned int /*channels*/,
+unsigned int /*firstChannel*/, unsigned int /*sampleRate*/,
+RtAudioFormat /*format*/, unsigned int * /*bufferSize*/,
+RtAudio::StreamOptions * /*options*/ )
+{
+// MUST be implemented in subclasses!
+return FAILURE;
+}
+void RtApi :: tickStreamTime( void )
+{
+// Subclasses that do not provide their own implementation of
+// getStreamTime should call this function once per buffer I/O to
+// provide basic stream time support.
+stream_.streamTime += ( stream_.bufferSize * 1.0 / stream_.sampleRate );
+#if defined( HAVE_GETTIMEOFDAY )
+gettimeofday( &stream_.lastTickTimestamp, NULL );
+#endif
+}
+long RtApi :: getStreamLatency( void )
+{
+verifyStream();
+long totalLatency = 0;
+if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
+totalLatency = stream_.latency[0];
+if ( stream_.mode == INPUT || stream_.mode == DUPLEX )
+totalLatency += stream_.latency[1];
+return totalLatency;
+}
+double RtApi :: getStreamTime( void )
+{
+verifyStream();
+#if defined( HAVE_GETTIMEOFDAY )
+// Return a very accurate estimate of the stream time by
+// adding in the elapsed time since the last tick.
+struct timeval then;
+struct timeval now;
+if ( stream_.state != STREAM_RUNNING || stream_.streamTime == 0.0 )
+return stream_.streamTime;
+gettimeofday( &now, NULL );
+then = stream_.lastTickTimestamp;
+return stream_.streamTime +
+((now.tv_sec + 0.000001 * now.tv_usec) -
+(then.tv_sec + 0.000001 * then.tv_usec));
+#else
+return stream_.streamTime;
+#endif
+}
+void RtApi :: setStreamTime( double time )
+{
+verifyStream();
+if ( time >= 0.0 )
+stream_.streamTime = time;
+#if defined( HAVE_GETTIMEOFDAY )
+gettimeofday( &stream_.lastTickTimestamp, NULL );
+#endif
+}
+unsigned int RtApi :: getStreamSampleRate( void )
+{
+verifyStream();
+return stream_.sampleRate;
+}
+// *************************************************** //
+//
+// OS/API-specific methods.
+//
+// *************************************************** //
+#if defined(__MACOSX_CORE__)
+#include <unistd.h>
+// The OS X CoreAudio API is designed to use a separate callback
+// procedure for each of its audio devices.  A single RtAudio duplex
+// stream using two different devices is supported here, though it
+// cannot be guaranteed to always behave correctly because we cannot
+// synchronize these two callbacks.
+//
+// A property listener is installed for over/underrun information.
+// However, no functionality is currently provided to allow property
+// listeners to trigger user handlers because it is unclear what could
+// be done if a critical stream parameter (buffer size, sample rate,
+// device disconnect) notification arrived.  The listeners entail
+// quite a bit of extra code and most likely, a user program wouldn't
+// be prepared for the result anyway.  However, we do provide a flag
+// to the client callback function to inform of an over/underrun.
+// A structure to hold various information related to the CoreAudio API
+// implementation.
+struct CoreHandle {
+AudioDeviceID id[2];    // device ids
+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
+AudioDeviceIOProcID procId[2];
+#endif
+UInt32 iStream[2];      // device stream index (or first if using multiple)
+UInt32 nStreams[2];     // number of streams to use
+bool xrun[2];
+char *deviceBuffer;
+pthread_cond_t condition;
+int drainCounter;       // Tracks callback counts when draining
+bool internalDrain;     // Indicates if stop is initiated from callback or not.
+CoreHandle()
+:deviceBuffer(0), drainCounter(0), internalDrain(false) { nStreams[0] = 1; nStreams[1] = 1; id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
+};
+RtApiCore:: RtApiCore()
+{
+#if defined( AVAILABLE_MAC_OS_X_VERSION_10_6_AND_LATER )
+// This is a largely undocumented but absolutely necessary
+// requirement starting with OS-X 10.6.  If not called, queries and
+// updates to various audio device properties are not handled
+// correctly.
+CFRunLoopRef theRunLoop = NULL;
+AudioObjectPropertyAddress property = { kAudioHardwarePropertyRunLoop,
+kAudioObjectPropertyScopeGlobal,
+kAudioObjectPropertyElementMaster };
+OSStatus result = AudioObjectSetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, sizeof(CFRunLoopRef), &theRunLoop);
+if ( result != noErr ) {
+errorText_ = "RtApiCore::RtApiCore: error setting run loop property!";
+error( RtAudioError::WARNING );
+}
+#endif
+}
+RtApiCore :: ~RtApiCore()
+{
+// The subclass destructor gets called before the base class
+// destructor, so close an existing stream before deallocating
+// apiDeviceId memory.
+if ( stream_.state != STREAM_CLOSED ) closeStream();
+}
+unsigned int RtApiCore :: getDeviceCount( void )
+{
+// Find out how many audio devices there are, if any.
+UInt32 dataSize;
+AudioObjectPropertyAddress propertyAddress = { kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
+OSStatus result = AudioObjectGetPropertyDataSize( kAudioObjectSystemObject, &propertyAddress, 0, NULL, &dataSize );
+if ( result != noErr ) {
+errorText_ = "RtApiCore::getDeviceCount: OS-X error getting device info!";
+error( RtAudioError::WARNING );
+return 0;
+}
+return dataSize / sizeof( AudioDeviceID );
+}
+unsigned int RtApiCore :: getDefaultInputDevice( void )
+{
+unsigned int nDevices = getDeviceCount();
+if ( nDevices <= 1 ) return 0;
+AudioDeviceID id;
+UInt32 dataSize = sizeof( AudioDeviceID );
+AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultInputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
+OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id );
+if ( result != noErr ) {
+errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device.";
+error( RtAudioError::WARNING );
+return 0;
+}
+dataSize *= nDevices;
+AudioDeviceID deviceList[ nDevices ];
+property.mSelector = kAudioHardwarePropertyDevices;
+result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList );
+if ( result != noErr ) {
+errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device IDs.";
+error( RtAudioError::WARNING );
+return 0;
+}
+for ( unsigned int i=0; i<nDevices; i++ )
+if ( id == deviceList[i] ) return i;
+errorText_ = "RtApiCore::getDefaultInputDevice: No default device found!";
+error( RtAudioError::WARNING );
+return 0;
+}
+unsigned int RtApiCore :: getDefaultOutputDevice( void )
+{
+unsigned int nDevices = getDeviceCount();
+if ( nDevices <= 1 ) return 0;
+AudioDeviceID id;
+UInt32 dataSize = sizeof( AudioDeviceID );
+AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultOutputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
+OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id );
+if ( result != noErr ) {
+errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device.";
+error( RtAudioError::WARNING );
+return 0;
+}
+dataSize = sizeof( AudioDeviceID ) * nDevices;
+AudioDeviceID deviceList[ nDevices ];
+property.mSelector = kAudioHardwarePropertyDevices;
+result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList );
+if ( result != noErr ) {
+errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device IDs.";
+error( RtAudioError::WARNING );
+return 0;
+}
+for ( unsigned int i=0; i<nDevices; i++ )
+if ( id == deviceList[i] ) return i;
+errorText_ = "RtApiCore::getDefaultOutputDevice: No default device found!";
+error( RtAudioError::WARNING );
+return 0;
+}
+RtAudio::DeviceInfo RtApiCore :: getDeviceInfo( unsigned int device )
+{
+RtAudio::DeviceInfo info;
+info.probed = false;
+// Get device ID
+unsigned int nDevices = getDeviceCount();
+if ( nDevices == 0 ) {
+errorText_ = "RtApiCore::getDeviceInfo: no devices found!";
+error( RtAudioError::INVALID_USE );
+return info;
+}
+if ( device >= nDevices ) {
+errorText_ = "RtApiCore::getDeviceInfo: device ID is invalid!";
+error( RtAudioError::INVALID_USE );
+return info;
+}
+AudioDeviceID deviceList[ nDevices ];
+UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;
+AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
+kAudioObjectPropertyScopeGlobal,
+kAudioObjectPropertyElementMaster };
+OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property,
+0, NULL, &dataSize, (void *) &deviceList );
+if ( result != noErr ) {
+errorText_ = "RtApiCore::getDeviceInfo: OS-X system error getting device IDs.";
+error( RtAudioError::WARNING );
+return info;
+}
+AudioDeviceID id = deviceList[ device ];
+// Get the device name.
+info.name.erase();
+CFStringRef cfname;
+dataSize = sizeof( CFStringRef );
+property.mSelector = kAudioObjectPropertyManufacturer;
+result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname );
+if ( result != noErr ) {
+errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device manufacturer.";
+errorText_ = errorStream_.str();
+error( RtAudioError::WARNING );
+return info;
+}
+//const char *mname = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() );
+int length = CFStringGetLength(cfname);
+char *mname = (char *)malloc(length * 3 + 1);
+#if defined( UNICODE ) || defined( _UNICODE )
+CFStringGetCString(cfname, mname, length * 3 + 1, kCFStringEncodingUTF8);
+#else
+CFStringGetCString(cfname, mname, length * 3 + 1, CFStringGetSystemEncoding());
+#endif
+info.name.append( (const char *)mname, strlen(mname) );
+info.name.append( ": " );
+CFRelease( cfname );
+free(mname);
+property.mSelector = kAudioObjectPropertyName;
+result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname );
+if ( result != noErr ) {
+errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device name.";
+errorText_ = errorStream_.str();
+error( RtAudioError::WARNING );
+return info;
+}
+//const char *name = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() );
+length = CFStringGetLength(cfname);
+char *name = (char *)malloc(length * 3 + 1);
+#if defined( UNICODE ) || defined( _UNICODE )
+CFStringGetCString(cfname, name, length * 3 + 1, kCFStringEncodingUTF8);
+#else
+CFStringGetCString(cfname, name, length * 3 + 1, CFStringGetSystemEncoding());
+#endif
+info.name.append( (const char *)name, strlen(name) );
+CFRelease( cfname );
+free(name);
+// Get the output stream "configuration".
+AudioBufferList	*bufferList = nil;
+property.mSelector = kAudioDevicePropertyStreamConfiguration;
+property.mScope = kAudioDevicePropertyScopeOutput;
+//  property.mElement = kAudioObjectPropertyElementWildcard;
+dataSize = 0;
+result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
+if ( result != noErr || dataSize == 0 ) {
+errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration info for device (" << device << ").";
+errorText_ = errorStream_.str();
+error( RtAudioError::WARNING );
+return info;
+}
+// Allocate the AudioBufferList.
+bufferList = (AudioBufferList *) malloc( dataSize );
+if ( bufferList == NULL ) {
+errorText_ = "RtApiCore::getDeviceInfo: memory error allocating output AudioBufferList.";
+error( RtAudioError::WARNING );
+return info;
+}
+result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
+if ( result != noErr || dataSize == 0 ) {
+free( bufferList );
+errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration for device (" << device << ").";
+errorText_ = errorStream_.str();
+error( RtAudioError::WARNING );
+return info;
+}
+// Get output channel information.
+unsigned int i, nStreams = bufferList->mNumberBuffers;
+for ( i=0; i<nStreams; i++ )
+info.outputChannels += bufferList->mBuffers[i].mNumberChannels;
+free( bufferList );
+// Get the input stream "configuration".
+property.mScope = kAudioDevicePropertyScopeInput;
+result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
+if ( result != noErr || dataSize == 0 ) {
+errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration info for device (" << device << ").";
+errorText_ = errorStream_.str();
+error( RtAudioError::WARNING );
+return info;
+}
+// Allocate the AudioBufferList.
+bufferList = (AudioBufferList *) malloc( dataSize );
+if ( bufferList == NULL ) {
+errorText_ = "RtApiCore::getDeviceInfo: memory error allocating input AudioBufferList.";
+error( RtAudioError::WARNING );
+return info;
+}
+result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
+if (result != noErr || dataSize == 0) {
+free( bufferList );
+errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration for device (" << device << ").";
+errorText_ = errorStream_.str();
+error( RtAudioError::WARNING );
+return info;
+}
+// Get input channel information.
+nStreams = bufferList->mNumberBuffers;
+for ( i=0; i<nStreams; i++ )
+info.inputChannels += bufferList->mBuffers[i].mNumberChannels;
+free( bufferList );
+// If device opens for both playback and capture, we determine the channels.
+if ( info.outputChannels > 0 && info.inputChannels > 0 )
+info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
+// Probe the device sample rates.
+bool isInput = false;
+if ( info.outputChannels == 0 ) isInput = true;
+// Determine the supported sample rates.
+property.mSelector = kAudioDevicePropertyAvailableNominalSampleRates;
+if ( isInput == false ) property.mScope = kAudioDevicePropertyScopeOutput;
+result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
+if ( result != kAudioHardwareNoError || dataSize == 0 ) {
+errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rate info.";
+errorText_ = errorStream_.str();
+error( RtAudioError::WARNING );
+return info;
+}
+UInt32 nRanges = dataSize / sizeof( AudioValueRange );
+AudioValueRange rangeList[ nRanges ];
+result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &rangeList );
+if ( result != kAudioHardwareNoError ) {
+errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rates.";
+errorText_ = errorStream_.str();
+error( RtAudioError::WARNING );
+return info;
+}
+// The sample rate reporting mechanism is a bit of a mystery.  It
+// seems that it can either return individual rates or a range of
+// rates.  I assume that if the min / max range values are the same,
+// then that represents a single supported rate and if the min / max
+// range values are different, the device supports an arbitrary
+// range of values (though there might be multiple ranges, so we'll
+// use the most conservative range).
+Float64 minimumRate = 1.0, maximumRate = 10000000000.0;
+bool haveValueRange = false;
+info.sampleRates.clear();
+for ( UInt32 i=0; i<nRanges; i++ ) {
+if ( rangeList[i].mMinimum == rangeList[i].mMaximum ) {
+unsigned int tmpSr = (unsigned int) rangeList[i].mMinimum;
+info.sampleRates.push_back( tmpSr );
+if ( !info.preferredSampleRate || ( tmpSr <= 48000 && tmpSr > info.preferredSampleRate ) )
+info.preferredSampleRate = tmpSr;
+} else {
+haveValueRange = true;
+if ( rangeList[i].mMinimum > minimumRate ) minimumRate = rangeList[i].mMinimum;
+if ( rangeList[i].mMaximum < maximumRate ) maximumRate = rangeList[i].mMaximum;
+}
+}
+if ( haveValueRange ) {
+for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
+if ( SAMPLE_RATES[k] >= (unsigned int) minimumRate && SAMPLE_RATES[k] <= (unsigned int) maximumRate ) {
+info.sampleRates.push_back( SAMPLE_RATES[k] );
+if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
+info.preferredSampleRate = SAMPLE_RATES[k];
+}
+}
+}
+// Sort and remove any redundant values
+std::sort( info.sampleRates.begin(), info.sampleRates.end() );
+info.sampleRates.erase( unique( info.sampleRates.begin(), info.sampleRates.end() ), info.sampleRates.end() );
+if ( info.sampleRates.size() == 0 ) {
+errorStream_ << "RtApiCore::probeDeviceInfo: No supported sample rates found for device (" << device << ").";
+errorText_ = errorStream_.str();
+error( RtAudioError::WARNING );
+return info;
+}
+// CoreAudio always uses 32-bit floating point data for PCM streams.
+// Thus, any other "physical" formats supported by the device are of
+// no interest to the client.
+info.nativeFormats = RTAUDIO_FLOAT32;
+if ( info.outputChannels > 0 )
+if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true;
+if ( info.inputChannels > 0 )
+if ( getDefaultInputDevice() == device ) info.isDefaultInput = true;
+info.probed = true;
+return info;
+}
+static OSStatus callbackHandler( AudioDeviceID inDevice,
+const AudioTimeStamp* /*inNow*/,
+const AudioBufferList* inInputData,
+const AudioTimeStamp* /*inInputTime*/,
+AudioBufferList* outOutputData,
+const AudioTimeStamp* /*inOutputTime*/,
+void* infoPointer )
+{
+CallbackInfo *info = (CallbackInfo *) infoPointer;
+RtApiCore *object = (RtApiCore *) info->object;
+if ( object->callbackEvent( inDevice, inInputData, outOutputData ) == false )
+return kAudioHardwareUnspecifiedError;
+else
+return kAudioHardwareNoError;
+}
+static OSStatus xrunListener( AudioObjectID /*inDevice*/,
+UInt32 nAddresses,
+const AudioObjectPropertyAddress properties[],
+void* handlePointer )
+{
+CoreHandle *handle = (CoreHandle *) handlePointer;
+for ( UInt32 i=0; i<nAddresses; i++ ) {
+if ( properties[i].mSelector == kAudioDeviceProcessorOverload ) {
+if ( properties[i].mScope == kAudioDevicePropertyScopeInput )
+handle->xrun[1] = true;
+else
+handle->xrun[0] = true;
+}
+}
+return kAudioHardwareNoError;
+}
+static OSStatus rateListener( AudioObjectID inDevice,
+UInt32 /*nAddresses*/,
+const AudioObjectPropertyAddress /*properties*/[],
+void* ratePointer )
+{
+Float64 *rate = (Float64 *) ratePointer;
+UInt32 dataSize = sizeof( Float64 );
+AudioObjectPropertyAddress property = { kAudioDevicePropertyNominalSampleRate,
+kAudioObjectPropertyScopeGlobal,
+kAudioObjectPropertyElementMaster };
+AudioObjectGetPropertyData( inDevice, &property, 0, NULL, &dataSize, rate );
+return kAudioHardwareNoError;
+}
+bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+unsigned int firstChannel, unsigned int sampleRate,
+RtAudioFormat format, unsigned int *bufferSize,
+RtAudio::StreamOptions *options )
+{
+// Get device ID
+unsigned int nDevices = getDeviceCount();
+if ( nDevices == 0 ) {
+// This should not happen because a check is made before this function is called.
+errorText_ = "RtApiCore::probeDeviceOpen: no devices found!";
+return FAILURE;
+}
+if ( device >= nDevices ) {
+// This should not happen because a check is made before this function is called.
+errorText_ = "RtApiCore::probeDeviceOpen: device ID is invalid!";
+return FAILURE;
+}
+AudioDeviceID deviceList[ nDevices ];
+UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;
+AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
+kAudioObjectPropertyScopeGlobal,
+kAudioObjectPropertyElementMaster };
+OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property,
+0, NULL, &dataSize, (void *) &deviceList );
+if ( result != noErr ) {
+errorText_ = "RtApiCore::probeDeviceOpen: OS-X system error getting device IDs.";
+return FAILURE;
+}
+AudioDeviceID id = deviceList[ device ];
+// Setup for stream mode.
+bool isInput = false;
+if ( mode == INPUT ) {
+isInput = true;
+property.mScope = kAudioDevicePropertyScopeInput;
+}
+else
+property.mScope = kAudioDevicePropertyScopeOutput;
+// Get the stream "configuration".
+AudioBufferList	*bufferList = nil;
+dataSize = 0;
+property.mSelector = kAudioDevicePropertyStreamConfiguration;
+result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
+if ( result != noErr || dataSize == 0 ) {
+errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration info for device (" << device << ").";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+// Allocate the AudioBufferList.
+bufferList = (AudioBufferList *) malloc( dataSize );
+if ( bufferList == NULL ) {
+errorText_ = "RtApiCore::probeDeviceOpen: memory error allocating AudioBufferList.";
+return FAILURE;
+}
+result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
+if (result != noErr || dataSize == 0) {
+free( bufferList );
+errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration for device (" << device << ").";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+// Search for one or more streams that contain the desired number of
+// channels. CoreAudio devices can have an arbitrary number of
+// streams and each stream can have an arbitrary number of channels.
+// For each stream, a single buffer of interleaved samples is
+// provided.  RtAudio prefers the use of one stream of interleaved
+// data or multiple consecutive single-channel streams.  However, we
+// now support multiple consecutive multi-channel streams of
+// interleaved data as well.
+UInt32 iStream, offsetCounter = firstChannel;
+UInt32 nStreams = bufferList->mNumberBuffers;
+bool monoMode = false;
+bool foundStream = false;
+// First check that the device supports the requested number of
+// channels.
+UInt32 deviceChannels = 0;
+for ( iStream=0; iStream<nStreams; iStream++ )
+deviceChannels += bufferList->mBuffers[iStream].mNumberChannels;
+if ( deviceChannels < ( channels + firstChannel ) ) {
+free( bufferList );
+errorStream_ << "RtApiCore::probeDeviceOpen: the device (" << device << ") does not support the requested channel count.";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+// Look for a single stream meeting our needs.
+UInt32 firstStream, streamCount = 1, streamChannels = 0, channelOffset = 0;
+for ( iStream=0; iStream<nStreams; iStream++ ) {
+streamChannels = bufferList->mBuffers[iStream].mNumberChannels;
+if ( streamChannels >= channels + offsetCounter ) {
+firstStream = iStream;
+channelOffset = offsetCounter;
+foundStream = true;
+break;
+}
+if ( streamChannels > offsetCounter ) break;
+offsetCounter -= streamChannels;
+}
+// If we didn't find a single stream above, then we should be able
+// to meet the channel specification with multiple streams.
+if ( foundStream == false ) {
+monoMode = true;
+offsetCounter = firstChannel;
+for ( iStream=0; iStream<nStreams; iStream++ ) {
+streamChannels = bufferList->mBuffers[iStream].mNumberChannels;
+if ( streamChannels > offsetCounter ) break;
+offsetCounter -= streamChannels;
+}
+firstStream = iStream;
+channelOffset = offsetCounter;
+Int32 channelCounter = channels + offsetCounter - streamChannels;
+if ( streamChannels > 1 ) monoMode = false;
+while ( channelCounter > 0 ) {
+streamChannels = bufferList->mBuffers[++iStream].mNumberChannels;
+if ( streamChannels > 1 ) monoMode = false;
+channelCounter -= streamChannels;
+streamCount++;
+}
+}
+free( bufferList );
+// Determine the buffer size.
+AudioValueRange	bufferRange;
+dataSize = sizeof( AudioValueRange );
+property.mSelector = kAudioDevicePropertyBufferFrameSizeRange;
+result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &bufferRange );
+if ( result != noErr ) {
+errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting buffer size range for device (" << device << ").";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+if ( bufferRange.mMinimum > *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMinimum;
+else if ( bufferRange.mMaximum < *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMaximum;
+if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) *bufferSize = (unsigned long) bufferRange.mMinimum;
+// Set the buffer size.  For multiple streams, I'm assuming we only
+// need to make this setting for the master channel.
+UInt32 theSize = (UInt32) *bufferSize;
+dataSize = sizeof( UInt32 );
+property.mSelector = kAudioDevicePropertyBufferFrameSize;
+result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &theSize );
+if ( result != noErr ) {
+errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting the buffer size for device (" << device << ").";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+// If attempting to setup a duplex stream, the bufferSize parameter
+// MUST be the same in both directions!
+*bufferSize = theSize;
+if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {
+errorStream_ << "RtApiCore::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << device << ").";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+stream_.bufferSize = *bufferSize;
+stream_.nBuffers = 1;
+// Try to set "hog" mode ... it's not clear to me this is working.
+if ( options && options->flags & RTAUDIO_HOG_DEVICE ) {
+pid_t hog_pid;
+dataSize = sizeof( hog_pid );
+property.mSelector = kAudioDevicePropertyHogMode;
+result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &hog_pid );
+if ( result != noErr ) {
+errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting 'hog' state!";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+if ( hog_pid != getpid() ) {
+hog_pid = getpid();
+result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &hog_pid );
+if ( result != noErr ) {
+errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting 'hog' state!";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+}
+}
+// Check and if necessary, change the sample rate for the device.
+Float64 nominalRate;
+dataSize = sizeof( Float64 );
+property.mSelector = kAudioDevicePropertyNominalSampleRate;
+result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &nominalRate );
+if ( result != noErr ) {
+errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting current sample rate.";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+// Only change the sample rate if off by more than 1 Hz.
+if ( fabs( nominalRate - (double)sampleRate ) > 1.0 ) {
+// Set a property listener for the sample rate change
+Float64 reportedRate = 0.0;
+AudioObjectPropertyAddress tmp = { kAudioDevicePropertyNominalSampleRate, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
+result = AudioObjectAddPropertyListener( id, &tmp, rateListener, (void *) &reportedRate );
+if ( result != noErr ) {
+errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate property listener for device (" << device << ").";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+nominalRate = (Float64) sampleRate;
+result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &nominalRate );
+if ( result != noErr ) {
+AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate );
+errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate for device (" << device << ").";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+// Now wait until the reported nominal rate is what we just set.
+UInt32 microCounter = 0;
+while ( reportedRate != nominalRate ) {
+microCounter += 5000;
+if ( microCounter > 5000000 ) break;
+usleep( 5000 );
+}
+// Remove the property listener.
+AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate );
+if ( microCounter > 5000000 ) {
+errorStream_ << "RtApiCore::probeDeviceOpen: timeout waiting for sample rate update for device (" << device << ").";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+}
+// Now set the stream format for all streams.  Also, check the
+// physical format of the device and change that if necessary.
+AudioStreamBasicDescription	description;
+dataSize = sizeof( AudioStreamBasicDescription );
+property.mSelector = kAudioStreamPropertyVirtualFormat;
+result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &description );
+if ( result != noErr ) {
+errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream format for device (" << device << ").";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+// Set the sample rate and data format id.  However, only make the
+// change if the sample rate is not within 1.0 of the desired
+// rate and the format is not linear pcm.
+bool updateFormat = false;
+if ( fabs( description.mSampleRate - (Float64)sampleRate ) > 1.0 ) {
+description.mSampleRate = (Float64) sampleRate;
+updateFormat = true;
+}
+if ( description.mFormatID != kAudioFormatLinearPCM ) {
+description.mFormatID = kAudioFormatLinearPCM;
+updateFormat = true;
+}
+if ( updateFormat ) {
+result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &description );
+if ( result != noErr ) {
+errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate or data format for device (" << device << ").";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+}
+// Now check the physical format.
+property.mSelector = kAudioStreamPropertyPhysicalFormat;
+result = AudioObjectGetPropertyData( id, &property, 0, NULL,  &dataSize, &description );
+if ( result != noErr ) {
+errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream physical format for device (" << device << ").";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+//std::cout << "Current physical stream format:" << std::endl;
+//std::cout << "   mBitsPerChan = " << description.mBitsPerChannel << std::endl;
+//std::cout << "   aligned high = " << (description.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (description.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl;
+//std::cout << "   bytesPerFrame = " << description.mBytesPerFrame << std::endl;
+//std::cout << "   sample rate = " << description.mSampleRate << std::endl;
+if ( description.mFormatID != kAudioFormatLinearPCM || description.mBitsPerChannel < 16 ) {
+description.mFormatID = kAudioFormatLinearPCM;
+//description.mSampleRate = (Float64) sampleRate;
+AudioStreamBasicDescription	testDescription = description;
+UInt32 formatFlags;
+// We'll try higher bit rates first and then work our way down.
+std::vector< std::pair<UInt32, UInt32>  > physicalFormats;
+formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsFloat) & ~kLinearPCMFormatFlagIsSignedInteger;
+physicalFormats.push_back( std::pair<Float32, UInt32>( 32, formatFlags ) );
+formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat;
+physicalFormats.push_back( std::pair<Float32, UInt32>( 32, formatFlags ) );
+physicalFormats.push_back( std::pair<Float32, UInt32>( 24, formatFlags ) );   // 24-bit packed
+formatFlags &= ~( kAudioFormatFlagIsPacked | kAudioFormatFlagIsAlignedHigh );
+physicalFormats.push_back( std::pair<Float32, UInt32>( 24.2, formatFlags ) ); // 24-bit in 4 bytes, aligned low
+formatFlags |= kAudioFormatFlagIsAlignedHigh;
+physicalFormats.push_back( std::pair<Float32, UInt32>( 24.4, formatFlags ) ); // 24-bit in 4 bytes, aligned high
+formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat;
+physicalFormats.push_back( std::pair<Float32, UInt32>( 16, formatFlags ) );
+physicalFormats.push_back( std::pair<Float32, UInt32>( 8, formatFlags ) );
+bool setPhysicalFormat = false;
+for( unsigned int i=0; i<physicalFormats.size(); i++ ) {
+testDescription = description;
+testDescription.mBitsPerChannel = (UInt32) physicalFormats[i].first;
+testDescription.mFormatFlags = physicalFormats[i].second;
+if ( (24 == (UInt32)physicalFormats[i].first) && ~( physicalFormats[i].second & kAudioFormatFlagIsPacked ) )
+testDescription.mBytesPerFrame =  4 * testDescription.mChannelsPerFrame;
+else
+testDescription.mBytesPerFrame =  testDescription.mBitsPerChannel/8 * testDescription.mChannelsPerFrame;
+testDescription.mBytesPerPacket = testDescription.mBytesPerFrame * testDescription.mFramesPerPacket;
+result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &testDescription );
+if ( result == noErr ) {
+setPhysicalFormat = true;
+//std::cout << "Updated physical stream format:" << std::endl;
+//std::cout << "   mBitsPerChan = " << testDescription.mBitsPerChannel << std::endl;
+//std::cout << "   aligned high = " << (testDescription.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (testDescription.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl;
+//std::cout << "   bytesPerFrame = " << testDescription.mBytesPerFrame << std::endl;
+//std::cout << "   sample rate = " << testDescription.mSampleRate << std::endl;
+break;
+}
+}
+if ( !setPhysicalFormat ) {
+errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting physical data format for device (" << device << ").";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+} // done setting virtual/physical formats.
+// Get the stream / device latency.
+UInt32 latency;
+dataSize = sizeof( UInt32 );
+property.mSelector = kAudioDevicePropertyLatency;
+if ( AudioObjectHasProperty( id, &property ) == true ) {
+result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &latency );
+if ( result == kAudioHardwareNoError ) stream_.latency[ mode ] = latency;
+else {
+errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting device latency for device (" << device << ").";
+errorText_ = errorStream_.str();
+error( RtAudioError::WARNING );
+}
+}
+// Byte-swapping: According to AudioHardware.h, the stream data will
+// always be presented in native-endian format, so we should never
+// need to byte swap.
+stream_.doByteSwap[mode] = false;
+// From the CoreAudio documentation, PCM data must be supplied as
+// 32-bit floats.
+stream_.userFormat = format;
+stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
+if ( streamCount == 1 )
+stream_.nDeviceChannels[mode] = description.mChannelsPerFrame;
+else // multiple streams
+stream_.nDeviceChannels[mode] = channels;
+stream_.nUserChannels[mode] = channels;
+stream_.channelOffset[mode] = channelOffset;  // offset within a CoreAudio stream
+if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
+else stream_.userInterleaved = true;
+stream_.deviceInterleaved[mode] = true;
+if ( monoMode == true ) stream_.deviceInterleaved[mode] = false;
+// Set flags for buffer conversion.
+stream_.doConvertBuffer[mode] = false;
+if ( stream_.userFormat != stream_.deviceFormat[mode] )
+stream_.doConvertBuffer[mode] = true;
+if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
+stream_.doConvertBuffer[mode] = true;
+if ( streamCount == 1 ) {
+if ( stream_.nUserChannels[mode] > 1 &&
+stream_.userInterleaved != stream_.deviceInterleaved[mode] )
+stream_.doConvertBuffer[mode] = true;
+}
+else if ( monoMode && stream_.userInterleaved )
+stream_.doConvertBuffer[mode] = true;
+// Allocate our CoreHandle structure for the stream.
+CoreHandle *handle = 0;
+if ( stream_.apiHandle == 0 ) {
+try {
+handle = new CoreHandle;
+}
+catch ( std::bad_alloc& ) {
+errorText_ = "RtApiCore::probeDeviceOpen: error allocating CoreHandle memory.";
+goto error;
+}
+if ( pthread_cond_init( &handle->condition, NULL ) ) {
+errorText_ = "RtApiCore::probeDeviceOpen: error initializing pthread condition variable.";
+goto error;
+}
+stream_.apiHandle = (void *) handle;
+}
+else
+handle = (CoreHandle *) stream_.apiHandle;
+handle->iStream[mode] = firstStream;
+handle->nStreams[mode] = streamCount;
+handle->id[mode] = id;
+// Allocate necessary internal buffers.
+unsigned long bufferBytes;
+bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+//  stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+stream_.userBuffer[mode] = (char *) malloc( bufferBytes * sizeof(char) );
+memset( stream_.userBuffer[mode], 0, bufferBytes * sizeof(char) );
+if ( stream_.userBuffer[mode] == NULL ) {
+errorText_ = "RtApiCore::probeDeviceOpen: error allocating user buffer memory.";
+goto error;
+}
+// If possible, we will make use of the CoreAudio stream buffers as
+// "device buffers".  However, we can't do this if using multiple
+// streams.
+if ( stream_.doConvertBuffer[mode] && handle->nStreams[mode] > 1 ) {
+bool makeBuffer = true;
+bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
+if ( mode == INPUT ) {
+if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
+unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
+if ( bufferBytes <= bytesOut ) makeBuffer = false;
+}
+}
+if ( makeBuffer ) {
+bufferBytes *= *bufferSize;
+if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
+stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
+if ( stream_.deviceBuffer == NULL ) {
+errorText_ = "RtApiCore::probeDeviceOpen: error allocating device buffer memory.";
+goto error;
+}
+}
+}
+stream_.sampleRate = sampleRate;
+stream_.device[mode] = device;
+stream_.state = STREAM_STOPPED;
+stream_.callbackInfo.object = (void *) this;
+// Setup the buffer conversion information structure.
+if ( stream_.doConvertBuffer[mode] ) {
+if ( streamCount > 1 ) setConvertInfo( mode, 0 );
+else setConvertInfo( mode, channelOffset );
+}
+if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device )
+// Only one callback procedure per device.
+stream_.mode = DUPLEX;
+else {
+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
+result = AudioDeviceCreateIOProcID( id, callbackHandler, (void *) &stream_.callbackInfo, &handle->procId[mode] );
+#else
+// deprecated in favor of AudioDeviceCreateIOProcID()
+result = AudioDeviceAddIOProc( id, callbackHandler, (void *) &stream_.callbackInfo );
+#endif
+if ( result != noErr ) {
+errorStream_ << "RtApiCore::probeDeviceOpen: system error setting callback for device (" << device << ").";
+errorText_ = errorStream_.str();
+goto error;
+}
+if ( stream_.mode == OUTPUT && mode == INPUT )
+stream_.mode = DUPLEX;
+else
+stream_.mode = mode;
+}
+// Setup the device property listener for over/underload.
+property.mSelector = kAudioDeviceProcessorOverload;
+property.mScope = kAudioObjectPropertyScopeGlobal;
+result = AudioObjectAddPropertyListener( id, &property, xrunListener, (void *) handle );
+return SUCCESS;
+error:
+if ( handle ) {
+pthread_cond_destroy( &handle->condition );
+delete handle;
+stream_.apiHandle = 0;
+}
+for ( int i=0; i<2; i++ ) {
+if ( stream_.userBuffer[i] ) {
+free( stream_.userBuffer[i] );
+stream_.userBuffer[i] = 0;
+}
+}
+if ( stream_.deviceBuffer ) {
+free( stream_.deviceBuffer );
+stream_.deviceBuffer = 0;
+}
+stream_.state = STREAM_CLOSED;
+return FAILURE;
+}
+void RtApiCore :: closeStream( void )
+{
+if ( stream_.state == STREAM_CLOSED ) {
+errorText_ = "RtApiCore::closeStream(): no open stream to close!";
+error( RtAudioError::WARNING );
+return;
+}
+CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
+if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+if (handle) {
+AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
+kAudioObjectPropertyScopeGlobal,
+kAudioObjectPropertyElementMaster };
+property.mSelector = kAudioDeviceProcessorOverload;
+property.mScope = kAudioObjectPropertyScopeGlobal;
+if (AudioObjectRemovePropertyListener( handle->id[0], &property, xrunListener, (void *) handle ) != noErr) {
+errorText_ = "RtApiCore::closeStream(): error removing property listener!";
+error( RtAudioError::WARNING );
+}
+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
+if ( stream_.state == STREAM_RUNNING )
+AudioDeviceStop( handle->id[0], handle->procId[0] );
+AudioDeviceDestroyIOProcID( handle->id[0], handle->procId[0] );
+#else // deprecated behaviour
+if ( stream_.state == STREAM_RUNNING )
+AudioDeviceStop( handle->id[0], callbackHandler );
+AudioDeviceRemoveIOProc( handle->id[0], callbackHandler );
+#endif
+}
+}
+if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
+if (handle) {
+AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
+kAudioObjectPropertyScopeGlobal,
+kAudioObjectPropertyElementMaster };
+property.mSelector = kAudioDeviceProcessorOverload;
+property.mScope = kAudioObjectPropertyScopeGlobal;
+if (AudioObjectRemovePropertyListener( handle->id[1], &property, xrunListener, (void *) handle ) != noErr) {
+errorText_ = "RtApiCore::closeStream(): error removing property listener!";
+error( RtAudioError::WARNING );
+}
+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
+if ( stream_.state == STREAM_RUNNING )
+AudioDeviceStop( handle->id[1], handle->procId[1] );
+AudioDeviceDestroyIOProcID( handle->id[1], handle->procId[1] );
+#else // deprecated behaviour
+if ( stream_.state == STREAM_RUNNING )
+AudioDeviceStop( handle->id[1], callbackHandler );
+AudioDeviceRemoveIOProc( handle->id[1], callbackHandler );
+#endif
+}
+}
+for ( int i=0; i<2; i++ ) {
+if ( stream_.userBuffer[i] ) {
+free( stream_.userBuffer[i] );
+stream_.userBuffer[i] = 0;
+}
+}
+if ( stream_.deviceBuffer ) {
+free( stream_.deviceBuffer );
+stream_.deviceBuffer = 0;
+}
+// Destroy pthread condition variable.
+pthread_cond_destroy( &handle->condition );
+delete handle;
+stream_.apiHandle = 0;
+stream_.mode = UNINITIALIZED;
+stream_.state = STREAM_CLOSED;
+}
+void RtApiCore :: startStream( void )
+{
+verifyStream();
+if ( stream_.state == STREAM_RUNNING ) {
+errorText_ = "RtApiCore::startStream(): the stream is already running!";
+error( RtAudioError::WARNING );
+return;
+}
+#if defined( HAVE_GETTIMEOFDAY )
+gettimeofday( &stream_.lastTickTimestamp, NULL );
+#endif
+OSStatus result = noErr;
+CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
+if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
+result = AudioDeviceStart( handle->id[0], handle->procId[0] );
+#else // deprecated behaviour
+result = AudioDeviceStart( handle->id[0], callbackHandler );
+#endif
+if ( result != noErr ) {
+errorStream_ << "RtApiCore::startStream: system error (" << getErrorCode( result ) << ") starting callback procedure on device (" << stream_.device[0] << ").";
+errorText_ = errorStream_.str();
+goto unlock;
+}
+}
+if ( stream_.mode == INPUT ||
+( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
+result = AudioDeviceStart( handle->id[1], handle->procId[1] );
+#else // deprecated behaviour
+result = AudioDeviceStart( handle->id[1], callbackHandler );
+#endif
+if ( result != noErr ) {
+errorStream_ << "RtApiCore::startStream: system error starting input callback procedure on device (" << stream_.device[1] << ").";
+errorText_ = errorStream_.str();
+goto unlock;
+}
+}
+handle->drainCounter = 0;
+handle->internalDrain = false;
+stream_.state = STREAM_RUNNING;
+unlock:
+if ( result == noErr ) return;
+error( RtAudioError::SYSTEM_ERROR );
+}
+void RtApiCore :: stopStream( void )
+{
+verifyStream();
+if ( stream_.state == STREAM_STOPPED ) {
+errorText_ = "RtApiCore::stopStream(): the stream is already stopped!";
+error( RtAudioError::WARNING );
+return;
+}
+OSStatus result = noErr;
+CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
+if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+if ( handle->drainCounter == 0 ) {
+handle->drainCounter = 2;
+pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
+}
+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
+result = AudioDeviceStop( handle->id[0], handle->procId[0] );
+#else // deprecated behaviour
+result = AudioDeviceStop( handle->id[0], callbackHandler );
+#endif
+if ( result != noErr ) {
+errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping callback procedure on device (" << stream_.device[0] << ").";
+errorText_ = errorStream_.str();
+goto unlock;
+}
+}
+if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
+result = AudioDeviceStop( handle->id[1], handle->procId[1] );
+#else  // deprecated behaviour
+result = AudioDeviceStop( handle->id[1], callbackHandler );
+#endif
+if ( result != noErr ) {
+errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping input callback procedure on device (" << stream_.device[1] << ").";
+errorText_ = errorStream_.str();
+goto unlock;
+}
+}
+stream_.state = STREAM_STOPPED;
+unlock:
+if ( result == noErr ) return;
+error( RtAudioError::SYSTEM_ERROR );
+}
+void RtApiCore :: abortStream( void )
+{
+verifyStream();
+if ( stream_.state == STREAM_STOPPED ) {
+errorText_ = "RtApiCore::abortStream(): the stream is already stopped!";
+error( RtAudioError::WARNING );
+return;
+}
+CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
+handle->drainCounter = 2;
+stopStream();
+}
+// This function will be called by a spawned thread when the user
+// callback function signals that the stream should be stopped or
+// aborted.  It is better to handle it this way because the
+// callbackEvent() function probably should return before the AudioDeviceStop()
+// function is called.
+static void *coreStopStream( void *ptr )
+{
+CallbackInfo *info = (CallbackInfo *) ptr;
+RtApiCore *object = (RtApiCore *) info->object;
+object->stopStream();
+pthread_exit( NULL );
+}
+bool RtApiCore :: callbackEvent( AudioDeviceID deviceId,
+const AudioBufferList *inBufferList,
+const AudioBufferList *outBufferList )
+{
+if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;
+if ( stream_.state == STREAM_CLOSED ) {
+errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";
+error( RtAudioError::WARNING );
+return FAILURE;
+}
+CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
+CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
+// Check if we were draining the stream and signal is finished.
+if ( handle->drainCounter > 3 ) {
+ThreadHandle threadId;
+stream_.state = STREAM_STOPPING;
+if ( handle->internalDrain == true )
+pthread_create( &threadId, NULL, coreStopStream, info );
+else // external call to stopStream()
+pthread_cond_signal( &handle->condition );
+return SUCCESS;
+}
+AudioDeviceID outputDevice = handle->id[0];
+// Invoke user callback to get fresh output data UNLESS we are
+// draining stream or duplex mode AND the input/output devices are
+// different AND this function is called for the input device.
+if ( handle->drainCounter == 0 && ( stream_.mode != DUPLEX || deviceId == outputDevice ) ) {
+RtAudioCallback callback = (RtAudioCallback) info->callback;
+double streamTime = getStreamTime();
+RtAudioStreamStatus status = 0;
+if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
+status |= RTAUDIO_OUTPUT_UNDERFLOW;
+handle->xrun[0] = false;
+}
+if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
+status |= RTAUDIO_INPUT_OVERFLOW;
+handle->xrun[1] = false;
+}
+int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+stream_.bufferSize, streamTime, status, info->userData );
+if ( cbReturnValue == 2 ) {
+stream_.state = STREAM_STOPPING;
+handle->drainCounter = 2;
+abortStream();
+return SUCCESS;
+}
+else if ( cbReturnValue == 1 ) {
+handle->drainCounter = 1;
+handle->internalDrain = true;
+}
+}
+if ( stream_.mode == OUTPUT || ( stream_.mode == DUPLEX && deviceId == outputDevice ) ) {
+if ( handle->drainCounter > 1 ) { // write zeros to the output stream
+if ( handle->nStreams[0] == 1 ) {
+memset( outBufferList->mBuffers[handle->iStream[0]].mData,
+0,
+outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );
+}
+else { // fill multiple streams with zeros
+for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) {
+memset( outBufferList->mBuffers[handle->iStream[0]+i].mData,
+0,
+outBufferList->mBuffers[handle->iStream[0]+i].mDataByteSize );
+}
+}
+}
+else if ( handle->nStreams[0] == 1 ) {
+if ( stream_.doConvertBuffer[0] ) { // convert directly to CoreAudio stream buffer
+convertBuffer( (char *) outBufferList->mBuffers[handle->iStream[0]].mData,
+stream_.userBuffer[0], stream_.convertInfo[0] );
+}
+else { // copy from user buffer
+memcpy( outBufferList->mBuffers[handle->iStream[0]].mData,
+stream_.userBuffer[0],
+outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );
+}
+}
+else { // fill multiple streams
+Float32 *inBuffer = (Float32 *) stream_.userBuffer[0];
+if ( stream_.doConvertBuffer[0] ) {
+convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
+inBuffer = (Float32 *) stream_.deviceBuffer;
+}
+if ( stream_.deviceInterleaved[0] == false ) { // mono mode
+UInt32 bufferBytes = outBufferList->mBuffers[handle->iStream[0]].mDataByteSize;
+for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
+memcpy( outBufferList->mBuffers[handle->iStream[0]+i].mData,
+(void *)&inBuffer[i*stream_.bufferSize], bufferBytes );
+}
+}
+else { // fill multiple multi-channel streams with interleaved data
+UInt32 streamChannels, channelsLeft, inJump, outJump, inOffset;
+Float32 *out, *in;
+bool inInterleaved = ( stream_.userInterleaved ) ? true : false;
+UInt32 inChannels = stream_.nUserChannels[0];
+if ( stream_.doConvertBuffer[0] ) {
+inInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode
+inChannels = stream_.nDeviceChannels[0];
+}
+if ( inInterleaved ) inOffset = 1;
+else inOffset = stream_.bufferSize;
+channelsLeft = inChannels;
+for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) {
+in = inBuffer;
+out = (Float32 *) outBufferList->mBuffers[handle->iStream[0]+i].mData;
+streamChannels = outBufferList->mBuffers[handle->iStream[0]+i].mNumberChannels;
+outJump = 0;
+// Account for possible channel offset in first stream
+if ( i == 0 && stream_.channelOffset[0] > 0 ) {
+streamChannels -= stream_.channelOffset[0];
+outJump = stream_.channelOffset[0];
+out += outJump;
+}
+// Account for possible unfilled channels at end of the last stream
+if ( streamChannels > channelsLeft ) {
+outJump = streamChannels - channelsLeft;
+streamChannels = channelsLeft;
+}
+// Determine input buffer offsets and skips
+if ( inInterleaved ) {
+inJump = inChannels;
+in += inChannels - channelsLeft;
+}
+else {
+inJump = 1;
+in += (inChannels - channelsLeft) * inOffset;
+}
+for ( unsigned int i=0; i<stream_.bufferSize; i++ ) {
+for ( unsigned int j=0; j<streamChannels; j++ ) {
+*out++ = in[j*inOffset];
+}
+out += outJump;
+in += inJump;
+}
+channelsLeft -= streamChannels;
+}
+}
+}
+}
+// Don't bother draining input
+if ( handle->drainCounter ) {
+handle->drainCounter++;
+goto unlock;
+}
+AudioDeviceID inputDevice;
+inputDevice = handle->id[1];
+if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && deviceId == inputDevice ) ) {
+if ( handle->nStreams[1] == 1 ) {
+if ( stream_.doConvertBuffer[1] ) { // convert directly from CoreAudio stream buffer
+convertBuffer( stream_.userBuffer[1],
+(char *) inBufferList->mBuffers[handle->iStream[1]].mData,
+stream_.convertInfo[1] );
+}
+else { // copy to user buffer
+memcpy( stream_.userBuffer[1],
+inBufferList->mBuffers[handle->iStream[1]].mData,
+inBufferList->mBuffers[handle->iStream[1]].mDataByteSize );
+}
+}
+else { // read from multiple streams
+Float32 *outBuffer = (Float32 *) stream_.userBuffer[1];
+if ( stream_.doConvertBuffer[1] ) outBuffer = (Float32 *) stream_.deviceBuffer;
+if ( stream_.deviceInterleaved[1] == false ) { // mono mode
+UInt32 bufferBytes = inBufferList->mBuffers[handle->iStream[1]].mDataByteSize;
+for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
+memcpy( (void *)&outBuffer[i*stream_.bufferSize],
+inBufferList->mBuffers[handle->iStream[1]+i].mData, bufferBytes );
+}
+}
+else { // read from multiple multi-channel streams
+UInt32 streamChannels, channelsLeft, inJump, outJump, outOffset;
+Float32 *out, *in;
+bool outInterleaved = ( stream_.userInterleaved ) ? true : false;
+UInt32 outChannels = stream_.nUserChannels[1];
+if ( stream_.doConvertBuffer[1] ) {
+outInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode
+outChannels = stream_.nDeviceChannels[1];
+}
+if ( outInterleaved ) outOffset = 1;
+else outOffset = stream_.bufferSize;
+channelsLeft = outChannels;
+for ( unsigned int i=0; i<handle->nStreams[1]; i++ ) {
+out = outBuffer;
+in = (Float32 *) inBufferList->mBuffers[handle->iStream[1]+i].mData;
+streamChannels = inBufferList->mBuffers[handle->iStream[1]+i].mNumberChannels;
+inJump = 0;
+// Account for possible channel offset in first stream
+if ( i == 0 && stream_.channelOffset[1] > 0 ) {
+streamChannels -= stream_.channelOffset[1];
+inJump = stream_.channelOffset[1];
+in += inJump;
+}
+// Account for possible unread channels at end of the last stream
+if ( streamChannels > channelsLeft ) {
+inJump = streamChannels - channelsLeft;
+streamChannels = channelsLeft;
+}
+// Determine output buffer offsets and skips
+if ( outInterleaved ) {
+outJump = outChannels;
+out += outChannels - channelsLeft;
+}
+else {
+outJump = 1;
+out += (outChannels - channelsLeft) * outOffset;
+}
+for ( unsigned int i=0; i<stream_.bufferSize; i++ ) {
+for ( unsigned int j=0; j<streamChannels; j++ ) {
+out[j*outOffset] = *in++;
+}
+out += outJump;
+in += inJump;
+}
+channelsLeft -= streamChannels;
+}
+}
+if ( stream_.doConvertBuffer[1] ) { // convert from our internal "device" buffer
+convertBuffer( stream_.userBuffer[1],
+stream_.deviceBuffer,
+stream_.convertInfo[1] );
+}
+}
+}
+unlock:
+//MUTEX_UNLOCK( &stream_.mutex );
+// Make sure to only tick duplex stream time once if using two devices
+if ( stream_.mode != DUPLEX || (stream_.mode == DUPLEX && handle->id[0] != handle->id[1] && deviceId == handle->id[0] ) )
+RtApi::tickStreamTime();
+return SUCCESS;
+}
+const char* RtApiCore :: getErrorCode( OSStatus code )
+{
+switch( code ) {
+case kAudioHardwareNotRunningError:
+return "kAudioHardwareNotRunningError";
+case kAudioHardwareUnspecifiedError:
+return "kAudioHardwareUnspecifiedError";
+case kAudioHardwareUnknownPropertyError:
+return "kAudioHardwareUnknownPropertyError";
+case kAudioHardwareBadPropertySizeError:
+return "kAudioHardwareBadPropertySizeError";
+case kAudioHardwareIllegalOperationError:
+return "kAudioHardwareIllegalOperationError";
+case kAudioHardwareBadObjectError:
+return "kAudioHardwareBadObjectError";
+case kAudioHardwareBadDeviceError:
+return "kAudioHardwareBadDeviceError";
+case kAudioHardwareBadStreamError:
+return "kAudioHardwareBadStreamError";
+case kAudioHardwareUnsupportedOperationError:
+return "kAudioHardwareUnsupportedOperationError";
+case kAudioDeviceUnsupportedFormatError:
+return "kAudioDeviceUnsupportedFormatError";
+case kAudioDevicePermissionsError:
+return "kAudioDevicePermissionsError";
+default:
+return "CoreAudio unknown error";
+}
+}
+//******************** End of __MACOSX_CORE__ *********************//
+#endif
+#if defined(__UNIX_JACK__)
+// JACK is a low-latency audio server, originally written for the
+// GNU/Linux operating system and now also ported to OS-X. It can
+// connect a number of different applications to an audio device, as
+// well as allowing them to share audio between themselves.
+//
+// When using JACK with RtAudio, "devices" refer to JACK clients that
+// have ports connected to the server.  The JACK server is typically
+// started in a terminal as follows:
+//
+// .jackd -d alsa -d hw:0
+//
+// or through an interface program such as qjackctl.  Many of the
+// parameters normally set for a stream are fixed by the JACK server
+// and can be specified when the JACK server is started.  In
+// particular,
+//
+// .jackd -d alsa -d hw:0 -r 44100 -p 512 -n 4
+//
+// specifies a sample rate of 44100 Hz, a buffer size of 512 sample
+// frames, and number of buffers = 4.  Once the server is running, it
+// is not possible to override these values.  If the values are not
+// specified in the command-line, the JACK server uses default values.
+//
+// The JACK server does not have to be running when an instance of
+// RtApiJack is created, though the function getDeviceCount() will
+// report 0 devices found until JACK has been started.  When no
+// devices are available (i.e., the JACK server is not running), a
+// stream cannot be opened.
+#include <jack/jack.h>
+#include <unistd.h>
+#include <cstdio>
+// A structure to hold various information related to the Jack API
+// implementation.
+struct JackHandle {
+jack_client_t *client;
+jack_port_t **ports[2];
+std::string deviceName[2];
+bool xrun[2];
+pthread_cond_t condition;
+int drainCounter;       // Tracks callback counts when draining
+bool internalDrain;     // Indicates if stop is initiated from callback or not.
+JackHandle()
+:client(0), drainCounter(0), internalDrain(false) { ports[0] = 0; ports[1] = 0; xrun[0] = false; xrun[1] = false; }
+};
+#if !defined(__RTAUDIO_DEBUG__)
+static void jackSilentError( const char * ) {};
+#endif
+RtApiJack :: RtApiJack()
+:shouldAutoconnect_(true) {
+// Nothing to do here.
+#if !defined(__RTAUDIO_DEBUG__)
+// Turn off Jack's internal error reporting.
+jack_set_error_function( &jackSilentError );
+#endif
+}
+RtApiJack :: ~RtApiJack()
+{
+if ( stream_.state != STREAM_CLOSED ) closeStream();
+}
+unsigned int RtApiJack :: getDeviceCount( void )
+{
+// See if we can become a jack client.
+jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption;
+jack_status_t *status = NULL;
+jack_client_t *client = jack_client_open( "RtApiJackCount", options, status );
+if ( client == 0 ) return 0;
+const char **ports;
+std::string port, previousPort;
+unsigned int nChannels = 0, nDevices = 0;
+ports = jack_get_ports( client, NULL, JACK_DEFAULT_AUDIO_TYPE, 0 );
+if ( ports ) {
+// Parse the port names up to the first colon (:).
+size_t iColon = 0;
+do {
+port = (char *) ports[ nChannels ];
+iColon = port.find(":");
+if ( iColon != std::string::npos ) {
+port = port.substr( 0, iColon + 1 );
+if ( port != previousPort ) {
+nDevices++;
+previousPort = port;
+}
+}
+} while ( ports[++nChannels] );
+free( ports );
+}
+jack_client_close( client );
+return nDevices;
+}
+RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device )
+{
+RtAudio::DeviceInfo info;
+info.probed = false;
+jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption
+jack_status_t *status = NULL;
+jack_client_t *client = jack_client_open( "RtApiJackInfo", options, status );
+if ( client == 0 ) {
+errorText_ = "RtApiJack::getDeviceInfo: Jack server not found or connection error!";
+error( RtAudioError::WARNING );
+return info;
+}
+const char **ports;
+std::string port, previousPort;
+unsigned int nPorts = 0, nDevices = 0;
+ports = jack_get_ports( client, NULL, JACK_DEFAULT_AUDIO_TYPE, 0 );
+if ( ports ) {
+// Parse the port names up to the first colon (:).
+size_t iColon = 0;
+do {
+port = (char *) ports[ nPorts ];
+iColon = port.find(":");
+if ( iColon != std::string::npos ) {
+port = port.substr( 0, iColon );
+if ( port != previousPort ) {
+if ( nDevices == device ) info.name = port;
+nDevices++;
+previousPort = port;
+}
+}
+} while ( ports[++nPorts] );
+free( ports );
+}
+if ( device >= nDevices ) {
+jack_client_close( client );
+errorText_ = "RtApiJack::getDeviceInfo: device ID is invalid!";
+error( RtAudioError::INVALID_USE );
+return info;
+}
+// Get the current jack server sample rate.
+info.sampleRates.clear();
+info.preferredSampleRate = jack_get_sample_rate( client );
+info.sampleRates.push_back( info.preferredSampleRate );
+// Count the available ports containing the client name as device
+// channels.  Jack "input ports" equal RtAudio output channels.
+unsigned int nChannels = 0;
+ports = jack_get_ports( client, info.name.c_str(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput );
+if ( ports ) {
+while ( ports[ nChannels ] ) nChannels++;
+free( ports );
+info.outputChannels = nChannels;
+}
+// Jack "output ports" equal RtAudio input channels.
+nChannels = 0;
+ports = jack_get_ports( client, info.name.c_str(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput );
+if ( ports ) {
+while ( ports[ nChannels ] ) nChannels++;
+free( ports );
+info.inputChannels = nChannels;
+}
+if ( info.outputChannels == 0 && info.inputChannels == 0 ) {
+jack_client_close(client);
+errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!";
+error( RtAudioError::WARNING );
+return info;
+}
+// If device opens for both playback and capture, we determine the channels.
+if ( info.outputChannels > 0 && info.inputChannels > 0 )
+info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
+// Jack always uses 32-bit floats.
+info.nativeFormats = RTAUDIO_FLOAT32;
+// Jack doesn't provide default devices so we'll use the first available one.
+if ( device == 0 && info.outputChannels > 0 )
+info.isDefaultOutput = true;
+if ( device == 0 && info.inputChannels > 0 )
+info.isDefaultInput = true;
+jack_client_close(client);
+info.probed = true;
+return info;
+}
+static int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer )
+{
+CallbackInfo *info = (CallbackInfo *) infoPointer;
+RtApiJack *object = (RtApiJack *) info->object;
+if ( object->callbackEvent( (unsigned long) nframes ) == false ) return 1;
+return 0;
+}
+// This function will be called by a spawned thread when the Jack
+// server signals that it is shutting down.  It is necessary to handle
+// it this way because the jackShutdown() function must return before
+// the jack_deactivate() function (in closeStream()) will return.
+static void *jackCloseStream( void *ptr )
+{
+CallbackInfo *info = (CallbackInfo *) ptr;
+RtApiJack *object = (RtApiJack *) info->object;
+object->closeStream();
+pthread_exit( NULL );
+}
+static void jackShutdown( void *infoPointer )
+{
+CallbackInfo *info = (CallbackInfo *) infoPointer;
+RtApiJack *object = (RtApiJack *) info->object;
+// Check current stream state.  If stopped, then we'll assume this
+// was called as a result of a call to RtApiJack::stopStream (the
+// deactivation of a client handle causes this function to be called).
+// If not, we'll assume the Jack server is shutting down or some
+// other problem occurred and we should close the stream.
+if ( object->isStreamRunning() == false ) return;
+ThreadHandle threadId;
+pthread_create( &threadId, NULL, jackCloseStream, info );
+std::cerr << "\nRtApiJack: the Jack server is shutting down this client ... stream stopped and closed!!\n" << std::endl;
+}
+static int jackXrun( void *infoPointer )
+{
+JackHandle *handle = *((JackHandle **) infoPointer);
+if ( handle->ports[0] ) handle->xrun[0] = true;
+if ( handle->ports[1] ) handle->xrun[1] = true;
+return 0;
+}
+bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+unsigned int firstChannel, unsigned int sampleRate,
+RtAudioFormat format, unsigned int *bufferSize,
+RtAudio::StreamOptions *options )
+{
+JackHandle *handle = (JackHandle *) stream_.apiHandle;
+// Look for jack server and try to become a client (only do once per stream).
+jack_client_t *client = 0;
+if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) {
+jack_options_t jackoptions = (jack_options_t) ( JackNoStartServer ); //JackNullOption;
+jack_status_t *status = NULL;
+if ( options && !options->streamName.empty() )
+client = jack_client_open( options->streamName.c_str(), jackoptions, status );
+else
+client = jack_client_open( "RtApiJack", jackoptions, status );
+if ( client == 0 ) {
+errorText_ = "RtApiJack::probeDeviceOpen: Jack server not found or connection error!";
+error( RtAudioError::WARNING );
+return FAILURE;
+}
+}
+else {
+// The handle must have been created on an earlier pass.
+client = handle->client;
+}
+const char **ports;
+std::string port, previousPort, deviceName;
+unsigned int nPorts = 0, nDevices = 0;
+ports = jack_get_ports( client, NULL, JACK_DEFAULT_AUDIO_TYPE, 0 );
+if ( ports ) {
+// Parse the port names up to the first colon (:).
+size_t iColon = 0;
+do {
+port = (char *) ports[ nPorts ];
+iColon = port.find(":");
+if ( iColon != std::string::npos ) {
+port = port.substr( 0, iColon );
+if ( port != previousPort ) {
+if ( nDevices == device ) deviceName = port;
+nDevices++;
+previousPort = port;
+}
+}
+} while ( ports[++nPorts] );
+free( ports );
+}
+if ( device >= nDevices ) {
+errorText_ = "RtApiJack::probeDeviceOpen: device ID is invalid!";
+return FAILURE;
+}
+unsigned long flag = JackPortIsInput;
+if ( mode == INPUT ) flag = JackPortIsOutput;
+if ( ! (options && (options->flags & RTAUDIO_JACK_DONT_CONNECT)) ) {
+// Count the available ports containing the client name as device
+// channels.  Jack "input ports" equal RtAudio output channels.
+unsigned int nChannels = 0;
+ports = jack_get_ports( client, deviceName.c_str(), JACK_DEFAULT_AUDIO_TYPE, flag );
+if ( ports ) {
+while ( ports[ nChannels ] ) nChannels++;
+free( ports );
+}
+// Compare the jack ports for specified client to the requested number of channels.
+if ( nChannels < (channels + firstChannel) ) {
+errorStream_ << "RtApiJack::probeDeviceOpen: requested number of channels (" << channels << ") + offset (" << firstChannel << ") not found for specified device (" << device << ":" << deviceName << ").";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+}
+// Check the jack server sample rate.
+unsigned int jackRate = jack_get_sample_rate( client );
+if ( sampleRate != jackRate ) {
+jack_client_close( client );
+errorStream_ << "RtApiJack::probeDeviceOpen: the requested sample rate (" << sampleRate << ") is different than the JACK server rate (" << jackRate << ").";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+stream_.sampleRate = jackRate;
+// Get the latency of the JACK port.
+ports = jack_get_ports( client, deviceName.c_str(), JACK_DEFAULT_AUDIO_TYPE, flag );
+if ( ports[ firstChannel ] ) {
+// Added by Ge Wang
+jack_latency_callback_mode_t cbmode = (mode == INPUT ? JackCaptureLatency : JackPlaybackLatency);
+// the range (usually the min and max are equal)
+jack_latency_range_t latrange; latrange.min = latrange.max = 0;
+// get the latency range
+jack_port_get_latency_range( jack_port_by_name( client, ports[firstChannel] ), cbmode, &latrange );
+// be optimistic, use the min!
+stream_.latency[mode] = latrange.min;
+//stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) );
+}
+free( ports );
+// The jack server always uses 32-bit floating-point data.
+stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
+stream_.userFormat = format;
+if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
+else stream_.userInterleaved = true;
+// Jack always uses non-interleaved buffers.
+stream_.deviceInterleaved[mode] = false;
+// Jack always provides host byte-ordered data.
+stream_.doByteSwap[mode] = false;
+// Get the buffer size.  The buffer size and number of buffers
+// (periods) is set when the jack server is started.
+stream_.bufferSize = (int) jack_get_buffer_size( client );
+*bufferSize = stream_.bufferSize;
+stream_.nDeviceChannels[mode] = channels;
+stream_.nUserChannels[mode] = channels;
+// Set flags for buffer conversion.
+stream_.doConvertBuffer[mode] = false;
+if ( stream_.userFormat != stream_.deviceFormat[mode] )
+stream_.doConvertBuffer[mode] = true;
+if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+stream_.nUserChannels[mode] > 1 )
+stream_.doConvertBuffer[mode] = true;
+// Allocate our JackHandle structure for the stream.
+if ( handle == 0 ) {
+try {
+handle = new JackHandle;
+}
+catch ( std::bad_alloc& ) {
+errorText_ = "RtApiJack::probeDeviceOpen: error allocating JackHandle memory.";
+goto error;
+}
+if ( pthread_cond_init(&handle->condition, NULL) ) {
+errorText_ = "RtApiJack::probeDeviceOpen: error initializing pthread condition variable.";
+goto error;
+}
+stream_.apiHandle = (void *) handle;
+handle->client = client;
+}
+handle->deviceName[mode] = deviceName;
+// Allocate necessary internal buffers.
+unsigned long bufferBytes;
+bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+if ( stream_.userBuffer[mode] == NULL ) {
+errorText_ = "RtApiJack::probeDeviceOpen: error allocating user buffer memory.";
+goto error;
+}
+if ( stream_.doConvertBuffer[mode] ) {
+bool makeBuffer = true;
+if ( mode == OUTPUT )
+bufferBytes = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
+else { // mode == INPUT
+bufferBytes = stream_.nDeviceChannels[1] * formatBytes( stream_.deviceFormat[1] );
+if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
+unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]);
+if ( bufferBytes < bytesOut ) makeBuffer = false;
+}
+}
+if ( makeBuffer ) {
+bufferBytes *= *bufferSize;
+if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
+stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
+if ( stream_.deviceBuffer == NULL ) {
+errorText_ = "RtApiJack::probeDeviceOpen: error allocating device buffer memory.";
+goto error;
+}
+}
+}
+// Allocate memory for the Jack ports (channels) identifiers.
+handle->ports[mode] = (jack_port_t **) malloc ( sizeof (jack_port_t *) * channels );
+if ( handle->ports[mode] == NULL )  {
+errorText_ = "RtApiJack::probeDeviceOpen: error allocating port memory.";
+goto error;
+}
+stream_.device[mode] = device;
+stream_.channelOffset[mode] = firstChannel;
+stream_.state = STREAM_STOPPED;
+stream_.callbackInfo.object = (void *) this;
+if ( stream_.mode == OUTPUT && mode == INPUT )
+// We had already set up the stream for output.
+stream_.mode = DUPLEX;
+else {
+stream_.mode = mode;
+jack_set_process_callback( handle->client, jackCallbackHandler, (void *) &stream_.callbackInfo );
+jack_set_xrun_callback( handle->client, jackXrun, (void *) &stream_.apiHandle );
+jack_on_shutdown( handle->client, jackShutdown, (void *) &stream_.callbackInfo );
+}
+// Register our ports.
+char label[64];
+if ( mode == OUTPUT ) {
+for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
+snprintf( label, 64, "outport %d", i );
+handle->ports[0][i] = jack_port_register( handle->client, (const char *)label,
+JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0 );
+}
+}
+else {
+for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
+snprintf( label, 64, "inport %d", i );
+handle->ports[1][i] = jack_port_register( handle->client, (const char *)label,
+JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0 );
+}
+}
+// Setup the buffer conversion information structure.  We don't use
+// buffers to do channel offsets, so we override that parameter
+// here.
+if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );
+if ( options && options->flags & RTAUDIO_JACK_DONT_CONNECT ) shouldAutoconnect_ = false;
+return SUCCESS;
+error:
+if ( handle ) {
+pthread_cond_destroy( &handle->condition );
+jack_client_close( handle->client );
+if ( handle->ports[0] ) free( handle->ports[0] );
+if ( handle->ports[1] ) free( handle->ports[1] );
+delete handle;
+stream_.apiHandle = 0;
+}
+for ( int i=0; i<2; i++ ) {
+if ( stream_.userBuffer[i] ) {
+free( stream_.userBuffer[i] );
+stream_.userBuffer[i] = 0;
+}
+}
+if ( stream_.deviceBuffer ) {
+free( stream_.deviceBuffer );
+stream_.deviceBuffer = 0;
+}
+return FAILURE;
+}
+void RtApiJack :: closeStream( void )
+{
+if ( stream_.state == STREAM_CLOSED ) {
+errorText_ = "RtApiJack::closeStream(): no open stream to close!";
+error( RtAudioError::WARNING );
+return;
+}
+JackHandle *handle = (JackHandle *) stream_.apiHandle;
+if ( handle ) {
+if ( stream_.state == STREAM_RUNNING )
+jack_deactivate( handle->client );
+jack_client_close( handle->client );
+}
+if ( handle ) {
+if ( handle->ports[0] ) free( handle->ports[0] );
+if ( handle->ports[1] ) free( handle->ports[1] );
+pthread_cond_destroy( &handle->condition );
+delete handle;
+stream_.apiHandle = 0;
+}
+for ( int i=0; i<2; i++ ) {
+if ( stream_.userBuffer[i] ) {
+free( stream_.userBuffer[i] );
+stream_.userBuffer[i] = 0;
+}
+}
+if ( stream_.deviceBuffer ) {
+free( stream_.deviceBuffer );
+stream_.deviceBuffer = 0;
+}
+stream_.mode = UNINITIALIZED;
+stream_.state = STREAM_CLOSED;
+}
+void RtApiJack :: startStream( void )
+{
+verifyStream();
+if ( stream_.state == STREAM_RUNNING ) {
+errorText_ = "RtApiJack::startStream(): the stream is already running!";
+error( RtAudioError::WARNING );
+return;
+}
+#if defined( HAVE_GETTIMEOFDAY )
+gettimeofday( &stream_.lastTickTimestamp, NULL );
+#endif
+JackHandle *handle = (JackHandle *) stream_.apiHandle;
+int result = jack_activate( handle->client );
+if ( result ) {
+errorText_ = "RtApiJack::startStream(): unable to activate JACK client!";
+goto unlock;
+}
+const char **ports;
+// Get the list of available ports.
+if ( shouldAutoconnect_ && (stream_.mode == OUTPUT || stream_.mode == DUPLEX) ) {
+result = 1;
+ports = jack_get_ports( handle->client, handle->deviceName[0].c_str(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput);
+if ( ports == NULL) {
+errorText_ = "RtApiJack::startStream(): error determining available JACK input ports!";
+goto unlock;
+}
+// Now make the port connections.  Since RtAudio wasn't designed to
+// allow the user to select particular channels of a device, we'll
+// just open the first "nChannels" ports with offset.
+for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
+result = 1;
+if ( ports[ stream_.channelOffset[0] + i ] )
+result = jack_connect( handle->client, jack_port_name( handle->ports[0][i] ), ports[ stream_.channelOffset[0] + i ] );
+if ( result ) {
+free( ports );
+errorText_ = "RtApiJack::startStream(): error connecting output ports!";
+goto unlock;
+}
+}
+free(ports);
+}
+if ( shouldAutoconnect_ && (stream_.mode == INPUT || stream_.mode == DUPLEX) ) {
+result = 1;
+ports = jack_get_ports( handle->client, handle->deviceName[1].c_str(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput );
+if ( ports == NULL) {
+errorText_ = "RtApiJack::startStream(): error determining available JACK output ports!";
+goto unlock;
+}
+// Now make the port connections.  See note above.
+for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
+result = 1;
+if ( ports[ stream_.channelOffset[1] + i ] )
+result = jack_connect( handle->client, ports[ stream_.channelOffset[1] + i ], jack_port_name( handle->ports[1][i] ) );
+if ( result ) {
+free( ports );
+errorText_ = "RtApiJack::startStream(): error connecting input ports!";
+goto unlock;
+}
+}
+free(ports);
+}
+handle->drainCounter = 0;
+handle->internalDrain = false;
+stream_.state = STREAM_RUNNING;
+unlock:
+if ( result == 0 ) return;
+error( RtAudioError::SYSTEM_ERROR );
+}
+void RtApiJack :: stopStream( void )
+{
+verifyStream();
+if ( stream_.state == STREAM_STOPPED ) {
+errorText_ = "RtApiJack::stopStream(): the stream is already stopped!";
+error( RtAudioError::WARNING );
+return;
+}
+JackHandle *handle = (JackHandle *) stream_.apiHandle;
+if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+if ( handle->drainCounter == 0 ) {
+handle->drainCounter = 2;
+pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
+}
+}
+jack_deactivate( handle->client );
+stream_.state = STREAM_STOPPED;
+}
+void RtApiJack :: abortStream( void )
+{
+verifyStream();
+if ( stream_.state == STREAM_STOPPED ) {
+errorText_ = "RtApiJack::abortStream(): the stream is already stopped!";
+error( RtAudioError::WARNING );
+return;
+}
+JackHandle *handle = (JackHandle *) stream_.apiHandle;
+handle->drainCounter = 2;
+stopStream();
+}
+// This function will be called by a spawned thread when the user
+// callback function signals that the stream should be stopped or
+// aborted.  It is necessary to handle it this way because the
+// callbackEvent() function must return before the jack_deactivate()
+// function will return.
+static void *jackStopStream( void *ptr )
+{
+CallbackInfo *info = (CallbackInfo *) ptr;
+RtApiJack *object = (RtApiJack *) info->object;
+object->stopStream();
+pthread_exit( NULL );
+}
+bool RtApiJack :: callbackEvent( unsigned long nframes )
+{
+if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;
+if ( stream_.state == STREAM_CLOSED ) {
+errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";
+error( RtAudioError::WARNING );
+return FAILURE;
+}
+if ( stream_.bufferSize != nframes ) {
+errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!";
+error( RtAudioError::WARNING );
+return FAILURE;
+}
+CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
+JackHandle *handle = (JackHandle *) stream_.apiHandle;
+// Check if we were draining the stream and signal is finished.
+if ( handle->drainCounter > 3 ) {
+ThreadHandle threadId;
+stream_.state = STREAM_STOPPING;
+if ( handle->internalDrain == true )
+pthread_create( &threadId, NULL, jackStopStream, info );
+else
+pthread_cond_signal( &handle->condition );
+return SUCCESS;
+}
+// Invoke user callback first, to get fresh output data.
+if ( handle->drainCounter == 0 ) {
+RtAudioCallback callback = (RtAudioCallback) info->callback;
+double streamTime = getStreamTime();
+RtAudioStreamStatus status = 0;
+if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
+status |= RTAUDIO_OUTPUT_UNDERFLOW;
+handle->xrun[0] = false;
+}
+if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
+status |= RTAUDIO_INPUT_OVERFLOW;
+handle->xrun[1] = false;
+}
+int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+stream_.bufferSize, streamTime, status, info->userData );
+if ( cbReturnValue == 2 ) {
+stream_.state = STREAM_STOPPING;
+handle->drainCounter = 2;
+ThreadHandle id;
+pthread_create( &id, NULL, jackStopStream, info );
+return SUCCESS;
+}
+else if ( cbReturnValue == 1 ) {
+handle->drainCounter = 1;
+handle->internalDrain = true;
+}
+}
+jack_default_audio_sample_t *jackbuffer;
+unsigned long bufferBytes = nframes * sizeof( jack_default_audio_sample_t );
+if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+if ( handle->drainCounter > 1 ) { // write zeros to the output stream
+for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
+jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
+memset( jackbuffer, 0, bufferBytes );
+}
+}
+else if ( stream_.doConvertBuffer[0] ) {
+convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
+for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
+jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
+memcpy( jackbuffer, &stream_.deviceBuffer[i*bufferBytes], bufferBytes );
+}
+}
+else { // no buffer conversion
+for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
+jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
+memcpy( jackbuffer, &stream_.userBuffer[0][i*bufferBytes], bufferBytes );
+}
+}
+}
+// Don't bother draining input
+if ( handle->drainCounter ) {
+handle->drainCounter++;
+goto unlock;
+}
+if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+if ( stream_.doConvertBuffer[1] ) {
+for ( unsigned int i=0; i<stream_.nDeviceChannels[1]; i++ ) {
+jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );
+memcpy( &stream_.deviceBuffer[i*bufferBytes], jackbuffer, bufferBytes );
+}
+convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
+}
+else { // no buffer conversion
+for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
+jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );
+memcpy( &stream_.userBuffer[1][i*bufferBytes], jackbuffer, bufferBytes );
+}
+}
+}
+unlock:
+RtApi::tickStreamTime();
+return SUCCESS;
+}
+//******************** End of __UNIX_JACK__ *********************//
+#endif
+#if defined(__WINDOWS_ASIO__) // ASIO API on Windows
+// The ASIO API is designed around a callback scheme, so this
+// implementation is similar to that used for OS-X CoreAudio and Linux
+// Jack.  The primary constraint with ASIO is that it only allows
+// access to a single driver at a time.  Thus, it is not possible to
+// have more than one simultaneous RtAudio stream.
+//
+// This implementation also requires a number of external ASIO files
+// and a few global variables.  The ASIO callback scheme does not
+// allow for the passing of user data, so we must create a global
+// pointer to our callbackInfo structure.
+//
+// On unix systems, we make use of a pthread condition variable.
+// Since there is no equivalent in Windows, I hacked something based
+// on information found in
+// http://www.cs.wustl.edu/~schmidt/win32-cv-1.html.
+#include "asiosys.h"
+#include "asio.h"
+#include "iasiothiscallresolver.h"
+#include "asiodrivers.h"
+#include <cmath>
+static AsioDrivers drivers;
+static ASIOCallbacks asioCallbacks;
+static ASIODriverInfo driverInfo;
+static CallbackInfo *asioCallbackInfo;
+static bool asioXRun;
+struct AsioHandle {
+int drainCounter;       // Tracks callback counts when draining
+bool internalDrain;     // Indicates if stop is initiated from callback or not.
+ASIOBufferInfo *bufferInfos;
+HANDLE condition;
+AsioHandle()
+:drainCounter(0), internalDrain(false), bufferInfos(0) {}
+};
+// Function declarations (definitions at end of section)
+static const char* getAsioErrorString( ASIOError result );
+static void sampleRateChanged( ASIOSampleRate sRate );
+static long asioMessages( long selector, long value, void* message, double* opt );
+RtApiAsio :: RtApiAsio()
+{
+// ASIO cannot run on a multi-threaded appartment. You can call
+// CoInitialize beforehand, but it must be for appartment threading
+// (in which case, CoInitilialize will return S_FALSE here).
+coInitialized_ = false;
+HRESULT hr = CoInitialize( NULL );
+if ( FAILED(hr) ) {
+errorText_ = "RtApiAsio::ASIO requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)";
+error( RtAudioError::WARNING );
+}
+coInitialized_ = true;
+drivers.removeCurrentDriver();
+driverInfo.asioVersion = 2;
+// See note in DirectSound implementation about GetDesktopWindow().
+driverInfo.sysRef = GetForegroundWindow();
+}
+RtApiAsio :: ~RtApiAsio()
+{
+if ( stream_.state != STREAM_CLOSED ) closeStream();
+if ( coInitialized_ ) CoUninitialize();
+}
+unsigned int RtApiAsio :: getDeviceCount( void )
+{
+return (unsigned int) drivers.asioGetNumDev();
+}
+RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device )
+{
+RtAudio::DeviceInfo info;
+info.probed = false;
+// Get device ID
+unsigned int nDevices = getDeviceCount();
+if ( nDevices == 0 ) {
+errorText_ = "RtApiAsio::getDeviceInfo: no devices found!";
+error( RtAudioError::INVALID_USE );
+return info;
+}
+if ( device >= nDevices ) {
+errorText_ = "RtApiAsio::getDeviceInfo: device ID is invalid!";
+error( RtAudioError::INVALID_USE );
+return info;
+}
+// If a stream is already open, we cannot probe other devices.  Thus, use the saved results.
+if ( stream_.state != STREAM_CLOSED ) {
+if ( device >= devices_.size() ) {
+errorText_ = "RtApiAsio::getDeviceInfo: device ID was not present before stream was opened.";
+error( RtAudioError::WARNING );
+return info;
+}
+return devices_[ device ];
+}
+char driverName[32];
+ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 );
+if ( result != ASE_OK ) {
+errorStream_ << "RtApiAsio::getDeviceInfo: unable to get driver name (" << getAsioErrorString( result ) << ").";
+errorText_ = errorStream_.str();
+error( RtAudioError::WARNING );
+return info;
+}
+info.name = driverName;
+if ( !drivers.loadDriver( driverName ) ) {
+errorStream_ << "RtApiAsio::getDeviceInfo: unable to load driver (" << driverName << ").";
+errorText_ = errorStream_.str();
+error( RtAudioError::WARNING );
+return info;
+}
+result = ASIOInit( &driverInfo );
+if ( result != ASE_OK ) {
+errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
+errorText_ = errorStream_.str();
+error( RtAudioError::WARNING );
+return info;
+}
+// Determine the device channel information.
+long inputChannels, outputChannels;
+result = ASIOGetChannels( &inputChannels, &outputChannels );
+if ( result != ASE_OK ) {
+drivers.removeCurrentDriver();
+errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
+errorText_ = errorStream_.str();
+error( RtAudioError::WARNING );
+return info;
+}
+info.outputChannels = outputChannels;
+info.inputChannels = inputChannels;
+if ( info.outputChannels > 0 && info.inputChannels > 0 )
+info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
+// Determine the supported sample rates.
+info.sampleRates.clear();
+for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {
+result = ASIOCanSampleRate( (ASIOSampleRate) SAMPLE_RATES[i] );
+if ( result == ASE_OK ) {
+info.sampleRates.push_back( SAMPLE_RATES[i] );
+if ( !info.preferredSampleRate || ( SAMPLE_RATES[i] <= 48000 && SAMPLE_RATES[i] > info.preferredSampleRate ) )
+info.preferredSampleRate = SAMPLE_RATES[i];
+}
+}
+// Determine supported data types ... just check first channel and assume rest are the same.
+ASIOChannelInfo channelInfo;
+channelInfo.channel = 0;
+channelInfo.isInput = true;
+if ( info.inputChannels <= 0 ) channelInfo.isInput = false;
+result = ASIOGetChannelInfo( &channelInfo );
+if ( result != ASE_OK ) {
+drivers.removeCurrentDriver();
+errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting driver channel info (" << driverName << ").";
+errorText_ = errorStream_.str();
+error( RtAudioError::WARNING );
+return info;
+}
+info.nativeFormats = 0;
+if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB )
+info.nativeFormats |= RTAUDIO_SINT16;
+else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB )
+info.nativeFormats |= RTAUDIO_SINT32;
+else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB )
+info.nativeFormats |= RTAUDIO_FLOAT32;
+else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB )
+info.nativeFormats |= RTAUDIO_FLOAT64;
+else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB )
+info.nativeFormats |= RTAUDIO_SINT24;
+if ( info.outputChannels > 0 )
+if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true;
+if ( info.inputChannels > 0 )
+if ( getDefaultInputDevice() == device ) info.isDefaultInput = true;
+info.probed = true;
+drivers.removeCurrentDriver();
+return info;
+}
+static void bufferSwitch( long index, ASIOBool /*processNow*/ )
+{
+RtApiAsio *object = (RtApiAsio *) asioCallbackInfo->object;
+object->callbackEvent( index );
+}
+void RtApiAsio :: saveDeviceInfo( void )
+{
+devices_.clear();
+unsigned int nDevices = getDeviceCount();
+devices_.resize( nDevices );
+for ( unsigned int i=0; i<nDevices; i++ )
+devices_[i] = getDeviceInfo( i );
+}
+bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+unsigned int firstChannel, unsigned int sampleRate,
+RtAudioFormat format, unsigned int *bufferSize,
+RtAudio::StreamOptions *options )
+{////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
+bool isDuplexInput =  mode == INPUT && stream_.mode == OUTPUT;
+// For ASIO, a duplex stream MUST use the same driver.
+if ( isDuplexInput && stream_.device[0] != device ) {
+errorText_ = "RtApiAsio::probeDeviceOpen: an ASIO duplex stream must use the same device for input and output!";
+return FAILURE;
+}
+char driverName[32];
+ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 );
+if ( result != ASE_OK ) {
+errorStream_ << "RtApiAsio::probeDeviceOpen: unable to get driver name (" << getAsioErrorString( result ) << ").";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+// Only load the driver once for duplex stream.
+if ( !isDuplexInput ) {
+// The getDeviceInfo() function will not work when a stream is open
+// because ASIO does not allow multiple devices to run at the same
+// time.  Thus, we'll probe the system before opening a stream and
+// save the results for use by getDeviceInfo().
+this->saveDeviceInfo();
+if ( !drivers.loadDriver( driverName ) ) {
+errorStream_ << "RtApiAsio::probeDeviceOpen: unable to load driver (" << driverName << ").";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+result = ASIOInit( &driverInfo );
+if ( result != ASE_OK ) {
+errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+}
+// keep them before any "goto error", they are used for error cleanup + goto device boundary checks
+bool buffersAllocated = false;
+AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+unsigned int nChannels;
+// Check the device channel count.
+long inputChannels, outputChannels;
+result = ASIOGetChannels( &inputChannels, &outputChannels );
+if ( result != ASE_OK ) {
+errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
+errorText_ = errorStream_.str();
+goto error;
+}
+if ( ( mode == OUTPUT && (channels+firstChannel) > (unsigned int) outputChannels) ||
+( mode == INPUT && (channels+firstChannel) > (unsigned int) inputChannels) ) {
+errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested channel count (" << channels << ") + offset (" << firstChannel << ").";
+errorText_ = errorStream_.str();
+goto error;
+}
+stream_.nDeviceChannels[mode] = channels;
+stream_.nUserChannels[mode] = channels;
+stream_.channelOffset[mode] = firstChannel;
+// Verify the sample rate is supported.
+result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate );
+if ( result != ASE_OK ) {
+errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested sample rate (" << sampleRate << ").";
+errorText_ = errorStream_.str();
+goto error;
+}
+// Get the current sample rate
+ASIOSampleRate currentRate;
+result = ASIOGetSampleRate( &currentRate );
+if ( result != ASE_OK ) {
+errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error getting sample rate.";
+errorText_ = errorStream_.str();
+goto error;
+}
+// Set the sample rate only if necessary
+if ( currentRate != sampleRate ) {
+result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate );
+if ( result != ASE_OK ) {
+errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ").";
+errorText_ = errorStream_.str();
+goto error;
+}
+}
+// Determine the driver data type.
+ASIOChannelInfo channelInfo;
+channelInfo.channel = 0;
+if ( mode == OUTPUT ) channelInfo.isInput = false;
+else channelInfo.isInput = true;
+result = ASIOGetChannelInfo( &channelInfo );
+if ( result != ASE_OK ) {
+errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting data format.";
+errorText_ = errorStream_.str();
+goto error;
+}
+// Assuming WINDOWS host is always little-endian.
+stream_.doByteSwap[mode] = false;
+stream_.userFormat = format;
+stream_.deviceFormat[mode] = 0;
+if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) {
+stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+if ( channelInfo.type == ASIOSTInt16MSB ) stream_.doByteSwap[mode] = true;
+}
+else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) {
+stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+if ( channelInfo.type == ASIOSTInt32MSB ) stream_.doByteSwap[mode] = true;
+}
+else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) {
+stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
+if ( channelInfo.type == ASIOSTFloat32MSB ) stream_.doByteSwap[mode] = true;
+}
+else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) {
+stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
+if ( channelInfo.type == ASIOSTFloat64MSB ) stream_.doByteSwap[mode] = true;
+}
+else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB ) {
+stream_.deviceFormat[mode] = RTAUDIO_SINT24;
+if ( channelInfo.type == ASIOSTInt24MSB ) stream_.doByteSwap[mode] = true;
+}
+if ( stream_.deviceFormat[mode] == 0 ) {
+errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") data format not supported by RtAudio.";
+errorText_ = errorStream_.str();
+goto error;
+}
+// Set the buffer size.  For a duplex stream, this will end up
+// setting the buffer size based on the input constraints, which
+// should be ok.
+long minSize, maxSize, preferSize, granularity;
+result = ASIOGetBufferSize( &minSize, &maxSize, &preferSize, &granularity );
+if ( result != ASE_OK ) {
+errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting buffer size.";
+errorText_ = errorStream_.str();
+goto error;
+}
+if ( isDuplexInput ) {
+// When this is the duplex input (output was opened before), then we have to use the same
+// buffersize as the output, because it might use the preferred buffer size, which most
+// likely wasn't passed as input to this. The buffer sizes have to be identically anyway,
+// So instead of throwing an error, make them equal. The caller uses the reference
+// to the "bufferSize" param as usual to set up processing buffers.
+*bufferSize = stream_.bufferSize;
+} else {
+if ( *bufferSize == 0 ) *bufferSize = preferSize;
+else if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;
+else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;
+else if ( granularity == -1 ) {
+// Make sure bufferSize is a power of two.
+int log2_of_min_size = 0;
+int log2_of_max_size = 0;
+for ( unsigned int i = 0; i < sizeof(long) * 8; i++ ) {
+if ( minSize & ((long)1 << i) ) log2_of_min_size = i;
+if ( maxSize & ((long)1 << i) ) log2_of_max_size = i;
+}
+long min_delta = std::abs( (long)*bufferSize - ((long)1 << log2_of_min_size) );
+int min_delta_num = log2_of_min_size;
+for (int i = log2_of_min_size + 1; i <= log2_of_max_size; i++) {
+long current_delta = std::abs( (long)*bufferSize - ((long)1 << i) );
+if (current_delta < min_delta) {
+min_delta = current_delta;
+min_delta_num = i;
+}
+}
+*bufferSize = ( (unsigned int)1 << min_delta_num );
+if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;
+else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;
+}
+else if ( granularity != 0 ) {
+// Set to an even multiple of granularity, rounding up.
+*bufferSize = (*bufferSize + granularity-1) / granularity * granularity;
+}
+}
+/*
+// we don't use it anymore, see above!
+// Just left it here for the case...
+if ( isDuplexInput && stream_.bufferSize != *bufferSize ) {
+errorText_ = "RtApiAsio::probeDeviceOpen: input/output buffersize discrepancy!";
+goto error;
+}
+*/
+stream_.bufferSize = *bufferSize;
+stream_.nBuffers = 2;
+if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
+else stream_.userInterleaved = true;
+// ASIO always uses non-interleaved buffers.
+stream_.deviceInterleaved[mode] = false;
+// Allocate, if necessary, our AsioHandle structure for the stream.
+if ( handle == 0 ) {
+try {
+handle = new AsioHandle;
+}
+catch ( std::bad_alloc& ) {
+errorText_ = "RtApiAsio::probeDeviceOpen: error allocating AsioHandle memory.";
+goto error;
+}
+handle->bufferInfos = 0;
+// Create a manual-reset event.
+handle->condition = CreateEvent( NULL,   // no security
+TRUE,   // manual-reset
+FALSE,  // non-signaled initially
+NULL ); // unnamed
+stream_.apiHandle = (void *) handle;
+}
+// Create the ASIO internal buffers.  Since RtAudio sets up input
+// and output separately, we'll have to dispose of previously
+// created output buffers for a duplex stream.
+if ( mode == INPUT && stream_.mode == OUTPUT ) {
+ASIODisposeBuffers();
+if ( handle->bufferInfos ) free( handle->bufferInfos );
+}
+// Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure.
+unsigned int i;
+nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
+handle->bufferInfos = (ASIOBufferInfo *) malloc( nChannels * sizeof(ASIOBufferInfo) );
+if ( handle->bufferInfos == NULL ) {
+errorStream_ << "RtApiAsio::probeDeviceOpen: error allocating bufferInfo memory for driver (" << driverName << ").";
+errorText_ = errorStream_.str();
+goto error;
+}
+ASIOBufferInfo *infos;
+infos = handle->bufferInfos;
+for ( i=0; i<stream_.nDeviceChannels[0]; i++, infos++ ) {
+infos->isInput = ASIOFalse;
+infos->channelNum = i + stream_.channelOffset[0];
+infos->buffers[0] = infos->buffers[1] = 0;
+}
+for ( i=0; i<stream_.nDeviceChannels[1]; i++, infos++ ) {
+infos->isInput = ASIOTrue;
+infos->channelNum = i + stream_.channelOffset[1];
+infos->buffers[0] = infos->buffers[1] = 0;
+}
+// prepare for callbacks
+stream_.sampleRate = sampleRate;
+stream_.device[mode] = device;
+stream_.mode = isDuplexInput ? DUPLEX : mode;
+// store this class instance before registering callbacks, that are going to use it
+asioCallbackInfo = &stream_.callbackInfo;
+stream_.callbackInfo.object = (void *) this;
+// Set up the ASIO callback structure and create the ASIO data buffers.
+asioCallbacks.bufferSwitch = &bufferSwitch;
+asioCallbacks.sampleRateDidChange = &sampleRateChanged;
+asioCallbacks.asioMessage = &asioMessages;
+asioCallbacks.bufferSwitchTimeInfo = NULL;
+result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks );
+if ( result != ASE_OK ) {
+// Standard method failed. This can happen with strict/misbehaving drivers that return valid buffer size ranges
+// but only accept the preferred buffer size as parameter for ASIOCreateBuffers (e.g. Creative's ASIO driver).
+// In that case, let's be naïve and try that instead.
+*bufferSize = preferSize;
+stream_.bufferSize = *bufferSize;
+result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks );
+}
+if ( result != ASE_OK ) {
+errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") creating buffers.";
+errorText_ = errorStream_.str();
+goto error;
+}
+buffersAllocated = true;
+stream_.state = STREAM_STOPPED;
+// Set flags for buffer conversion.
+stream_.doConvertBuffer[mode] = false;
+if ( stream_.userFormat != stream_.deviceFormat[mode] )
+stream_.doConvertBuffer[mode] = true;
+if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+stream_.nUserChannels[mode] > 1 )
+stream_.doConvertBuffer[mode] = true;
+// Allocate necessary internal buffers
+unsigned long bufferBytes;
+bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+if ( stream_.userBuffer[mode] == NULL ) {
+errorText_ = "RtApiAsio::probeDeviceOpen: error allocating user buffer memory.";
+goto error;
+}
+if ( stream_.doConvertBuffer[mode] ) {
+bool makeBuffer = true;
+bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
+if ( isDuplexInput && stream_.deviceBuffer ) {
+unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
+if ( bufferBytes <= bytesOut ) makeBuffer = false;
+}
+if ( makeBuffer ) {
+bufferBytes *= *bufferSize;
+if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
+stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
+if ( stream_.deviceBuffer == NULL ) {
+errorText_ = "RtApiAsio::probeDeviceOpen: error allocating device buffer memory.";
+goto error;
+}
+}
+}
+// Determine device latencies
+long inputLatency, outputLatency;
+result = ASIOGetLatencies( &inputLatency, &outputLatency );
+if ( result != ASE_OK ) {
+errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting latency.";
+errorText_ = errorStream_.str();
+error( RtAudioError::WARNING); // warn but don't fail
+}
+else {
+stream_.latency[0] = outputLatency;
+stream_.latency[1] = inputLatency;
+}
+// Setup the buffer conversion information structure.  We don't use
+// buffers to do channel offsets, so we override that parameter
+// here.
+if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );
+return SUCCESS;
+error:
+if ( !isDuplexInput ) {
+// the cleanup for error in the duplex input, is done by RtApi::openStream
+// So we clean up for single channel only
+if ( buffersAllocated )
+ASIODisposeBuffers();
+drivers.removeCurrentDriver();
+if ( handle ) {
+CloseHandle( handle->condition );
+if ( handle->bufferInfos )
+free( handle->bufferInfos );
+delete handle;
+stream_.apiHandle = 0;
+}
+if ( stream_.userBuffer[mode] ) {
+free( stream_.userBuffer[mode] );
+stream_.userBuffer[mode] = 0;
+}
+if ( stream_.deviceBuffer ) {
+free( stream_.deviceBuffer );
+stream_.deviceBuffer = 0;
+}
+}
+return FAILURE;
+}////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
+void RtApiAsio :: closeStream()
+{
+if ( stream_.state == STREAM_CLOSED ) {
+errorText_ = "RtApiAsio::closeStream(): no open stream to close!";
+error( RtAudioError::WARNING );
+return;
+}
+if ( stream_.state == STREAM_RUNNING ) {
+stream_.state = STREAM_STOPPED;
+ASIOStop();
+}
+ASIODisposeBuffers();
+drivers.removeCurrentDriver();
+AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+if ( handle ) {
+CloseHandle( handle->condition );
+if ( handle->bufferInfos )
+free( handle->bufferInfos );
+delete handle;
+stream_.apiHandle = 0;
+}
+for ( int i=0; i<2; i++ ) {
+if ( stream_.userBuffer[i] ) {
+free( stream_.userBuffer[i] );
+stream_.userBuffer[i] = 0;
+}
+}
+if ( stream_.deviceBuffer ) {
+free( stream_.deviceBuffer );
+stream_.deviceBuffer = 0;
+}
+stream_.mode = UNINITIALIZED;
+stream_.state = STREAM_CLOSED;
+}
+bool stopThreadCalled = false;
+void RtApiAsio :: startStream()
+{
+verifyStream();
+if ( stream_.state == STREAM_RUNNING ) {
+errorText_ = "RtApiAsio::startStream(): the stream is already running!";
+error( RtAudioError::WARNING );
+return;
+}
+#if defined( HAVE_GETTIMEOFDAY )
+gettimeofday( &stream_.lastTickTimestamp, NULL );
+#endif
+AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+ASIOError result = ASIOStart();
+if ( result != ASE_OK ) {
+errorStream_ << "RtApiAsio::startStream: error (" << getAsioErrorString( result ) << ") starting device.";
+errorText_ = errorStream_.str();
+goto unlock;
+}
+handle->drainCounter = 0;
+handle->internalDrain = false;
+ResetEvent( handle->condition );
+stream_.state = STREAM_RUNNING;
+asioXRun = false;
+unlock:
+stopThreadCalled = false;
+if ( result == ASE_OK ) return;
+error( RtAudioError::SYSTEM_ERROR );
+}
+void RtApiAsio :: stopStream()
+{
+verifyStream();
+if ( stream_.state == STREAM_STOPPED ) {
+errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!";
+error( RtAudioError::WARNING );
+return;
+}
+AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+if ( handle->drainCounter == 0 ) {
+handle->drainCounter = 2;
+WaitForSingleObject( handle->condition, INFINITE );  // block until signaled
+}
+}
+stream_.state = STREAM_STOPPED;
+ASIOError result = ASIOStop();
+if ( result != ASE_OK ) {
+errorStream_ << "RtApiAsio::stopStream: error (" << getAsioErrorString( result ) << ") stopping device.";
+errorText_ = errorStream_.str();
+}
+if ( result == ASE_OK ) return;
+error( RtAudioError::SYSTEM_ERROR );
+}
+void RtApiAsio :: abortStream()
+{
+verifyStream();
+if ( stream_.state == STREAM_STOPPED ) {
+errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!";
+error( RtAudioError::WARNING );
+return;
+}
+// The following lines were commented-out because some behavior was
+// noted where the device buffers need to be zeroed to avoid
+// continuing sound, even when the device buffers are completely
+// disposed.  So now, calling abort is the same as calling stop.
+// AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+// handle->drainCounter = 2;
+stopStream();
+}
+// This function will be called by a spawned thread when the user
+// callback function signals that the stream should be stopped or
+// aborted.  It is necessary to handle it this way because the
+// callbackEvent() function must return before the ASIOStop()
+// function will return.
+static unsigned __stdcall asioStopStream( void *ptr )
+{
+CallbackInfo *info = (CallbackInfo *) ptr;
+RtApiAsio *object = (RtApiAsio *) info->object;
+object->stopStream();
+_endthreadex( 0 );
+return 0;
+}
+bool RtApiAsio :: callbackEvent( long bufferIndex )
+{
+if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;
+if ( stream_.state == STREAM_CLOSED ) {
+errorText_ = "RtApiAsio::callbackEvent(): the stream is closed ... this shouldn't happen!";
+error( RtAudioError::WARNING );
+return FAILURE;
+}
+CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
+AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+// Check if we were draining the stream and signal if finished.
+if ( handle->drainCounter > 3 ) {
+stream_.state = STREAM_STOPPING;
+if ( handle->internalDrain == false )
+SetEvent( handle->condition );
+else { // spawn a thread to stop the stream
+unsigned threadId;
+stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream,
+&stream_.callbackInfo, 0, &threadId );
+}
+return SUCCESS;
+}
+// Invoke user callback to get fresh output data UNLESS we are
+// draining stream.
+if ( handle->drainCounter == 0 ) {
+RtAudioCallback callback = (RtAudioCallback) info->callback;
+double streamTime = getStreamTime();
+RtAudioStreamStatus status = 0;
+if ( stream_.mode != INPUT && asioXRun == true ) {
+status |= RTAUDIO_OUTPUT_UNDERFLOW;
+asioXRun = false;
+}
+if ( stream_.mode != OUTPUT && asioXRun == true ) {
+status |= RTAUDIO_INPUT_OVERFLOW;
+asioXRun = false;
+}
+int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+stream_.bufferSize, streamTime, status, info->userData );
+if ( cbReturnValue == 2 ) {
+stream_.state = STREAM_STOPPING;
+handle->drainCounter = 2;
+unsigned threadId;
+stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream,
+&stream_.callbackInfo, 0, &threadId );
+return SUCCESS;
+}
+else if ( cbReturnValue == 1 ) {
+handle->drainCounter = 1;
+handle->internalDrain = true;
+}
+}
+unsigned int nChannels, bufferBytes, i, j;
+nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
+if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+bufferBytes = stream_.bufferSize * formatBytes( stream_.deviceFormat[0] );
+if ( handle->drainCounter > 1 ) { // write zeros to the output stream
+for ( i=0, j=0; i<nChannels; i++ ) {
+if ( handle->bufferInfos[i].isInput != ASIOTrue )
+memset( handle->bufferInfos[i].buffers[bufferIndex], 0, bufferBytes );
+}
+}
+else if ( stream_.doConvertBuffer[0] ) {
+convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
+if ( stream_.doByteSwap[0] )
+byteSwapBuffer( stream_.deviceBuffer,
+stream_.bufferSize * stream_.nDeviceChannels[0],
+stream_.deviceFormat[0] );
+for ( i=0, j=0; i<nChannels; i++ ) {
+if ( handle->bufferInfos[i].isInput != ASIOTrue )
+memcpy( handle->bufferInfos[i].buffers[bufferIndex],
+&stream_.deviceBuffer[j++*bufferBytes], bufferBytes );
+}
+}
+else {
+if ( stream_.doByteSwap[0] )
+byteSwapBuffer( stream_.userBuffer[0],
+stream_.bufferSize * stream_.nUserChannels[0],
+stream_.userFormat );
+for ( i=0, j=0; i<nChannels; i++ ) {
+if ( handle->bufferInfos[i].isInput != ASIOTrue )
+memcpy( handle->bufferInfos[i].buffers[bufferIndex],
+&stream_.userBuffer[0][bufferBytes*j++], bufferBytes );
+}
+}
+}
+// Don't bother draining input
+if ( handle->drainCounter ) {
+handle->drainCounter++;
+goto unlock;
+}
+if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+bufferBytes = stream_.bufferSize * formatBytes(stream_.deviceFormat[1]);
+if (stream_.doConvertBuffer[1]) {
+// Always interleave ASIO input data.
+for ( i=0, j=0; i<nChannels; i++ ) {
+if ( handle->bufferInfos[i].isInput == ASIOTrue )
+memcpy( &stream_.deviceBuffer[j++*bufferBytes],
+handle->bufferInfos[i].buffers[bufferIndex],
+bufferBytes );
+}
+if ( stream_.doByteSwap[1] )
+byteSwapBuffer( stream_.deviceBuffer,
+stream_.bufferSize * stream_.nDeviceChannels[1],
+stream_.deviceFormat[1] );
+convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
+}
+else {
+for ( i=0, j=0; i<nChannels; i++ ) {
+if ( handle->bufferInfos[i].isInput == ASIOTrue ) {
+memcpy( &stream_.userBuffer[1][bufferBytes*j++],
+handle->bufferInfos[i].buffers[bufferIndex],
+bufferBytes );
+}
+}
+if ( stream_.doByteSwap[1] )
+byteSwapBuffer( stream_.userBuffer[1],
+stream_.bufferSize * stream_.nUserChannels[1],
+stream_.userFormat );
+}
+}
+unlock:
+// The following call was suggested by Malte Clasen.  While the API
+// documentation indicates it should not be required, some device
+// drivers apparently do not function correctly without it.
+ASIOOutputReady();
+RtApi::tickStreamTime();
+return SUCCESS;
+}
+static void sampleRateChanged( ASIOSampleRate sRate )
+{
+// The ASIO documentation says that this usually only happens during
+// external sync.  Audio processing is not stopped by the driver,
+// actual sample rate might not have even changed, maybe only the
+// sample rate status of an AES/EBU or S/PDIF digital input at the
+// audio device.
+RtApi *object = (RtApi *) asioCallbackInfo->object;
+try {
+object->stopStream();
+}
+catch ( RtAudioError &exception ) {
+std::cerr << "\nRtApiAsio: sampleRateChanged() error (" << exception.getMessage() << ")!\n" << std::endl;
+return;
+}
+std::cerr << "\nRtApiAsio: driver reports sample rate changed to " << sRate << " ... stream stopped!!!\n" << std::endl;
+}
+static long asioMessages( long selector, long value, void* /*message*/, double* /*opt*/ )
+{
+long ret = 0;
+switch( selector ) {
+case kAsioSelectorSupported:
+if ( value == kAsioResetRequest
+|| value == kAsioEngineVersion
+|| value == kAsioResyncRequest
+|| value == kAsioLatenciesChanged
+// The following three were added for ASIO 2.0, you don't
+// necessarily have to support them.
+|| value == kAsioSupportsTimeInfo
+|| value == kAsioSupportsTimeCode
+|| value == kAsioSupportsInputMonitor)
+ret = 1L;
+break;
+case kAsioResetRequest:
+// Defer the task and perform the reset of the driver during the
+// next "safe" situation.  You cannot reset the driver right now,
+// as this code is called from the driver.  Reset the driver is
+// done by completely destruct is. I.e. ASIOStop(),
+// ASIODisposeBuffers(), Destruction Afterwards you initialize the
+// driver again.
+std::cerr << "\nRtApiAsio: driver reset requested!!!" << std::endl;
+ret = 1L;
+break;
+case kAsioResyncRequest:
+// This informs the application that the driver encountered some
+// non-fatal data loss.  It is used for synchronization purposes
+// of different media.  Added mainly to work around the Win16Mutex
+// problems in Windows 95/98 with the Windows Multimedia system,
+// which could lose data because the Mutex was held too long by
+// another thread.  However a driver can issue it in other
+// situations, too.
+// std::cerr << "\nRtApiAsio: driver resync requested!!!" << std::endl;
+asioXRun = true;
+ret = 1L;
+break;
+case kAsioLatenciesChanged:
+// This will inform the host application that the drivers were
+// latencies changed.  Beware, it this does not mean that the
+// buffer sizes have changed!  You might need to update internal
+// delay data.
+std::cerr << "\nRtApiAsio: driver latency may have changed!!!" << std::endl;
+ret = 1L;
+break;
+case kAsioEngineVersion:
+// Return the supported ASIO version of the host application.  If
+// a host application does not implement this selector, ASIO 1.0
+// is assumed by the driver.
+ret = 2L;
+break;
+case kAsioSupportsTimeInfo:
+// Informs the driver whether the
+// asioCallbacks.bufferSwitchTimeInfo() callback is supported.
+// For compatibility with ASIO 1.0 drivers the host application
+// should always support the "old" bufferSwitch method, too.
+ret = 0;
+break;
+case kAsioSupportsTimeCode:
+// Informs the driver whether application is interested in time
+// code info.  If an application does not need to know about time
+// code, the driver has less work to do.
+ret = 0;
+break;
+}
+return ret;
+}
+static const char* getAsioErrorString( ASIOError result )
+{
+struct Messages
+{
+ASIOError value;
+const char*message;
+};
+static const Messages m[] =
+{
+{   ASE_NotPresent,    "Hardware input or output is not present or available." },
+{   ASE_HWMalfunction,  "Hardware is malfunctioning." },
+{   ASE_InvalidParameter, "Invalid input parameter." },
+{   ASE_InvalidMode,      "Invalid mode." },
+{   ASE_SPNotAdvancing,     "Sample position not advancing." },
+{   ASE_NoClock,            "Sample clock or rate cannot be determined or is not present." },
+{   ASE_NoMemory,           "Not enough memory to complete the request." }
+};
+for ( unsigned int i = 0; i < sizeof(m)/sizeof(m[0]); ++i )
+if ( m[i].value == result ) return m[i].message;
+return "Unknown error.";
+}
+//******************** End of __WINDOWS_ASIO__ *********************//
+#endif
+#if defined(__WINDOWS_WASAPI__) // Windows WASAPI API
+// Authored by Marcus Tomlinson <themarcustomlinson@gmail.com>, April 2014
+// - Introduces support for the Windows WASAPI API
+// - Aims to deliver bit streams to and from hardware at the lowest possible latency, via the absolute minimum buffer sizes required
+// - Provides flexible stream configuration to an otherwise strict and inflexible WASAPI interface
+// - Includes automatic internal conversion of sample rate and buffer size between hardware and the user
+#ifndef INITGUID
+#define INITGUID
+#endif
+#include <mfapi.h>
+#include <mferror.h>
+#include <mfplay.h>
+#include <mftransform.h>
+#include <wmcodecdsp.h>
+#include <audioclient.h>
+#include <avrt.h>
+#include <mmdeviceapi.h>
+#include <functiondiscoverykeys_devpkey.h>
+#ifndef MF_E_TRANSFORM_NEED_MORE_INPUT
+#define MF_E_TRANSFORM_NEED_MORE_INPUT _HRESULT_TYPEDEF_(0xc00d6d72)
+#endif
+#ifndef MFSTARTUP_NOSOCKET
+#define MFSTARTUP_NOSOCKET 0x1
+#endif
+#ifdef _MSC_VER
+#pragma comment( lib, "ksuser" )
+#pragma comment( lib, "mfplat.lib" )
+#pragma comment( lib, "mfuuid.lib" )
+#pragma comment( lib, "wmcodecdspuuid" )
+#endif
+//=============================================================================
+#define SAFE_RELEASE( objectPtr )\
+if ( objectPtr )\
+{\
+objectPtr->Release();\
+objectPtr = NULL;\
+}
+typedef HANDLE ( __stdcall *TAvSetMmThreadCharacteristicsPtr )( LPCWSTR TaskName, LPDWORD TaskIndex );
+//-----------------------------------------------------------------------------
+// WASAPI dictates stream sample rate, format, channel count, and in some cases, buffer size.
+// Therefore we must perform all necessary conversions to user buffers in order to satisfy these
+// requirements. WasapiBuffer ring buffers are used between HwIn->UserIn and UserOut->HwOut to
+// provide intermediate storage for read / write synchronization.
+class WasapiBuffer
+{
+public:
+WasapiBuffer()
+: buffer_( NULL ),
+bufferSize_( 0 ),
+inIndex_( 0 ),
+outIndex_( 0 ) {}
+~WasapiBuffer() {
+free( buffer_ );
+}
+// sets the length of the internal ring buffer
+void setBufferSize( unsigned int bufferSize, unsigned int formatBytes ) {
+free( buffer_ );
+buffer_ = ( char* ) calloc( bufferSize, formatBytes );
+bufferSize_ = bufferSize;
+inIndex_ = 0;
+outIndex_ = 0;
+}
+// attempt to push a buffer into the ring buffer at the current "in" index
+bool pushBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format )
+{
+if ( !buffer ||                 // incoming buffer is NULL
+bufferSize == 0 ||         // incoming buffer has no data
+bufferSize > bufferSize_ ) // incoming buffer too large
+{
+return false;
+}
+unsigned int relOutIndex = outIndex_;
+unsigned int inIndexEnd = inIndex_ + bufferSize;
+if ( relOutIndex < inIndex_ && inIndexEnd >= bufferSize_ ) {
+relOutIndex += bufferSize_;
+}
+// the "IN" index CAN BEGIN at the "OUT" index
+// the "IN" index CANNOT END at the "OUT" index
+if ( inIndex_ < relOutIndex && inIndexEnd >= relOutIndex ) {
+return false; // not enough space between "in" index and "out" index
+}
+// copy buffer from external to internal
+int fromZeroSize = inIndex_ + bufferSize - bufferSize_;
+fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize;
+int fromInSize = bufferSize - fromZeroSize;
+switch( format )
+{
+case RTAUDIO_SINT8:
+memcpy( &( ( char* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( char ) );
+memcpy( buffer_, &( ( char* ) buffer )[fromInSize], fromZeroSize * sizeof( char ) );
+break;
+case RTAUDIO_SINT16:
+memcpy( &( ( short* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( short ) );
+memcpy( buffer_, &( ( short* ) buffer )[fromInSize], fromZeroSize * sizeof( short ) );
+break;
+case RTAUDIO_SINT24:
+memcpy( &( ( S24* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( S24 ) );
+memcpy( buffer_, &( ( S24* ) buffer )[fromInSize], fromZeroSize * sizeof( S24 ) );
+break;
+case RTAUDIO_SINT32:
+memcpy( &( ( int* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( int ) );
+memcpy( buffer_, &( ( int* ) buffer )[fromInSize], fromZeroSize * sizeof( int ) );
+break;
+case RTAUDIO_FLOAT32:
+memcpy( &( ( float* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( float ) );
+memcpy( buffer_, &( ( float* ) buffer )[fromInSize], fromZeroSize * sizeof( float ) );
+break;
+case RTAUDIO_FLOAT64:
+memcpy( &( ( double* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( double ) );
+memcpy( buffer_, &( ( double* ) buffer )[fromInSize], fromZeroSize * sizeof( double ) );
+break;
+}
+// update "in" index
+inIndex_ += bufferSize;
+inIndex_ %= bufferSize_;
+return true;
+}
+// attempt to pull a buffer from the ring buffer from the current "out" index
+bool pullBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format )
+{
+if ( !buffer ||                 // incoming buffer is NULL
+bufferSize == 0 ||         // incoming buffer has no data
+bufferSize > bufferSize_ ) // incoming buffer too large
+{
+return false;
+}
+unsigned int relInIndex = inIndex_;
+unsigned int outIndexEnd = outIndex_ + bufferSize;
+if ( relInIndex < outIndex_ && outIndexEnd >= bufferSize_ ) {
+relInIndex += bufferSize_;
+}
+// the "OUT" index CANNOT BEGIN at the "IN" index
+// the "OUT" index CAN END at the "IN" index
+if ( outIndex_ <= relInIndex && outIndexEnd > relInIndex ) {
+return false; // not enough space between "out" index and "in" index
+}
+// copy buffer from internal to external
+int fromZeroSize = outIndex_ + bufferSize - bufferSize_;
+fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize;
+int fromOutSize = bufferSize - fromZeroSize;
+switch( format )
+{
+case RTAUDIO_SINT8:
+memcpy( buffer, &( ( char* ) buffer_ )[outIndex_], fromOutSize * sizeof( char ) );
+memcpy( &( ( char* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( char ) );
+break;
+case RTAUDIO_SINT16:
+memcpy( buffer, &( ( short* ) buffer_ )[outIndex_], fromOutSize * sizeof( short ) );
+memcpy( &( ( short* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( short ) );
+break;
+case RTAUDIO_SINT24:
+memcpy( buffer, &( ( S24* ) buffer_ )[outIndex_], fromOutSize * sizeof( S24 ) );
+memcpy( &( ( S24* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( S24 ) );
+break;
+case RTAUDIO_SINT32:
+memcpy( buffer, &( ( int* ) buffer_ )[outIndex_], fromOutSize * sizeof( int ) );
+memcpy( &( ( int* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( int ) );
+break;
+case RTAUDIO_FLOAT32:
+memcpy( buffer, &( ( float* ) buffer_ )[outIndex_], fromOutSize * sizeof( float ) );
+memcpy( &( ( float* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( float ) );
+break;
+case RTAUDIO_FLOAT64:
+memcpy( buffer, &( ( double* ) buffer_ )[outIndex_], fromOutSize * sizeof( double ) );
+memcpy( &( ( double* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( double ) );
+break;
+}
+// update "out" index
+outIndex_ += bufferSize;
+outIndex_ %= bufferSize_;
+return true;
+}
+private:
+char* buffer_;
+unsigned int bufferSize_;
+unsigned int inIndex_;
+unsigned int outIndex_;
+};
+//-----------------------------------------------------------------------------
+// In order to satisfy WASAPI's buffer requirements, we need a means of converting sample rate
+// between HW and the user. The WasapiResampler class is used to perform this conversion between
+// HwIn->UserIn and UserOut->HwOut during the stream callback loop.
+class WasapiResampler
+{
+public:
+WasapiResampler( bool isFloat, unsigned int bitsPerSample, unsigned int channelCount,
+unsigned int inSampleRate, unsigned int outSampleRate )
+: _bytesPerSample( bitsPerSample / 8 )
+, _channelCount( channelCount )
+, _sampleRatio( ( float ) outSampleRate / inSampleRate )
+, _transformUnk( NULL )
+, _transform( NULL )
+, _mediaType( NULL )
+, _inputMediaType( NULL )
+, _outputMediaType( NULL )
+#ifdef __IWMResamplerProps_FWD_DEFINED__
+, _resamplerProps( NULL )
+#endif
+{
+// 1. Initialization
+MFStartup( MF_VERSION, MFSTARTUP_NOSOCKET );
+// 2. Create Resampler Transform Object
+CoCreateInstance( CLSID_CResamplerMediaObject, NULL, CLSCTX_INPROC_SERVER,
+IID_IUnknown, ( void** ) &_transformUnk );
+_transformUnk->QueryInterface( IID_PPV_ARGS( &_transform ) );
+#ifdef __IWMResamplerProps_FWD_DEFINED__
+_transformUnk->QueryInterface( IID_PPV_ARGS( &_resamplerProps ) );
+_resamplerProps->SetHalfFilterLength( 60 ); // best conversion quality
+#endif
+// 3. Specify input / output format
+MFCreateMediaType( &_mediaType );
+_mediaType->SetGUID( MF_MT_MAJOR_TYPE, MFMediaType_Audio );
+_mediaType->SetGUID( MF_MT_SUBTYPE, isFloat ? MFAudioFormat_Float : MFAudioFormat_PCM );
+_mediaType->SetUINT32( MF_MT_AUDIO_NUM_CHANNELS, channelCount );
+_mediaType->SetUINT32( MF_MT_AUDIO_SAMPLES_PER_SECOND, inSampleRate );
+_mediaType->SetUINT32( MF_MT_AUDIO_BLOCK_ALIGNMENT, _bytesPerSample * channelCount );
+_mediaType->SetUINT32( MF_MT_AUDIO_AVG_BYTES_PER_SECOND, _bytesPerSample * channelCount * inSampleRate );
+_mediaType->SetUINT32( MF_MT_AUDIO_BITS_PER_SAMPLE, bitsPerSample );
+_mediaType->SetUINT32( MF_MT_ALL_SAMPLES_INDEPENDENT, TRUE );
+MFCreateMediaType( &_inputMediaType );
+_mediaType->CopyAllItems( _inputMediaType );
+_transform->SetInputType( 0, _inputMediaType, 0 );
+MFCreateMediaType( &_outputMediaType );
+_mediaType->CopyAllItems( _outputMediaType );
+_outputMediaType->SetUINT32( MF_MT_AUDIO_SAMPLES_PER_SECOND, outSampleRate );
+_outputMediaType->SetUINT32( MF_MT_AUDIO_AVG_BYTES_PER_SECOND, _bytesPerSample * channelCount * outSampleRate );
+_transform->SetOutputType( 0, _outputMediaType, 0 );
+// 4. Send stream start messages to Resampler
+_transform->ProcessMessage( MFT_MESSAGE_COMMAND_FLUSH, 0 );
+_transform->ProcessMessage( MFT_MESSAGE_NOTIFY_BEGIN_STREAMING, 0 );
+_transform->ProcessMessage( MFT_MESSAGE_NOTIFY_START_OF_STREAM, 0 );
+}
+~WasapiResampler()
+{
+// 8. Send stream stop messages to Resampler
+_transform->ProcessMessage( MFT_MESSAGE_NOTIFY_END_OF_STREAM, 0 );
+_transform->ProcessMessage( MFT_MESSAGE_NOTIFY_END_STREAMING, 0 );
+// 9. Cleanup
+MFShutdown();
+SAFE_RELEASE( _transformUnk );
+SAFE_RELEASE( _transform );
+SAFE_RELEASE( _mediaType );
+SAFE_RELEASE( _inputMediaType );
+SAFE_RELEASE( _outputMediaType );
+#ifdef __IWMResamplerProps_FWD_DEFINED__
+SAFE_RELEASE( _resamplerProps );
+#endif
+}
+void Convert( char* outBuffer, const char* inBuffer, unsigned int inSampleCount, unsigned int& outSampleCount, int maxOutSampleCount = -1 )
+{
+unsigned int inputBufferSize = _bytesPerSample * _channelCount * inSampleCount;
+if ( _sampleRatio == 1 )
+{
+// no sample rate conversion required
+memcpy( outBuffer, inBuffer, inputBufferSize );
+outSampleCount = inSampleCount;
+return;
+}
+unsigned int outputBufferSize = 0;
+if ( maxOutSampleCount != -1 )
+{
+outputBufferSize = _bytesPerSample * _channelCount * maxOutSampleCount;
+}
+else
+{
+outputBufferSize = ( unsigned int ) ceilf( inputBufferSize * _sampleRatio ) + ( _bytesPerSample * _channelCount );
+}
+IMFMediaBuffer* rInBuffer;
+IMFSample* rInSample;
+BYTE* rInByteBuffer = NULL;
+// 5. Create Sample object from input data
+MFCreateMemoryBuffer( inputBufferSize, &rInBuffer );
+rInBuffer->Lock( &rInByteBuffer, NULL, NULL );
+memcpy( rInByteBuffer, inBuffer, inputBufferSize );
+rInBuffer->Unlock();
+rInByteBuffer = NULL;
+rInBuffer->SetCurrentLength( inputBufferSize );
+MFCreateSample( &rInSample );
+rInSample->AddBuffer( rInBuffer );
+// 6. Pass input data to Resampler
+_transform->ProcessInput( 0, rInSample, 0 );
+SAFE_RELEASE( rInBuffer );
+SAFE_RELEASE( rInSample );
+// 7. Perform sample rate conversion
+IMFMediaBuffer* rOutBuffer = NULL;
+BYTE* rOutByteBuffer = NULL;
+MFT_OUTPUT_DATA_BUFFER rOutDataBuffer;
+DWORD rStatus;
+DWORD rBytes = outputBufferSize; // maximum bytes accepted per ProcessOutput
+// 7.1 Create Sample object for output data
+memset( &rOutDataBuffer, 0, sizeof rOutDataBuffer );
+MFCreateSample( &( rOutDataBuffer.pSample ) );
+MFCreateMemoryBuffer( rBytes, &rOutBuffer );
+rOutDataBuffer.pSample->AddBuffer( rOutBuffer );
+rOutDataBuffer.dwStreamID = 0;
+rOutDataBuffer.dwStatus = 0;
+rOutDataBuffer.pEvents = NULL;
+// 7.2 Get output data from Resampler
+if ( _transform->ProcessOutput( 0, 1, &rOutDataBuffer, &rStatus ) == MF_E_TRANSFORM_NEED_MORE_INPUT )
+{
+outSampleCount = 0;
+SAFE_RELEASE( rOutBuffer );
+SAFE_RELEASE( rOutDataBuffer.pSample );
+return;
+}
+// 7.3 Write output data to outBuffer
+SAFE_RELEASE( rOutBuffer );
+rOutDataBuffer.pSample->ConvertToContiguousBuffer( &rOutBuffer );
+rOutBuffer->GetCurrentLength( &rBytes );
+rOutBuffer->Lock( &rOutByteBuffer, NULL, NULL );
+memcpy( outBuffer, rOutByteBuffer, rBytes );
+rOutBuffer->Unlock();
+rOutByteBuffer = NULL;
+outSampleCount = rBytes / _bytesPerSample / _channelCount;
+SAFE_RELEASE( rOutBuffer );
+SAFE_RELEASE( rOutDataBuffer.pSample );
+}
+private:
+unsigned int _bytesPerSample;
+unsigned int _channelCount;
+float _sampleRatio;
+IUnknown* _transformUnk;
+IMFTransform* _transform;
+IMFMediaType* _mediaType;
+IMFMediaType* _inputMediaType;
+IMFMediaType* _outputMediaType;
+#ifdef __IWMResamplerProps_FWD_DEFINED__
+IWMResamplerProps* _resamplerProps;
+#endif
+};
+//-----------------------------------------------------------------------------
+// A structure to hold various information related to the WASAPI implementation.
+struct WasapiHandle
+{
+IAudioClient* captureAudioClient;
+IAudioClient* renderAudioClient;
+IAudioCaptureClient* captureClient;
+IAudioRenderClient* renderClient;
+HANDLE captureEvent;
+HANDLE renderEvent;
+WasapiHandle()
+: captureAudioClient( NULL ),
+renderAudioClient( NULL ),
+captureClient( NULL ),
+renderClient( NULL ),
+captureEvent( NULL ),
+renderEvent( NULL ) {}
+};
+//=============================================================================
+RtApiWasapi::RtApiWasapi()
+: coInitialized_( false ), deviceEnumerator_( NULL )
+{
+// WASAPI can run either apartment or multi-threaded
+HRESULT hr = CoInitialize( NULL );
+if ( !FAILED( hr ) )
+coInitialized_ = true;
+// Instantiate device enumerator
+hr = CoCreateInstance( __uuidof( MMDeviceEnumerator ), NULL,
+CLSCTX_ALL, __uuidof( IMMDeviceEnumerator ),
+( void** ) &deviceEnumerator_ );
+// If this runs on an old Windows, it will fail. Ignore and proceed.
+if ( FAILED( hr ) )
+deviceEnumerator_ = NULL;
+}
+//-----------------------------------------------------------------------------
+RtApiWasapi::~RtApiWasapi()
+{
+if ( stream_.state != STREAM_CLOSED )
+closeStream();
+SAFE_RELEASE( deviceEnumerator_ );
+// If this object previously called CoInitialize()
+if ( coInitialized_ )
+CoUninitialize();
+}
+//=============================================================================
+unsigned int RtApiWasapi::getDeviceCount( void )
+{
+unsigned int captureDeviceCount = 0;
+unsigned int renderDeviceCount = 0;
+IMMDeviceCollection* captureDevices = NULL;
+IMMDeviceCollection* renderDevices = NULL;
+if ( !deviceEnumerator_ )
+return 0;
+// Count capture devices
+errorText_.clear();
+HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
+if ( FAILED( hr ) ) {
+errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device collection.";
+goto Exit;
+}
+hr = captureDevices->GetCount( &captureDeviceCount );
+if ( FAILED( hr ) ) {
+errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device count.";
+goto Exit;
+}
+// Count render devices
+hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
+if ( FAILED( hr ) ) {
+errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device collection.";
+goto Exit;
+}
+hr = renderDevices->GetCount( &renderDeviceCount );
+if ( FAILED( hr ) ) {
+errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device count.";
+goto Exit;
+}
+Exit:
+// release all references
+SAFE_RELEASE( captureDevices );
+SAFE_RELEASE( renderDevices );
+if ( errorText_.empty() )
+return captureDeviceCount + renderDeviceCount;
+error( RtAudioError::DRIVER_ERROR );
+return 0;
+}
+//-----------------------------------------------------------------------------
+RtAudio::DeviceInfo RtApiWasapi::getDeviceInfo( unsigned int device )
+{
+RtAudio::DeviceInfo info;
+unsigned int captureDeviceCount = 0;
+unsigned int renderDeviceCount = 0;
+std::string defaultDeviceName;
+bool isCaptureDevice = false;
+PROPVARIANT deviceNameProp;
+PROPVARIANT defaultDeviceNameProp;
+IMMDeviceCollection* captureDevices = NULL;
+IMMDeviceCollection* renderDevices = NULL;
+IMMDevice* devicePtr = NULL;
+IMMDevice* defaultDevicePtr = NULL;
+IAudioClient* audioClient = NULL;
+IPropertyStore* devicePropStore = NULL;
+IPropertyStore* defaultDevicePropStore = NULL;
+WAVEFORMATEX* deviceFormat = NULL;
+WAVEFORMATEX* closestMatchFormat = NULL;
+// probed
+info.probed = false;
+// Count capture devices
+errorText_.clear();
+RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
+HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
+if ( FAILED( hr ) ) {
+errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device collection.";
+goto Exit;
+}
+hr = captureDevices->GetCount( &captureDeviceCount );
+if ( FAILED( hr ) ) {
+errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device count.";
+goto Exit;
+}
+// Count render devices
+hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
+if ( FAILED( hr ) ) {
+errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device collection.";
+goto Exit;
+}
+hr = renderDevices->GetCount( &renderDeviceCount );
+if ( FAILED( hr ) ) {
+errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device count.";
+goto Exit;
+}
+// validate device index
+if ( device >= captureDeviceCount + renderDeviceCount ) {
+errorText_ = "RtApiWasapi::getDeviceInfo: Invalid device index.";
+errorType = RtAudioError::INVALID_USE;
+goto Exit;
+}
+// determine whether index falls within capture or render devices
+if ( device >= renderDeviceCount ) {
+hr = captureDevices->Item( device - renderDeviceCount, &devicePtr );
+if ( FAILED( hr ) ) {
+errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device handle.";
+goto Exit;
+}
+isCaptureDevice = true;
+}
+else {
+hr = renderDevices->Item( device, &devicePtr );
+if ( FAILED( hr ) ) {
+errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device handle.";
+goto Exit;
+}
+isCaptureDevice = false;
+}
+// get default device name
+if ( isCaptureDevice ) {
+hr = deviceEnumerator_->GetDefaultAudioEndpoint( eCapture, eConsole, &defaultDevicePtr );
+if ( FAILED( hr ) ) {
+errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default capture device handle.";
+goto Exit;
+}
+}
+else {
+hr = deviceEnumerator_->GetDefaultAudioEndpoint( eRender, eConsole, &defaultDevicePtr );
+if ( FAILED( hr ) ) {
+errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default render device handle.";
+goto Exit;
+}
+}
+hr = defaultDevicePtr->OpenPropertyStore( STGM_READ, &defaultDevicePropStore );
+if ( FAILED( hr ) ) {
+errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open default device property store.";
+goto Exit;
+}
+PropVariantInit( &defaultDeviceNameProp );
+hr = defaultDevicePropStore->GetValue( PKEY_Device_FriendlyName, &defaultDeviceNameProp );
+if ( FAILED( hr ) ) {
+errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default device property: PKEY_Device_FriendlyName.";
+goto Exit;
+}
+defaultDeviceName = convertCharPointerToStdString(defaultDeviceNameProp.pwszVal);
+// name
+hr = devicePtr->OpenPropertyStore( STGM_READ, &devicePropStore );
+if ( FAILED( hr ) ) {
+errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open device property store.";
+goto Exit;
+}
+PropVariantInit( &deviceNameProp );
+hr = devicePropStore->GetValue( PKEY_Device_FriendlyName, &deviceNameProp );
+if ( FAILED( hr ) ) {
+errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device property: PKEY_Device_FriendlyName.";
+goto Exit;
+}
+info.name =convertCharPointerToStdString(deviceNameProp.pwszVal);
+// is default
+if ( isCaptureDevice ) {
+info.isDefaultInput = info.name == defaultDeviceName;
+info.isDefaultOutput = false;
+}
+else {
+info.isDefaultInput = false;
+info.isDefaultOutput = info.name == defaultDeviceName;
+}
+// channel count
+hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, NULL, ( void** ) &audioClient );
+if ( FAILED( hr ) ) {
+errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device audio client.";
+goto Exit;
+}
+hr = audioClient->GetMixFormat( &deviceFormat );
+if ( FAILED( hr ) ) {
+errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device mix format.";
+goto Exit;
+}
+if ( isCaptureDevice ) {
+info.inputChannels = deviceFormat->nChannels;
+info.outputChannels = 0;
+info.duplexChannels = 0;
+}
+else {
+info.inputChannels = 0;
+info.outputChannels = deviceFormat->nChannels;
+info.duplexChannels = 0;
+}
+// sample rates
+info.sampleRates.clear();
+// allow support for all sample rates as we have a built-in sample rate converter
+for ( unsigned int i = 0; i < MAX_SAMPLE_RATES; i++ ) {
+info.sampleRates.push_back( SAMPLE_RATES[i] );
+}
+info.preferredSampleRate = deviceFormat->nSamplesPerSec;
+// native format
+info.nativeFormats = 0;
+if ( deviceFormat->wFormatTag == WAVE_FORMAT_IEEE_FLOAT ||
+( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
+( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_IEEE_FLOAT ) )
+{
+if ( deviceFormat->wBitsPerSample == 32 ) {
+info.nativeFormats |= RTAUDIO_FLOAT32;
+}
+else if ( deviceFormat->wBitsPerSample == 64 ) {
+info.nativeFormats |= RTAUDIO_FLOAT64;
+}
+}
+else if ( deviceFormat->wFormatTag == WAVE_FORMAT_PCM ||
+( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
+( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_PCM ) )
+{
+if ( deviceFormat->wBitsPerSample == 8 ) {
+info.nativeFormats |= RTAUDIO_SINT8;
+}
+else if ( deviceFormat->wBitsPerSample == 16 ) {
+info.nativeFormats |= RTAUDIO_SINT16;
+}
+else if ( deviceFormat->wBitsPerSample == 24 ) {
+info.nativeFormats |= RTAUDIO_SINT24;
+}
+else if ( deviceFormat->wBitsPerSample == 32 ) {
+info.nativeFormats |= RTAUDIO_SINT32;
+}
+}
+// probed
+info.probed = true;
+Exit:
+// release all references
+PropVariantClear( &deviceNameProp );
+PropVariantClear( &defaultDeviceNameProp );
+SAFE_RELEASE( captureDevices );
+SAFE_RELEASE( renderDevices );
+SAFE_RELEASE( devicePtr );
+SAFE_RELEASE( defaultDevicePtr );
+SAFE_RELEASE( audioClient );
+SAFE_RELEASE( devicePropStore );
+SAFE_RELEASE( defaultDevicePropStore );
+CoTaskMemFree( deviceFormat );
+CoTaskMemFree( closestMatchFormat );
+if ( !errorText_.empty() )
+error( errorType );
+return info;
+}
+//-----------------------------------------------------------------------------
+unsigned int RtApiWasapi::getDefaultOutputDevice( void )
+{
+for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {
+if ( getDeviceInfo( i ).isDefaultOutput ) {
+return i;
+}
+}
+return 0;
+}
+//-----------------------------------------------------------------------------
+unsigned int RtApiWasapi::getDefaultInputDevice( void )
+{
+for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {
+if ( getDeviceInfo( i ).isDefaultInput ) {
+return i;
+}
+}
+return 0;
+}
+//-----------------------------------------------------------------------------
+void RtApiWasapi::closeStream( void )
+{
+if ( stream_.state == STREAM_CLOSED ) {
+errorText_ = "RtApiWasapi::closeStream: No open stream to close.";
+error( RtAudioError::WARNING );
+return;
+}
+if ( stream_.state != STREAM_STOPPED )
+stopStream();
+// clean up stream memory
+SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient )
+SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient )
+SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureClient )
+SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderClient )
+if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent )
+CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent );
+if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent )
+CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent );
+delete ( WasapiHandle* ) stream_.apiHandle;
+stream_.apiHandle = NULL;
+for ( int i = 0; i < 2; i++ ) {
+if ( stream_.userBuffer[i] ) {
+free( stream_.userBuffer[i] );
+stream_.userBuffer[i] = 0;
+}
+}
+if ( stream_.deviceBuffer ) {
+free( stream_.deviceBuffer );
+stream_.deviceBuffer = 0;
+}
+// update stream state
+stream_.state = STREAM_CLOSED;
+}
+//-----------------------------------------------------------------------------
+void RtApiWasapi::startStream( void )
+{
+verifyStream();
+if ( stream_.state == STREAM_RUNNING ) {
+errorText_ = "RtApiWasapi::startStream: The stream is already running.";
+error( RtAudioError::WARNING );
+return;
+}
+#if defined( HAVE_GETTIMEOFDAY )
+gettimeofday( &stream_.lastTickTimestamp, NULL );
+#endif
+// update stream state
+stream_.state = STREAM_RUNNING;
+// create WASAPI stream thread
+stream_.callbackInfo.thread = ( ThreadHandle ) CreateThread( NULL, 0, runWasapiThread, this, CREATE_SUSPENDED, NULL );
+if ( !stream_.callbackInfo.thread ) {
+errorText_ = "RtApiWasapi::startStream: Unable to instantiate callback thread.";
+error( RtAudioError::THREAD_ERROR );
+}
+else {
+SetThreadPriority( ( void* ) stream_.callbackInfo.thread, stream_.callbackInfo.priority );
+ResumeThread( ( void* ) stream_.callbackInfo.thread );
+}
+}
+//-----------------------------------------------------------------------------
+void RtApiWasapi::stopStream( void )
+{
+verifyStream();
+if ( stream_.state == STREAM_STOPPED ) {
+errorText_ = "RtApiWasapi::stopStream: The stream is already stopped.";
+error( RtAudioError::WARNING );
+return;
+}
+// inform stream thread by setting stream state to STREAM_STOPPING
+stream_.state = STREAM_STOPPING;
+// wait until stream thread is stopped
+for (int i=0; i < 2 && stream_.state != STREAM_STOPPED; ++i ) {
+Sleep( 1000 * stream_.bufferSize / stream_.sampleRate );
+}
+// close thread handle
+if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
+errorText_ = "RtApiWasapi::stopStream: Unable to close callback thread.";
+error( RtAudioError::THREAD_ERROR );
+return;
+}
+stream_.callbackInfo.thread = (ThreadHandle) NULL;
+}
+//-----------------------------------------------------------------------------
+void RtApiWasapi::abortStream( void )
+{
+verifyStream();
+if ( stream_.state == STREAM_STOPPED ) {
+errorText_ = "RtApiWasapi::abortStream: The stream is already stopped.";
+error( RtAudioError::WARNING );
+return;
+}
+// inform stream thread by setting stream state to STREAM_STOPPING
+stream_.state = STREAM_STOPPING;
+// wait until stream thread is stopped
+while ( stream_.state != STREAM_STOPPED ) {
+Sleep( 1 );
+}
+// close thread handle
+if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
+errorText_ = "RtApiWasapi::abortStream: Unable to close callback thread.";
+error( RtAudioError::THREAD_ERROR );
+return;
+}
+stream_.callbackInfo.thread = (ThreadHandle) NULL;
+}
+//-----------------------------------------------------------------------------
+bool RtApiWasapi::probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+unsigned int firstChannel, unsigned int sampleRate,
+RtAudioFormat format, unsigned int* bufferSize,
+RtAudio::StreamOptions* options )
+{
+bool methodResult = FAILURE;
+unsigned int captureDeviceCount = 0;
+unsigned int renderDeviceCount = 0;
+IMMDeviceCollection* captureDevices = NULL;
+IMMDeviceCollection* renderDevices = NULL;
+IMMDevice* devicePtr = NULL;
+WAVEFORMATEX* deviceFormat = NULL;
+unsigned int bufferBytes;
+stream_.state = STREAM_STOPPED;
+// create API Handle if not already created
+if ( !stream_.apiHandle )
+stream_.apiHandle = ( void* ) new WasapiHandle();
+// Count capture devices
+errorText_.clear();
+RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
+HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
+if ( FAILED( hr ) ) {
+errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device collection.";
+goto Exit;
+}
+hr = captureDevices->GetCount( &captureDeviceCount );
+if ( FAILED( hr ) ) {
+errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device count.";
+goto Exit;
+}
+// Count render devices
+hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
+if ( FAILED( hr ) ) {
+errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device collection.";
+goto Exit;
+}
+hr = renderDevices->GetCount( &renderDeviceCount );
+if ( FAILED( hr ) ) {
+errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device count.";
+goto Exit;
+}
+// validate device index
+if ( device >= captureDeviceCount + renderDeviceCount ) {
+errorType = RtAudioError::INVALID_USE;
+errorText_ = "RtApiWasapi::probeDeviceOpen: Invalid device index.";
+goto Exit;
+}
+// if device index falls within capture devices
+if ( device >= renderDeviceCount ) {
+if ( mode != INPUT ) {
+errorType = RtAudioError::INVALID_USE;
+errorText_ = "RtApiWasapi::probeDeviceOpen: Capture device selected as output device.";
+goto Exit;
+}
+// retrieve captureAudioClient from devicePtr
+IAudioClient*& captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
+hr = captureDevices->Item( device - renderDeviceCount, &devicePtr );
+if ( FAILED( hr ) ) {
+errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device handle.";
+goto Exit;
+}
+hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
+NULL, ( void** ) &captureAudioClient );
+if ( FAILED( hr ) ) {
+errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device audio client.";
+goto Exit;
+}
+hr = captureAudioClient->GetMixFormat( &deviceFormat );
+if ( FAILED( hr ) ) {
+errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device mix format.";
+goto Exit;
+}
+stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
+captureAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
+}
+// if device index falls within render devices and is configured for loopback
+if ( device < renderDeviceCount && mode == INPUT )
+{
+// if renderAudioClient is not initialised, initialise it now
+IAudioClient*& renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
+if ( !renderAudioClient )
+{
+probeDeviceOpen( device, OUTPUT, channels, firstChannel, sampleRate, format, bufferSize, options );
+}
+// retrieve captureAudioClient from devicePtr
+IAudioClient*& captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
+hr = renderDevices->Item( device, &devicePtr );
+if ( FAILED( hr ) ) {
+errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device handle.";
+goto Exit;
+}
+hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
+NULL, ( void** ) &captureAudioClient );
+if ( FAILED( hr ) ) {
+errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device audio client.";
+goto Exit;
+}
+hr = captureAudioClient->GetMixFormat( &deviceFormat );
+if ( FAILED( hr ) ) {
+errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device mix format.";
+goto Exit;
+}
+stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
+captureAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
+}
+// if device index falls within render devices and is configured for output
+if ( device < renderDeviceCount && mode == OUTPUT )
+{
+// if renderAudioClient is already initialised, don't initialise it again
+IAudioClient*& renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
+if ( renderAudioClient )
+{
+methodResult = SUCCESS;
+goto Exit;
+}
+hr = renderDevices->Item( device, &devicePtr );
+if ( FAILED( hr ) ) {
+errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device handle.";
+goto Exit;
+}
+hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
+NULL, ( void** ) &renderAudioClient );
+if ( FAILED( hr ) ) {
+errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device audio client.";
+goto Exit;
+}
+hr = renderAudioClient->GetMixFormat( &deviceFormat );
+if ( FAILED( hr ) ) {
+errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device mix format.";
+goto Exit;
+}
+stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
+renderAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
+}
+// fill stream data
+if ( ( stream_.mode == OUTPUT && mode == INPUT ) ||
+( stream_.mode == INPUT && mode == OUTPUT ) ) {
+stream_.mode = DUPLEX;
+}
+else {
+stream_.mode = mode;
+}
+stream_.device[mode] = device;
+stream_.doByteSwap[mode] = false;
+stream_.sampleRate = sampleRate;
+stream_.bufferSize = *bufferSize;
+stream_.nBuffers = 1;
+stream_.nUserChannels[mode] = channels;
+stream_.channelOffset[mode] = firstChannel;
+stream_.userFormat = format;
+stream_.deviceFormat[mode] = getDeviceInfo( device ).nativeFormats;
+if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
+stream_.userInterleaved = false;
+else
+stream_.userInterleaved = true;
+stream_.deviceInterleaved[mode] = true;
+// Set flags for buffer conversion.
+stream_.doConvertBuffer[mode] = false;
+if ( stream_.userFormat != stream_.deviceFormat[mode] ||
+stream_.nUserChannels[0] != stream_.nDeviceChannels[0] ||
+stream_.nUserChannels[1] != stream_.nDeviceChannels[1] )
+stream_.doConvertBuffer[mode] = true;
+else if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+stream_.nUserChannels[mode] > 1 )
+stream_.doConvertBuffer[mode] = true;
+if ( stream_.doConvertBuffer[mode] )
+setConvertInfo( mode, firstChannel );
+// Allocate necessary internal buffers
+bufferBytes = stream_.nUserChannels[mode] * stream_.bufferSize * formatBytes( stream_.userFormat );
+stream_.userBuffer[mode] = ( char* ) calloc( bufferBytes, 1 );
+if ( !stream_.userBuffer[mode] ) {
+errorType = RtAudioError::MEMORY_ERROR;
+errorText_ = "RtApiWasapi::probeDeviceOpen: Error allocating user buffer memory.";
+goto Exit;
+}
+if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME )
+stream_.callbackInfo.priority = 15;
+else
+stream_.callbackInfo.priority = 0;
+///! TODO: RTAUDIO_MINIMIZE_LATENCY // Provide stream buffers directly to callback
+///! TODO: RTAUDIO_HOG_DEVICE       // Exclusive mode
+methodResult = SUCCESS;
+Exit:
+//clean up
+SAFE_RELEASE( captureDevices );
+SAFE_RELEASE( renderDevices );
+SAFE_RELEASE( devicePtr );
+CoTaskMemFree( deviceFormat );
+// if method failed, close the stream
+if ( methodResult == FAILURE )
+closeStream();
+if ( !errorText_.empty() )
+error( errorType );
+return methodResult;
+}
+//=============================================================================
+DWORD WINAPI RtApiWasapi::runWasapiThread( void* wasapiPtr )
+{
+if ( wasapiPtr )
+( ( RtApiWasapi* ) wasapiPtr )->wasapiThread();
+return 0;
+}
+DWORD WINAPI RtApiWasapi::stopWasapiThread( void* wasapiPtr )
+{
+if ( wasapiPtr )
+( ( RtApiWasapi* ) wasapiPtr )->stopStream();
+return 0;
+}
+DWORD WINAPI RtApiWasapi::abortWasapiThread( void* wasapiPtr )
+{
+if ( wasapiPtr )
+( ( RtApiWasapi* ) wasapiPtr )->abortStream();
+return 0;
+}
+//-----------------------------------------------------------------------------
+void RtApiWasapi::wasapiThread()
+{
+// as this is a new thread, we must CoInitialize it
+CoInitialize( NULL );
+HRESULT hr;
+IAudioClient* captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
+IAudioClient* renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
+IAudioCaptureClient* captureClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureClient;
+IAudioRenderClient* renderClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderClient;
+HANDLE captureEvent = ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent;
+HANDLE renderEvent = ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent;
+WAVEFORMATEX* captureFormat = NULL;
+WAVEFORMATEX* renderFormat = NULL;
+float captureSrRatio = 0.0f;
+float renderSrRatio = 0.0f;
+WasapiBuffer captureBuffer;
+WasapiBuffer renderBuffer;
+WasapiResampler* captureResampler = NULL;
+WasapiResampler* renderResampler = NULL;
+// declare local stream variables
+RtAudioCallback callback = ( RtAudioCallback ) stream_.callbackInfo.callback;
+BYTE* streamBuffer = NULL;
+DWORD captureFlags = 0;
+unsigned int bufferFrameCount = 0;
+unsigned int numFramesPadding = 0;
+unsigned int convBufferSize = 0;
+bool loopbackEnabled = stream_.device[INPUT] == stream_.device[OUTPUT];
+bool callbackPushed = true;
+bool callbackPulled = false;
+bool callbackStopped = false;
+int callbackResult = 0;
+// convBuffer is used to store converted buffers between WASAPI and the user
+char* convBuffer = NULL;
+unsigned int convBuffSize = 0;
+unsigned int deviceBuffSize = 0;
+std::string errorText;
+RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
+// Attempt to assign "Pro Audio" characteristic to thread
+HMODULE AvrtDll = LoadLibraryW( L"AVRT.dll" );
+if ( AvrtDll ) {
+DWORD taskIndex = 0;
+TAvSetMmThreadCharacteristicsPtr AvSetMmThreadCharacteristicsPtr =
+( TAvSetMmThreadCharacteristicsPtr ) (void(*)()) GetProcAddress( AvrtDll, "AvSetMmThreadCharacteristicsW" );
+AvSetMmThreadCharacteristicsPtr( L"Pro Audio", &taskIndex );
+FreeLibrary( AvrtDll );
+}
+// start capture stream if applicable
+if ( captureAudioClient ) {
+hr = captureAudioClient->GetMixFormat( &captureFormat );
+if ( FAILED( hr ) ) {
+errorText = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
+goto Exit;
+}
+// init captureResampler
+captureResampler = new WasapiResampler( stream_.deviceFormat[INPUT] == RTAUDIO_FLOAT32 || stream_.deviceFormat[INPUT] == RTAUDIO_FLOAT64,
+formatBytes( stream_.deviceFormat[INPUT] ) * 8, stream_.nDeviceChannels[INPUT],
+captureFormat->nSamplesPerSec, stream_.sampleRate );
+captureSrRatio = ( ( float ) captureFormat->nSamplesPerSec / stream_.sampleRate );
+if ( !captureClient ) {
+hr = captureAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
+loopbackEnabled ? AUDCLNT_STREAMFLAGS_LOOPBACK : AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
+0,
+0,
+captureFormat,
+NULL );
+if ( FAILED( hr ) ) {
+errorText = "RtApiWasapi::wasapiThread: Unable to initialize capture audio client.";
+goto Exit;
+}
+hr = captureAudioClient->GetService( __uuidof( IAudioCaptureClient ),
+( void** ) &captureClient );
+if ( FAILED( hr ) ) {
+errorText = "RtApiWasapi::wasapiThread: Unable to retrieve capture client handle.";
+goto Exit;
+}
+// don't configure captureEvent if in loopback mode
+if ( !loopbackEnabled )
+{
+// configure captureEvent to trigger on every available capture buffer
+captureEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
+if ( !captureEvent ) {
+errorType = RtAudioError::SYSTEM_ERROR;
+errorText = "RtApiWasapi::wasapiThread: Unable to create capture event.";
+goto Exit;
+}
+hr = captureAudioClient->SetEventHandle( captureEvent );
+if ( FAILED( hr ) ) {
+errorText = "RtApiWasapi::wasapiThread: Unable to set capture event handle.";
+goto Exit;
+}
+( ( WasapiHandle* ) stream_.apiHandle )->captureEvent = captureEvent;
+}
+( ( WasapiHandle* ) stream_.apiHandle )->captureClient = captureClient;
+// reset the capture stream
+hr = captureAudioClient->Reset();
+if ( FAILED( hr ) ) {
+errorText = "RtApiWasapi::wasapiThread: Unable to reset capture stream.";
+goto Exit;
+}
+// start the capture stream
+hr = captureAudioClient->Start();
+if ( FAILED( hr ) ) {
+errorText = "RtApiWasapi::wasapiThread: Unable to start capture stream.";
+goto Exit;
+}
+}
+unsigned int inBufferSize = 0;
+hr = captureAudioClient->GetBufferSize( &inBufferSize );
+if ( FAILED( hr ) ) {
+errorText = "RtApiWasapi::wasapiThread: Unable to get capture buffer size.";
+goto Exit;
+}
+// scale outBufferSize according to stream->user sample rate ratio
+unsigned int outBufferSize = ( unsigned int ) ceilf( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT];
+inBufferSize *= stream_.nDeviceChannels[INPUT];
+// set captureBuffer size
+captureBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[INPUT] ) );
+}
+// start render stream if applicable
+if ( renderAudioClient ) {
+hr = renderAudioClient->GetMixFormat( &renderFormat );
+if ( FAILED( hr ) ) {
+errorText = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
+goto Exit;
+}
+// init renderResampler
+renderResampler = new WasapiResampler( stream_.deviceFormat[OUTPUT] == RTAUDIO_FLOAT32 || stream_.deviceFormat[OUTPUT] == RTAUDIO_FLOAT64,
+formatBytes( stream_.deviceFormat[OUTPUT] ) * 8, stream_.nDeviceChannels[OUTPUT],
+stream_.sampleRate, renderFormat->nSamplesPerSec );
+renderSrRatio = ( ( float ) renderFormat->nSamplesPerSec / stream_.sampleRate );
+if ( !renderClient ) {
+hr = renderAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
+AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
+0,
+0,
+renderFormat,
+NULL );
+if ( FAILED( hr ) ) {
+errorText = "RtApiWasapi::wasapiThread: Unable to initialize render audio client.";
+goto Exit;
+}
+hr = renderAudioClient->GetService( __uuidof( IAudioRenderClient ),
+( void** ) &renderClient );
+if ( FAILED( hr ) ) {
+errorText = "RtApiWasapi::wasapiThread: Unable to retrieve render client handle.";
+goto Exit;
+}
+// configure renderEvent to trigger on every available render buffer
+renderEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
+if ( !renderEvent ) {
+errorType = RtAudioError::SYSTEM_ERROR;
+errorText = "RtApiWasapi::wasapiThread: Unable to create render event.";
+goto Exit;
+}
+hr = renderAudioClient->SetEventHandle( renderEvent );
+if ( FAILED( hr ) ) {
+errorText = "RtApiWasapi::wasapiThread: Unable to set render event handle.";
+goto Exit;
+}
+( ( WasapiHandle* ) stream_.apiHandle )->renderClient = renderClient;
+( ( WasapiHandle* ) stream_.apiHandle )->renderEvent = renderEvent;
+// reset the render stream
+hr = renderAudioClient->Reset();
+if ( FAILED( hr ) ) {
+errorText = "RtApiWasapi::wasapiThread: Unable to reset render stream.";
+goto Exit;
+}
+// start the render stream
+hr = renderAudioClient->Start();
+if ( FAILED( hr ) ) {
+errorText = "RtApiWasapi::wasapiThread: Unable to start render stream.";
+goto Exit;
+}
+}
+unsigned int outBufferSize = 0;
+hr = renderAudioClient->GetBufferSize( &outBufferSize );
+if ( FAILED( hr ) ) {
+errorText = "RtApiWasapi::wasapiThread: Unable to get render buffer size.";
+goto Exit;
+}
+// scale inBufferSize according to user->stream sample rate ratio
+unsigned int inBufferSize = ( unsigned int ) ceilf( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT];
+outBufferSize *= stream_.nDeviceChannels[OUTPUT];
+// set renderBuffer size
+renderBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[OUTPUT] ) );
+}
+// malloc buffer memory
+if ( stream_.mode == INPUT )
+{
+using namespace std; // for ceilf
+convBuffSize = ( size_t ) ( ceilf( stream_.bufferSize * captureSrRatio ) ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
+deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
+}
+else if ( stream_.mode == OUTPUT )
+{
+convBuffSize = ( size_t ) ( ceilf( stream_.bufferSize * renderSrRatio ) ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
+deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
+}
+else if ( stream_.mode == DUPLEX )
+{
+convBuffSize = std::max( ( size_t ) ( ceilf( stream_.bufferSize * captureSrRatio ) ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
+( size_t ) ( ceilf( stream_.bufferSize * renderSrRatio ) ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
+deviceBuffSize = std::max( stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
+stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
+}
+convBuffSize *= 2; // allow overflow for *SrRatio remainders
+convBuffer = ( char* ) calloc( convBuffSize, 1 );
+stream_.deviceBuffer = ( char* ) calloc( deviceBuffSize, 1 );
+if ( !convBuffer || !stream_.deviceBuffer ) {
+errorType = RtAudioError::MEMORY_ERROR;
+errorText = "RtApiWasapi::wasapiThread: Error allocating device buffer memory.";
+goto Exit;
+}
+// stream process loop
+while ( stream_.state != STREAM_STOPPING ) {
+if ( !callbackPulled ) {
+// Callback Input
+// ==============
+// 1. Pull callback buffer from inputBuffer
+// 2. If 1. was successful: Convert callback buffer to user sample rate and channel count
+//                          Convert callback buffer to user format
+if ( captureAudioClient )
+{
+int samplesToPull = ( unsigned int ) floorf( stream_.bufferSize * captureSrRatio );
+convBufferSize = 0;
+while ( convBufferSize < stream_.bufferSize )
+{
+// Pull callback buffer from inputBuffer
+callbackPulled = captureBuffer.pullBuffer( convBuffer,
+samplesToPull * stream_.nDeviceChannels[INPUT],
+stream_.deviceFormat[INPUT] );
+if ( !callbackPulled )
+{
+break;
+}
+// Convert callback buffer to user sample rate
+unsigned int deviceBufferOffset = convBufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
+unsigned int convSamples = 0;
+captureResampler->Convert( stream_.deviceBuffer + deviceBufferOffset,
+convBuffer,
+samplesToPull,
+convSamples,
+convBufferSize == 0 ? -1 : stream_.bufferSize - convBufferSize );
+convBufferSize += convSamples;
+samplesToPull = 1; // now pull one sample at a time until we have stream_.bufferSize samples
+}
+if ( callbackPulled )
+{
+if ( stream_.doConvertBuffer[INPUT] ) {
+// Convert callback buffer to user format
+convertBuffer( stream_.userBuffer[INPUT],
+stream_.deviceBuffer,
+stream_.convertInfo[INPUT] );
+}
+else {
+// no further conversion, simple copy deviceBuffer to userBuffer
+memcpy( stream_.userBuffer[INPUT],
+stream_.deviceBuffer,
+stream_.bufferSize * stream_.nUserChannels[INPUT] * formatBytes( stream_.userFormat ) );
+}
+}
+}
+else {
+// if there is no capture stream, set callbackPulled flag
+callbackPulled = true;
+}
+// Execute Callback
+// ================
+// 1. Execute user callback method
+// 2. Handle return value from callback
+// if callback has not requested the stream to stop
+if ( callbackPulled && !callbackStopped ) {
+// Execute user callback method
+callbackResult = callback( stream_.userBuffer[OUTPUT],
+stream_.userBuffer[INPUT],
+stream_.bufferSize,
+getStreamTime(),
+captureFlags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY ? RTAUDIO_INPUT_OVERFLOW : 0,
+stream_.callbackInfo.userData );
+// tick stream time
+RtApi::tickStreamTime();
+// Handle return value from callback
+if ( callbackResult == 1 ) {
+// instantiate a thread to stop this thread
+HANDLE threadHandle = CreateThread( NULL, 0, stopWasapiThread, this, 0, NULL );
+if ( !threadHandle ) {
+errorType = RtAudioError::THREAD_ERROR;
+errorText = "RtApiWasapi::wasapiThread: Unable to instantiate stream stop thread.";
+goto Exit;
+}
+else if ( !CloseHandle( threadHandle ) ) {
+errorType = RtAudioError::THREAD_ERROR;
+errorText = "RtApiWasapi::wasapiThread: Unable to close stream stop thread handle.";
+goto Exit;
+}
+callbackStopped = true;
+}
+else if ( callbackResult == 2 ) {
+// instantiate a thread to stop this thread
+HANDLE threadHandle = CreateThread( NULL, 0, abortWasapiThread, this, 0, NULL );
+if ( !threadHandle ) {
+errorType = RtAudioError::THREAD_ERROR;
+errorText = "RtApiWasapi::wasapiThread: Unable to instantiate stream abort thread.";
+goto Exit;
+}
+else if ( !CloseHandle( threadHandle ) ) {
+errorType = RtAudioError::THREAD_ERROR;
+errorText = "RtApiWasapi::wasapiThread: Unable to close stream abort thread handle.";
+goto Exit;
+}
+callbackStopped = true;
+}
+}
+}
+// Callback Output
+// ===============
+// 1. Convert callback buffer to stream format
+// 2. Convert callback buffer to stream sample rate and channel count
+// 3. Push callback buffer into outputBuffer
+if ( renderAudioClient && callbackPulled )
+{
+// if the last call to renderBuffer.PushBuffer() was successful
+if ( callbackPushed || convBufferSize == 0 )
+{
+if ( stream_.doConvertBuffer[OUTPUT] )
+{
+// Convert callback buffer to stream format
+convertBuffer( stream_.deviceBuffer,
+stream_.userBuffer[OUTPUT],
+stream_.convertInfo[OUTPUT] );
+}
+else {
+// no further conversion, simple copy userBuffer to deviceBuffer
+memcpy( stream_.deviceBuffer,
+stream_.userBuffer[OUTPUT],
+stream_.bufferSize * stream_.nUserChannels[OUTPUT] * formatBytes( stream_.userFormat ) );
+}
+// Convert callback buffer to stream sample rate
+renderResampler->Convert( convBuffer,
+stream_.deviceBuffer,
+stream_.bufferSize,
+convBufferSize );
+}
+// Push callback buffer into outputBuffer
+callbackPushed = renderBuffer.pushBuffer( convBuffer,
+convBufferSize * stream_.nDeviceChannels[OUTPUT],
+stream_.deviceFormat[OUTPUT] );
+}
+else {
+// if there is no render stream, set callbackPushed flag
+callbackPushed = true;
+}
+// Stream Capture
+// ==============
+// 1. Get capture buffer from stream
+// 2. Push capture buffer into inputBuffer
+// 3. If 2. was successful: Release capture buffer
+if ( captureAudioClient ) {
+// if the callback input buffer was not pulled from captureBuffer, wait for next capture event
+if ( !callbackPulled ) {
+WaitForSingleObject( loopbackEnabled ? renderEvent : captureEvent, INFINITE );
+}
+// Get capture buffer from stream
+hr = captureClient->GetBuffer( &streamBuffer,
+&bufferFrameCount,
+&captureFlags, NULL, NULL );
+if ( FAILED( hr ) ) {
+errorText = "RtApiWasapi::wasapiThread: Unable to retrieve capture buffer.";
+goto Exit;
+}
+if ( bufferFrameCount != 0 ) {
+// Push capture buffer into inputBuffer
+if ( captureBuffer.pushBuffer( ( char* ) streamBuffer,
+bufferFrameCount * stream_.nDeviceChannels[INPUT],
+stream_.deviceFormat[INPUT] ) )
+{
+// Release capture buffer
+hr = captureClient->ReleaseBuffer( bufferFrameCount );
+if ( FAILED( hr ) ) {
+errorText = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
+goto Exit;
+}
+}
+else
+{
+// Inform WASAPI that capture was unsuccessful
+hr = captureClient->ReleaseBuffer( 0 );
+if ( FAILED( hr ) ) {
+errorText = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
+goto Exit;
+}
+}
+}
+else
+{
+// Inform WASAPI that capture was unsuccessful
+hr = captureClient->ReleaseBuffer( 0 );
+if ( FAILED( hr ) ) {
+errorText = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
+goto Exit;
+}
+}
+}
+// Stream Render
+// =============
+// 1. Get render buffer from stream
+// 2. Pull next buffer from outputBuffer
+// 3. If 2. was successful: Fill render buffer with next buffer
+//                          Release render buffer
+if ( renderAudioClient ) {
+// if the callback output buffer was not pushed to renderBuffer, wait for next render event
+if ( callbackPulled && !callbackPushed ) {
+WaitForSingleObject( renderEvent, INFINITE );
+}
+// Get render buffer from stream
+hr = renderAudioClient->GetBufferSize( &bufferFrameCount );
+if ( FAILED( hr ) ) {
+errorText = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer size.";
+goto Exit;
+}
+hr = renderAudioClient->GetCurrentPadding( &numFramesPadding );
+if ( FAILED( hr ) ) {
+errorText = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer padding.";
+goto Exit;
+}
+bufferFrameCount -= numFramesPadding;
+if ( bufferFrameCount != 0 ) {
+hr = renderClient->GetBuffer( bufferFrameCount, &streamBuffer );
+if ( FAILED( hr ) ) {
+errorText = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer.";
+goto Exit;
+}
+// Pull next buffer from outputBuffer
+// Fill render buffer with next buffer
+if ( renderBuffer.pullBuffer( ( char* ) streamBuffer,
+bufferFrameCount * stream_.nDeviceChannels[OUTPUT],
+stream_.deviceFormat[OUTPUT] ) )
+{
+// Release render buffer
+hr = renderClient->ReleaseBuffer( bufferFrameCount, 0 );
+if ( FAILED( hr ) ) {
+errorText = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
+goto Exit;
+}
+}
+else
+{
+// Inform WASAPI that render was unsuccessful
+hr = renderClient->ReleaseBuffer( 0, 0 );
+if ( FAILED( hr ) ) {
+errorText = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
+goto Exit;
+}
+}
+}
+else
+{
+// Inform WASAPI that render was unsuccessful
+hr = renderClient->ReleaseBuffer( 0, 0 );
+if ( FAILED( hr ) ) {
+errorText = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
+goto Exit;
+}
+}
+}
+// if the callback buffer was pushed renderBuffer reset callbackPulled flag
+if ( callbackPushed ) {
+// unsetting the callbackPulled flag lets the stream know that
+// the audio device is ready for another callback output buffer.
+callbackPulled = false;
+}
+}
+Exit:
+// clean up
+CoTaskMemFree( captureFormat );
+CoTaskMemFree( renderFormat );
+free ( convBuffer );
+delete renderResampler;
+delete captureResampler;
+CoUninitialize();
+// update stream state
+stream_.state = STREAM_STOPPED;
+if ( !errorText.empty() )
+{
+errorText_ = errorText;
+error( errorType );
+}
+}
+//******************** End of __WINDOWS_WASAPI__ *********************//
+#endif
+#if defined(__WINDOWS_DS__) // Windows DirectSound API
+// Modified by Robin Davies, October 2005
+// - Improvements to DirectX pointer chasing.
+// - Bug fix for non-power-of-two Asio granularity used by Edirol PCR-A30.
+// - Auto-call CoInitialize for DSOUND and ASIO platforms.
+// Various revisions for RtAudio 4.0 by Gary Scavone, April 2007
+// Changed device query structure for RtAudio 4.0.7, January 2010
+#include <windows.h>
+#include <process.h>
+#include <mmsystem.h>
+#include <mmreg.h>
+#include <dsound.h>
+#include <assert.h>
+#include <algorithm>
+#if defined(__MINGW32__)
+// missing from latest mingw winapi
+#define WAVE_FORMAT_96M08 0x00010000 /* 96 kHz, Mono, 8-bit */
+#define WAVE_FORMAT_96S08 0x00020000 /* 96 kHz, Stereo, 8-bit */
+#define WAVE_FORMAT_96M16 0x00040000 /* 96 kHz, Mono, 16-bit */
+#define WAVE_FORMAT_96S16 0x00080000 /* 96 kHz, Stereo, 16-bit */
+#endif
+#define MINIMUM_DEVICE_BUFFER_SIZE 32768
+#ifdef _MSC_VER // if Microsoft Visual C++
+#pragma comment( lib, "winmm.lib" ) // then, auto-link winmm.lib. Otherwise, it has to be added manually.
+#endif
+static inline DWORD dsPointerBetween( DWORD pointer, DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize )
+{
+if ( pointer > bufferSize ) pointer -= bufferSize;
+if ( laterPointer < earlierPointer ) laterPointer += bufferSize;
+if ( pointer < earlierPointer ) pointer += bufferSize;
+return pointer >= earlierPointer && pointer < laterPointer;
+}
+// A structure to hold various information related to the DirectSound
+// API implementation.
+struct DsHandle {
+unsigned int drainCounter; // Tracks callback counts when draining
+bool internalDrain;        // Indicates if stop is initiated from callback or not.
+void *id[2];
+void *buffer[2];
+bool xrun[2];
+UINT bufferPointer[2];
+DWORD dsBufferSize[2];
+DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by.
+HANDLE condition;
+DsHandle()
+:drainCounter(0), internalDrain(false) { id[0] = 0; id[1] = 0; buffer[0] = 0; buffer[1] = 0; xrun[0] = false; xrun[1] = false; bufferPointer[0] = 0; bufferPointer[1] = 0; }
+};
+// Declarations for utility functions, callbacks, and structures
+// specific to the DirectSound implementation.
+static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
+LPCTSTR description,
+LPCTSTR module,
+LPVOID lpContext );
+static const char* getErrorString( int code );
+static unsigned __stdcall callbackHandler( void *ptr );
+struct DsDevice {
+LPGUID id[2];
+bool validId[2];
+bool found;
+std::string name;
+DsDevice()
+: found(false) { validId[0] = false; validId[1] = false; }
+};
+struct DsProbeData {
+bool isInput;
+std::vector<struct DsDevice>* dsDevices;
+};
+RtApiDs :: RtApiDs()
+{
+// Dsound will run both-threaded. If CoInitialize fails, then just
+// accept whatever the mainline chose for a threading model.
+coInitialized_ = false;
+HRESULT hr = CoInitialize( NULL );
+if ( !FAILED( hr ) ) coInitialized_ = true;
+}
+RtApiDs :: ~RtApiDs()
+{
+if ( stream_.state != STREAM_CLOSED ) closeStream();
+if ( coInitialized_ ) CoUninitialize(); // balanced call.
+}
+// The DirectSound default output is always the first device.
+unsigned int RtApiDs :: getDefaultOutputDevice( void )
+{
+return 0;
+}
+// The DirectSound default input is always the first input device,
+// which is the first capture device enumerated.
+unsigned int RtApiDs :: getDefaultInputDevice( void )
+{
+return 0;
+}
+unsigned int RtApiDs :: getDeviceCount( void )
+{
+// Set query flag for previously found devices to false, so that we
+// can check for any devices that have disappeared.
+for ( unsigned int i=0; i<dsDevices.size(); i++ )
+dsDevices[i].found = false;
+// Query DirectSound devices.
+struct DsProbeData probeInfo;
+probeInfo.isInput = false;
+probeInfo.dsDevices = &dsDevices;
+HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );
+if ( FAILED( result ) ) {
+errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!";
+errorText_ = errorStream_.str();
+error( RtAudioError::WARNING );
+}
+// Query DirectSoundCapture devices.
+probeInfo.isInput = true;
+result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );
+if ( FAILED( result ) ) {
+errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!";
+errorText_ = errorStream_.str();
+error( RtAudioError::WARNING );
+}
+// Clean out any devices that may have disappeared (code update submitted by Eli Zehngut).
+for ( unsigned int i=0; i<dsDevices.size(); ) {
+if ( dsDevices[i].found == false ) dsDevices.erase( dsDevices.begin() + i );
+else i++;
+}
+return static_cast<unsigned int>(dsDevices.size());
+}
+RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device )
+{
+RtAudio::DeviceInfo info;
+info.probed = false;
+if ( dsDevices.size() == 0 ) {
+// Force a query of all devices
+getDeviceCount();
+if ( dsDevices.size() == 0 ) {
+errorText_ = "RtApiDs::getDeviceInfo: no devices found!";
+error( RtAudioError::INVALID_USE );
+return info;
+}
+}
+if ( device >= dsDevices.size() ) {
+errorText_ = "RtApiDs::getDeviceInfo: device ID is invalid!";
+error( RtAudioError::INVALID_USE );
+return info;
+}
+HRESULT result;
+if ( dsDevices[ device ].validId[0] == false ) goto probeInput;
+LPDIRECTSOUND output;
+DSCAPS outCaps;
+result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );
+if ( FAILED( result ) ) {
+errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!";
+errorText_ = errorStream_.str();
+error( RtAudioError::WARNING );
+goto probeInput;
+}
+outCaps.dwSize = sizeof( outCaps );
+result = output->GetCaps( &outCaps );
+if ( FAILED( result ) ) {
+output->Release();
+errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting capabilities!";
+errorText_ = errorStream_.str();
+error( RtAudioError::WARNING );
+goto probeInput;
+}
+// Get output channel information.
+info.outputChannels = ( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1;
+// Get sample rate information.
+info.sampleRates.clear();
+for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
+if ( SAMPLE_RATES[k] >= (unsigned int) outCaps.dwMinSecondarySampleRate &&
+SAMPLE_RATES[k] <= (unsigned int) outCaps.dwMaxSecondarySampleRate ) {
+info.sampleRates.push_back( SAMPLE_RATES[k] );
+if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
+info.preferredSampleRate = SAMPLE_RATES[k];
+}
+}
+// Get format information.
+if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT ) info.nativeFormats |= RTAUDIO_SINT16;
+if ( outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) info.nativeFormats |= RTAUDIO_SINT8;
+output->Release();
+if ( getDefaultOutputDevice() == device )
+info.isDefaultOutput = true;
+if ( dsDevices[ device ].validId[1] == false ) {
+info.name = dsDevices[ device ].name;
+info.probed = true;
+return info;
+}
+probeInput:
+LPDIRECTSOUNDCAPTURE input;
+result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );
+if ( FAILED( result ) ) {
+errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";
+errorText_ = errorStream_.str();
+error( RtAudioError::WARNING );
+return info;
+}
+DSCCAPS inCaps;
+inCaps.dwSize = sizeof( inCaps );
+result = input->GetCaps( &inCaps );
+if ( FAILED( result ) ) {
+input->Release();
+errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsDevices[ device ].name << ")!";
+errorText_ = errorStream_.str();
+error( RtAudioError::WARNING );
+return info;
+}
+// Get input channel information.
+info.inputChannels = inCaps.dwChannels;
+// Get sample rate and format information.
+std::vector<unsigned int> rates;
+if ( inCaps.dwChannels >= 2 ) {
+if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.nativeFormats |= RTAUDIO_SINT16;
+if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.nativeFormats |= RTAUDIO_SINT16;
+if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.nativeFormats |= RTAUDIO_SINT16;
+if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.nativeFormats |= RTAUDIO_SINT16;
+if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.nativeFormats |= RTAUDIO_SINT8;
+if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.nativeFormats |= RTAUDIO_SINT8;
+if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.nativeFormats |= RTAUDIO_SINT8;
+if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.nativeFormats |= RTAUDIO_SINT8;
+if ( info.nativeFormats & RTAUDIO_SINT16 ) {
+if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) rates.push_back( 11025 );
+if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) rates.push_back( 22050 );
+if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) rates.push_back( 44100 );
+if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) rates.push_back( 96000 );
+}
+else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
+if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) rates.push_back( 11025 );
+if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) rates.push_back( 22050 );
+if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) rates.push_back( 44100 );
+if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) rates.push_back( 96000 );
+}
+}
+else if ( inCaps.dwChannels == 1 ) {
+if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.nativeFormats |= RTAUDIO_SINT16;
+if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.nativeFormats |= RTAUDIO_SINT16;
+if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.nativeFormats |= RTAUDIO_SINT16;
+if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.nativeFormats |= RTAUDIO_SINT16;
+if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.nativeFormats |= RTAUDIO_SINT8;
+if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.nativeFormats |= RTAUDIO_SINT8;
+if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.nativeFormats |= RTAUDIO_SINT8;
+if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.nativeFormats |= RTAUDIO_SINT8;
+if ( info.nativeFormats & RTAUDIO_SINT16 ) {
+if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) rates.push_back( 11025 );
+if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) rates.push_back( 22050 );
+if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) rates.push_back( 44100 );
+if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) rates.push_back( 96000 );
+}
+else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
+if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) rates.push_back( 11025 );
+if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) rates.push_back( 22050 );
+if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) rates.push_back( 44100 );
+if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) rates.push_back( 96000 );
+}
+}
+else info.inputChannels = 0; // technically, this would be an error
+input->Release();
+if ( info.inputChannels == 0 ) return info;
+// Copy the supported rates to the info structure but avoid duplication.
+bool found;
+for ( unsigned int i=0; i<rates.size(); i++ ) {
+found = false;
+for ( unsigned int j=0; j<info.sampleRates.size(); j++ ) {
+if ( rates[i] == info.sampleRates[j] ) {
+found = true;
+break;
+}
+}
+if ( found == false ) info.sampleRates.push_back( rates[i] );
+}
+std::sort( info.sampleRates.begin(), info.sampleRates.end() );
+// If device opens for both playback and capture, we determine the channels.
+if ( info.outputChannels > 0 && info.inputChannels > 0 )
+info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
+if ( device == 0 ) info.isDefaultInput = true;
+// Copy name and return.
+info.name = dsDevices[ device ].name;
+info.probed = true;
+return info;
+}
+bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+unsigned int firstChannel, unsigned int sampleRate,
+RtAudioFormat format, unsigned int *bufferSize,
+RtAudio::StreamOptions *options )
+{
+if ( channels + firstChannel > 2 ) {
+errorText_ = "RtApiDs::probeDeviceOpen: DirectSound does not support more than 2 channels per device.";
+return FAILURE;
+}
+size_t nDevices = dsDevices.size();
+if ( nDevices == 0 ) {
+// This should not happen because a check is made before this function is called.
+errorText_ = "RtApiDs::probeDeviceOpen: no devices found!";
+return FAILURE;
+}
+if ( device >= nDevices ) {
+// This should not happen because a check is made before this function is called.
+errorText_ = "RtApiDs::probeDeviceOpen: device ID is invalid!";
+return FAILURE;
+}
+if ( mode == OUTPUT ) {
+if ( dsDevices[ device ].validId[0] == false ) {
+errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support output!";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+}
+else { // mode == INPUT
+if ( dsDevices[ device ].validId[1] == false ) {
+errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support input!";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+}
+// According to a note in PortAudio, using GetDesktopWindow()
+// instead of GetForegroundWindow() is supposed to avoid problems
+// that occur when the application's window is not the foreground
+// window.  Also, if the application window closes before the
+// DirectSound buffer, DirectSound can crash.  In the past, I had
+// problems when using GetDesktopWindow() but it seems fine now
+// (January 2010).  I'll leave it commented here.
+// HWND hWnd = GetForegroundWindow();
+HWND hWnd = GetDesktopWindow();
+// Check the numberOfBuffers parameter and limit the lowest value to
+// two.  This is a judgement call and a value of two is probably too
+// low for capture, but it should work for playback.
+int nBuffers = 0;
+if ( options ) nBuffers = options->numberOfBuffers;
+if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) nBuffers = 2;
+if ( nBuffers < 2 ) nBuffers = 3;
+// Check the lower range of the user-specified buffer size and set
+// (arbitrarily) to a lower bound of 32.
+if ( *bufferSize < 32 ) *bufferSize = 32;
+// Create the wave format structure.  The data format setting will
+// be determined later.
+WAVEFORMATEX waveFormat;
+ZeroMemory( &waveFormat, sizeof(WAVEFORMATEX) );
+waveFormat.wFormatTag = WAVE_FORMAT_PCM;
+waveFormat.nChannels = channels + firstChannel;
+waveFormat.nSamplesPerSec = (unsigned long) sampleRate;
+// Determine the device buffer size. By default, we'll use the value
+// defined above (32K), but we will grow it to make allowances for
+// very large software buffer sizes.
+DWORD dsBufferSize = MINIMUM_DEVICE_BUFFER_SIZE;
+DWORD dsPointerLeadTime = 0;
+void *ohandle = 0, *bhandle = 0;
+HRESULT result;
+if ( mode == OUTPUT ) {
+LPDIRECTSOUND output;
+result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );
+if ( FAILED( result ) ) {
+errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+DSCAPS outCaps;
+outCaps.dwSize = sizeof( outCaps );
+result = output->GetCaps( &outCaps );
+if ( FAILED( result ) ) {
+output->Release();
+errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting capabilities (" << dsDevices[ device ].name << ")!";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+// Check channel information.
+if ( channels + firstChannel == 2 && !( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ) {
+errorStream_ << "RtApiDs::getDeviceInfo: the output device (" << dsDevices[ device ].name << ") does not support stereo playback.";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+// Check format information.  Use 16-bit format unless not
+// supported or user requests 8-bit.
+if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT &&
+!( format == RTAUDIO_SINT8 && outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) ) {
+waveFormat.wBitsPerSample = 16;
+stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+}
+else {
+waveFormat.wBitsPerSample = 8;
+stream_.deviceFormat[mode] = RTAUDIO_SINT8;
+}
+stream_.userFormat = format;
+// Update wave format structure and buffer information.
+waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
+waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
+dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
+// If the user wants an even bigger buffer, increase the device buffer size accordingly.
+while ( dsPointerLeadTime * 2U > dsBufferSize )
+dsBufferSize *= 2;
+// Set cooperative level to DSSCL_EXCLUSIVE ... sound stops when window focus changes.
+// result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE );
+// Set cooperative level to DSSCL_PRIORITY ... sound remains when window focus changes.
+result = output->SetCooperativeLevel( hWnd, DSSCL_PRIORITY );
+if ( FAILED( result ) ) {
+output->Release();
+errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting cooperative level (" << dsDevices[ device ].name << ")!";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+// Even though we will write to the secondary buffer, we need to
+// access the primary buffer to set the correct output format
+// (since the default is 8-bit, 22 kHz!).  Setup the DS primary
+// buffer description.
+DSBUFFERDESC bufferDescription;
+ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
+bufferDescription.dwSize = sizeof( DSBUFFERDESC );
+bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER;
+// Obtain the primary buffer
+LPDIRECTSOUNDBUFFER buffer;
+result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
+if ( FAILED( result ) ) {
+output->Release();
+errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") accessing primary buffer (" << dsDevices[ device ].name << ")!";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+// Set the primary DS buffer sound format.
+result = buffer->SetFormat( &waveFormat );
+if ( FAILED( result ) ) {
+output->Release();
+errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting primary buffer format (" << dsDevices[ device ].name << ")!";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+// Setup the secondary DS buffer description.
+ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
+bufferDescription.dwSize = sizeof( DSBUFFERDESC );
+bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
+DSBCAPS_GLOBALFOCUS |
+DSBCAPS_GETCURRENTPOSITION2 |
+DSBCAPS_LOCHARDWARE );  // Force hardware mixing
+bufferDescription.dwBufferBytes = dsBufferSize;
+bufferDescription.lpwfxFormat = &waveFormat;
+// Try to create the secondary DS buffer.  If that doesn't work,
+// try to use software mixing.  Otherwise, there's a problem.
+result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
+if ( FAILED( result ) ) {
+bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
+DSBCAPS_GLOBALFOCUS |
+DSBCAPS_GETCURRENTPOSITION2 |
+DSBCAPS_LOCSOFTWARE );  // Force software mixing
+result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
+if ( FAILED( result ) ) {
+output->Release();
+errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating secondary buffer (" << dsDevices[ device ].name << ")!";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+}
+// Get the buffer size ... might be different from what we specified.
+DSBCAPS dsbcaps;
+dsbcaps.dwSize = sizeof( DSBCAPS );
+result = buffer->GetCaps( &dsbcaps );
+if ( FAILED( result ) ) {
+output->Release();
+buffer->Release();
+errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+dsBufferSize = dsbcaps.dwBufferBytes;
+// Lock the DS buffer
+LPVOID audioPtr;
+DWORD dataLen;
+result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );
+if ( FAILED( result ) ) {
+output->Release();
+buffer->Release();
+errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking buffer (" << dsDevices[ device ].name << ")!";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+// Zero the DS buffer
+ZeroMemory( audioPtr, dataLen );
+// Unlock the DS buffer
+result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
+if ( FAILED( result ) ) {
+output->Release();
+buffer->Release();
+errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking buffer (" << dsDevices[ device ].name << ")!";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+ohandle = (void *) output;
+bhandle = (void *) buffer;
+}
+if ( mode == INPUT ) {
+LPDIRECTSOUNDCAPTURE input;
+result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );
+if ( FAILED( result ) ) {
+errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+DSCCAPS inCaps;
+inCaps.dwSize = sizeof( inCaps );
+result = input->GetCaps( &inCaps );
+if ( FAILED( result ) ) {
+input->Release();
+errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsDevices[ device ].name << ")!";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+// Check channel information.
+if ( inCaps.dwChannels < channels + firstChannel ) {
+errorText_ = "RtApiDs::getDeviceInfo: the input device does not support requested input channels.";
+return FAILURE;
+}
+// Check format information.  Use 16-bit format unless user
+// requests 8-bit.
+DWORD deviceFormats;
+if ( channels + firstChannel == 2 ) {
+deviceFormats = WAVE_FORMAT_1S08 | WAVE_FORMAT_2S08 | WAVE_FORMAT_4S08 | WAVE_FORMAT_96S08;
+if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
+waveFormat.wBitsPerSample = 8;
+stream_.deviceFormat[mode] = RTAUDIO_SINT8;
+}
+else { // assume 16-bit is supported
+waveFormat.wBitsPerSample = 16;
+stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+}
+}
+else { // channel == 1
+deviceFormats = WAVE_FORMAT_1M08 | WAVE_FORMAT_2M08 | WAVE_FORMAT_4M08 | WAVE_FORMAT_96M08;
+if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
+waveFormat.wBitsPerSample = 8;
+stream_.deviceFormat[mode] = RTAUDIO_SINT8;
+}
+else { // assume 16-bit is supported
+waveFormat.wBitsPerSample = 16;
+stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+}
+}
+stream_.userFormat = format;
+// Update wave format structure and buffer information.
+waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
+waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
+dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
+// If the user wants an even bigger buffer, increase the device buffer size accordingly.
+while ( dsPointerLeadTime * 2U > dsBufferSize )
+dsBufferSize *= 2;
+// Setup the secondary DS buffer description.
+DSCBUFFERDESC bufferDescription;
+ZeroMemory( &bufferDescription, sizeof( DSCBUFFERDESC ) );
+bufferDescription.dwSize = sizeof( DSCBUFFERDESC );
+bufferDescription.dwFlags = 0;
+bufferDescription.dwReserved = 0;
+bufferDescription.dwBufferBytes = dsBufferSize;
+bufferDescription.lpwfxFormat = &waveFormat;
+// Create the capture buffer.
+LPDIRECTSOUNDCAPTUREBUFFER buffer;
+result = input->CreateCaptureBuffer( &bufferDescription, &buffer, NULL );
+if ( FAILED( result ) ) {
+input->Release();
+errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating input buffer (" << dsDevices[ device ].name << ")!";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+// Get the buffer size ... might be different from what we specified.
+DSCBCAPS dscbcaps;
+dscbcaps.dwSize = sizeof( DSCBCAPS );
+result = buffer->GetCaps( &dscbcaps );
+if ( FAILED( result ) ) {
+input->Release();
+buffer->Release();
+errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+dsBufferSize = dscbcaps.dwBufferBytes;
+// NOTE: We could have a problem here if this is a duplex stream
+// and the play and capture hardware buffer sizes are different
+// (I'm actually not sure if that is a problem or not).
+// Currently, we are not verifying that.
+// Lock the capture buffer
+LPVOID audioPtr;
+DWORD dataLen;
+result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );
+if ( FAILED( result ) ) {
+input->Release();
+buffer->Release();
+errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking input buffer (" << dsDevices[ device ].name << ")!";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+// Zero the buffer
+ZeroMemory( audioPtr, dataLen );
+// Unlock the buffer
+result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
+if ( FAILED( result ) ) {
+input->Release();
+buffer->Release();
+errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking input buffer (" << dsDevices[ device ].name << ")!";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+ohandle = (void *) input;
+bhandle = (void *) buffer;
+}
+// Set various stream parameters
+DsHandle *handle = 0;
+stream_.nDeviceChannels[mode] = channels + firstChannel;
+stream_.nUserChannels[mode] = channels;
+stream_.bufferSize = *bufferSize;
+stream_.channelOffset[mode] = firstChannel;
+stream_.deviceInterleaved[mode] = true;
+if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
+else stream_.userInterleaved = true;
+// Set flag for buffer conversion
+stream_.doConvertBuffer[mode] = false;
+if (stream_.nUserChannels[mode] != stream_.nDeviceChannels[mode])
+stream_.doConvertBuffer[mode] = true;
+if (stream_.userFormat != stream_.deviceFormat[mode])
+stream_.doConvertBuffer[mode] = true;
+if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+stream_.nUserChannels[mode] > 1 )
+stream_.doConvertBuffer[mode] = true;
+// Allocate necessary internal buffers
+long bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+if ( stream_.userBuffer[mode] == NULL ) {
+errorText_ = "RtApiDs::probeDeviceOpen: error allocating user buffer memory.";
+goto error;
+}
+if ( stream_.doConvertBuffer[mode] ) {
+bool makeBuffer = true;
+bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
+if ( mode == INPUT ) {
+if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
+unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
+if ( bufferBytes <= (long) bytesOut ) makeBuffer = false;
+}
+}
+if ( makeBuffer ) {
+bufferBytes *= *bufferSize;
+if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
+stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
+if ( stream_.deviceBuffer == NULL ) {
+errorText_ = "RtApiDs::probeDeviceOpen: error allocating device buffer memory.";
+goto error;
+}
+}
+}
+// Allocate our DsHandle structures for the stream.
+if ( stream_.apiHandle == 0 ) {
+try {
+handle = new DsHandle;
+}
+catch ( std::bad_alloc& ) {
+errorText_ = "RtApiDs::probeDeviceOpen: error allocating AsioHandle memory.";
+goto error;
+}
+// Create a manual-reset event.
+handle->condition = CreateEvent( NULL,   // no security
+TRUE,   // manual-reset
+FALSE,  // non-signaled initially
+NULL ); // unnamed
+stream_.apiHandle = (void *) handle;
+}
+else
+handle = (DsHandle *) stream_.apiHandle;
+handle->id[mode] = ohandle;
+handle->buffer[mode] = bhandle;
+handle->dsBufferSize[mode] = dsBufferSize;
+handle->dsPointerLeadTime[mode] = dsPointerLeadTime;
+stream_.device[mode] = device;
+stream_.state = STREAM_STOPPED;
+if ( stream_.mode == OUTPUT && mode == INPUT )
+// We had already set up an output stream.
+stream_.mode = DUPLEX;
+else
+stream_.mode = mode;
+stream_.nBuffers = nBuffers;
+stream_.sampleRate = sampleRate;
+// Setup the buffer conversion information structure.
+if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
+// Setup the callback thread.
+if ( stream_.callbackInfo.isRunning == false ) {
+unsigned threadId;
+stream_.callbackInfo.isRunning = true;
+stream_.callbackInfo.object = (void *) this;
+stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &callbackHandler,
+&stream_.callbackInfo, 0, &threadId );
+if ( stream_.callbackInfo.thread == 0 ) {
+errorText_ = "RtApiDs::probeDeviceOpen: error creating callback thread!";
+goto error;
+}
+// Boost DS thread priority
+SetThreadPriority( (HANDLE) stream_.callbackInfo.thread, THREAD_PRIORITY_HIGHEST );
+}
+return SUCCESS;
+error:
+if ( handle ) {
+if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
+LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
+LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+if ( buffer ) buffer->Release();
+object->Release();
+}
+if ( handle->buffer[1] ) {
+LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
+LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+if ( buffer ) buffer->Release();
+object->Release();
+}
+CloseHandle( handle->condition );
+delete handle;
+stream_.apiHandle = 0;
+}
+for ( int i=0; i<2; i++ ) {
+if ( stream_.userBuffer[i] ) {
+free( stream_.userBuffer[i] );
+stream_.userBuffer[i] = 0;
+}
+}
+if ( stream_.deviceBuffer ) {
+free( stream_.deviceBuffer );
+stream_.deviceBuffer = 0;
+}
+stream_.state = STREAM_CLOSED;
+return FAILURE;
+}
+void RtApiDs :: closeStream()
+{
+if ( stream_.state == STREAM_CLOSED ) {
+errorText_ = "RtApiDs::closeStream(): no open stream to close!";
+error( RtAudioError::WARNING );
+return;
+}
+// Stop the callback thread.
+stream_.callbackInfo.isRunning = false;
+WaitForSingleObject( (HANDLE) stream_.callbackInfo.thread, INFINITE );
+CloseHandle( (HANDLE) stream_.callbackInfo.thread );
+DsHandle *handle = (DsHandle *) stream_.apiHandle;
+if ( handle ) {
+if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
+LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
+LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+if ( buffer ) {
+buffer->Stop();
+buffer->Release();
+}
+object->Release();
+}
+if ( handle->buffer[1] ) {
+LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
+LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+if ( buffer ) {
+buffer->Stop();
+buffer->Release();
+}
+object->Release();
+}
+CloseHandle( handle->condition );
+delete handle;
+stream_.apiHandle = 0;
+}
+for ( int i=0; i<2; i++ ) {
+if ( stream_.userBuffer[i] ) {
+free( stream_.userBuffer[i] );
+stream_.userBuffer[i] = 0;
+}
+}
+if ( stream_.deviceBuffer ) {
+free( stream_.deviceBuffer );
+stream_.deviceBuffer = 0;
+}
+stream_.mode = UNINITIALIZED;
+stream_.state = STREAM_CLOSED;
+}
+void RtApiDs :: startStream()
+{
+verifyStream();
+if ( stream_.state == STREAM_RUNNING ) {
+errorText_ = "RtApiDs::startStream(): the stream is already running!";
+error( RtAudioError::WARNING );
+return;
+}
+#if defined( HAVE_GETTIMEOFDAY )
+gettimeofday( &stream_.lastTickTimestamp, NULL );
+#endif
+DsHandle *handle = (DsHandle *) stream_.apiHandle;
+// Increase scheduler frequency on lesser windows (a side-effect of
+// increasing timer accuracy).  On greater windows (Win2K or later),
+// this is already in effect.
+timeBeginPeriod( 1 );
+buffersRolling = false;
+duplexPrerollBytes = 0;
+if ( stream_.mode == DUPLEX ) {
+// 0.5 seconds of silence in DUPLEX mode while the devices spin up and synchronize.
+duplexPrerollBytes = (int) ( 0.5 * stream_.sampleRate * formatBytes( stream_.deviceFormat[1] ) * stream_.nDeviceChannels[1] );
+}
+HRESULT result = 0;
+if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+result = buffer->Play( 0, 0, DSBPLAY_LOOPING );
+if ( FAILED( result ) ) {
+errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting output buffer!";
+errorText_ = errorStream_.str();
+goto unlock;
+}
+}
+if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+result = buffer->Start( DSCBSTART_LOOPING );
+if ( FAILED( result ) ) {
+errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting input buffer!";
+errorText_ = errorStream_.str();
+goto unlock;
+}
+}
+handle->drainCounter = 0;
+handle->internalDrain = false;
+ResetEvent( handle->condition );
+stream_.state = STREAM_RUNNING;
+unlock:
+if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );
+}
+void RtApiDs :: stopStream()
+{
+verifyStream();
+if ( stream_.state == STREAM_STOPPED ) {
+errorText_ = "RtApiDs::stopStream(): the stream is already stopped!";
+error( RtAudioError::WARNING );
+return;
+}
+HRESULT result = 0;
+LPVOID audioPtr;
+DWORD dataLen;
+DsHandle *handle = (DsHandle *) stream_.apiHandle;
+if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+if ( handle->drainCounter == 0 ) {
+handle->drainCounter = 2;
+WaitForSingleObject( handle->condition, INFINITE );  // block until signaled
+}
+stream_.state = STREAM_STOPPED;
+MUTEX_LOCK( &stream_.mutex );
+// Stop the buffer and clear memory
+LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+result = buffer->Stop();
+if ( FAILED( result ) ) {
+errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping output buffer!";
+errorText_ = errorStream_.str();
+goto unlock;
+}
+// Lock the buffer and clear it so that if we start to play again,
+// we won't have old data playing.
+result = buffer->Lock( 0, handle->dsBufferSize[0], &audioPtr, &dataLen, NULL, NULL, 0 );
+if ( FAILED( result ) ) {
+errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking output buffer!";
+errorText_ = errorStream_.str();
+goto unlock;
+}
+// Zero the DS buffer
+ZeroMemory( audioPtr, dataLen );
+// Unlock the DS buffer
+result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
+if ( FAILED( result ) ) {
+errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking output buffer!";
+errorText_ = errorStream_.str();
+goto unlock;
+}
+// If we start playing again, we must begin at beginning of buffer.
+handle->bufferPointer[0] = 0;
+}
+if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+audioPtr = NULL;
+dataLen = 0;
+stream_.state = STREAM_STOPPED;
+if ( stream_.mode != DUPLEX )
+MUTEX_LOCK( &stream_.mutex );
+result = buffer->Stop();
+if ( FAILED( result ) ) {
+errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping input buffer!";
+errorText_ = errorStream_.str();
+goto unlock;
+}
+// Lock the buffer and clear it so that if we start to play again,
+// we won't have old data playing.
+result = buffer->Lock( 0, handle->dsBufferSize[1], &audioPtr, &dataLen, NULL, NULL, 0 );
+if ( FAILED( result ) ) {
+errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking input buffer!";
+errorText_ = errorStream_.str();
+goto unlock;
+}
+// Zero the DS buffer
+ZeroMemory( audioPtr, dataLen );
+// Unlock the DS buffer
+result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
+if ( FAILED( result ) ) {
+errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking input buffer!";
+errorText_ = errorStream_.str();
+goto unlock;
+}
+// If we start recording again, we must begin at beginning of buffer.
+handle->bufferPointer[1] = 0;
+}
+unlock:
+timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows.
+MUTEX_UNLOCK( &stream_.mutex );
+if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );
+}
+void RtApiDs :: abortStream()
+{
+verifyStream();
+if ( stream_.state == STREAM_STOPPED ) {
+errorText_ = "RtApiDs::abortStream(): the stream is already stopped!";
+error( RtAudioError::WARNING );
+return;
+}
+DsHandle *handle = (DsHandle *) stream_.apiHandle;
+handle->drainCounter = 2;
+stopStream();
+}
+void RtApiDs :: callbackEvent()
+{
+if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) {
+Sleep( 50 ); // sleep 50 milliseconds
+return;
+}
+if ( stream_.state == STREAM_CLOSED ) {
+errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!";
+error( RtAudioError::WARNING );
+return;
+}
+CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
+DsHandle *handle = (DsHandle *) stream_.apiHandle;
+// Check if we were draining the stream and signal is finished.
+if ( handle->drainCounter > stream_.nBuffers + 2 ) {
+stream_.state = STREAM_STOPPING;
+if ( handle->internalDrain == false )
+SetEvent( handle->condition );
+else
+stopStream();
+return;
+}
+// Invoke user callback to get fresh output data UNLESS we are
+// draining stream.
+if ( handle->drainCounter == 0 ) {
+RtAudioCallback callback = (RtAudioCallback) info->callback;
+double streamTime = getStreamTime();
+RtAudioStreamStatus status = 0;
+if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
+status |= RTAUDIO_OUTPUT_UNDERFLOW;
+handle->xrun[0] = false;
+}
+if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
+status |= RTAUDIO_INPUT_OVERFLOW;
+handle->xrun[1] = false;
+}
+int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+stream_.bufferSize, streamTime, status, info->userData );
+if ( cbReturnValue == 2 ) {
+stream_.state = STREAM_STOPPING;
+handle->drainCounter = 2;
+abortStream();
+return;
+}
+else if ( cbReturnValue == 1 ) {
+handle->drainCounter = 1;
+handle->internalDrain = true;
+}
+}
+HRESULT result;
+DWORD currentWritePointer, safeWritePointer;
+DWORD currentReadPointer, safeReadPointer;
+UINT nextWritePointer;
+LPVOID buffer1 = NULL;
+LPVOID buffer2 = NULL;
+DWORD bufferSize1 = 0;
+DWORD bufferSize2 = 0;
+char *buffer;
+long bufferBytes;
+MUTEX_LOCK( &stream_.mutex );
+if ( stream_.state == STREAM_STOPPED ) {
+MUTEX_UNLOCK( &stream_.mutex );
+return;
+}
+if ( buffersRolling == false ) {
+if ( stream_.mode == DUPLEX ) {
+//assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
+// It takes a while for the devices to get rolling. As a result,
+// there's no guarantee that the capture and write device pointers
+// will move in lockstep.  Wait here for both devices to start
+// rolling, and then set our buffer pointers accordingly.
+// e.g. Crystal Drivers: the capture buffer starts up 5700 to 9600
+// bytes later than the write buffer.
+// Stub: a serious risk of having a pre-emptive scheduling round
+// take place between the two GetCurrentPosition calls... but I'm
+// really not sure how to solve the problem.  Temporarily boost to
+// Realtime priority, maybe; but I'm not sure what priority the
+// DirectSound service threads run at. We *should* be roughly
+// within a ms or so of correct.
+LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+DWORD startSafeWritePointer, startSafeReadPointer;
+result = dsWriteBuffer->GetCurrentPosition( NULL, &startSafeWritePointer );
+if ( FAILED( result ) ) {
+errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
+errorText_ = errorStream_.str();
+MUTEX_UNLOCK( &stream_.mutex );
+error( RtAudioError::SYSTEM_ERROR );
+return;
+}
+result = dsCaptureBuffer->GetCurrentPosition( NULL, &startSafeReadPointer );
+if ( FAILED( result ) ) {
+errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
+errorText_ = errorStream_.str();
+MUTEX_UNLOCK( &stream_.mutex );
+error( RtAudioError::SYSTEM_ERROR );
+return;
+}
+while ( true ) {
+result = dsWriteBuffer->GetCurrentPosition( NULL, &safeWritePointer );
+if ( FAILED( result ) ) {
+errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
+errorText_ = errorStream_.str();
+MUTEX_UNLOCK( &stream_.mutex );
+error( RtAudioError::SYSTEM_ERROR );
+return;
+}
+result = dsCaptureBuffer->GetCurrentPosition( NULL, &safeReadPointer );
+if ( FAILED( result ) ) {
+errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
+errorText_ = errorStream_.str();
+MUTEX_UNLOCK( &stream_.mutex );
+error( RtAudioError::SYSTEM_ERROR );
+return;
+}
+if ( safeWritePointer != startSafeWritePointer && safeReadPointer != startSafeReadPointer ) break;
+Sleep( 1 );
+}
+//assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
+handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];
+if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];
+handle->bufferPointer[1] = safeReadPointer;
+}
+else if ( stream_.mode == OUTPUT ) {
+// Set the proper nextWritePosition after initial startup.
+LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+result = dsWriteBuffer->GetCurrentPosition( &currentWritePointer, &safeWritePointer );
+if ( FAILED( result ) ) {
+errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
+errorText_ = errorStream_.str();
+MUTEX_UNLOCK( &stream_.mutex );
+error( RtAudioError::SYSTEM_ERROR );
+return;
+}
+handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];
+if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];
+}
+buffersRolling = true;
+}
+if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+if ( handle->drainCounter > 1 ) { // write zeros to the output stream
+bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
+bufferBytes *= formatBytes( stream_.userFormat );
+memset( stream_.userBuffer[0], 0, bufferBytes );
+}
+// Setup parameters and do buffer conversion if necessary.
+if ( stream_.doConvertBuffer[0] ) {
+buffer = stream_.deviceBuffer;
+convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
+bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[0];
+bufferBytes *= formatBytes( stream_.deviceFormat[0] );
+}
+else {
+buffer = stream_.userBuffer[0];
+bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
+bufferBytes *= formatBytes( stream_.userFormat );
+}
+// No byte swapping necessary in DirectSound implementation.
+// Ahhh ... windoze.  16-bit data is signed but 8-bit data is
+// unsigned.  So, we need to convert our signed 8-bit data here to
+// unsigned.
+if ( stream_.deviceFormat[0] == RTAUDIO_SINT8 )
+for ( int i=0; i<bufferBytes; i++ ) buffer[i] = (unsigned char) ( buffer[i] + 128 );
+DWORD dsBufferSize = handle->dsBufferSize[0];
+nextWritePointer = handle->bufferPointer[0];
+DWORD endWrite, leadPointer;
+while ( true ) {
+// Find out where the read and "safe write" pointers are.
+result = dsBuffer->GetCurrentPosition( &currentWritePointer, &safeWritePointer );
+if ( FAILED( result ) ) {
+errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
+errorText_ = errorStream_.str();
+MUTEX_UNLOCK( &stream_.mutex );
+error( RtAudioError::SYSTEM_ERROR );
+return;
+}
+// We will copy our output buffer into the region between
+// safeWritePointer and leadPointer.  If leadPointer is not
+// beyond the next endWrite position, wait until it is.
+leadPointer = safeWritePointer + handle->dsPointerLeadTime[0];
+//std::cout << "safeWritePointer = " << safeWritePointer << ", leadPointer = " << leadPointer << ", nextWritePointer = " << nextWritePointer << std::endl;
+if ( leadPointer > dsBufferSize ) leadPointer -= dsBufferSize;
+if ( leadPointer < nextWritePointer ) leadPointer += dsBufferSize; // unwrap offset
+endWrite = nextWritePointer + bufferBytes;
+// Check whether the entire write region is behind the play pointer.
+if ( leadPointer >= endWrite ) break;
+// If we are here, then we must wait until the leadPointer advances
+// beyond the end of our next write region. We use the
+// Sleep() function to suspend operation until that happens.
+double millis = ( endWrite - leadPointer ) * 1000.0;
+millis /= ( formatBytes( stream_.deviceFormat[0]) * stream_.nDeviceChannels[0] * stream_.sampleRate);
+if ( millis < 1.0 ) millis = 1.0;
+Sleep( (DWORD) millis );
+}
+if ( dsPointerBetween( nextWritePointer, safeWritePointer, currentWritePointer, dsBufferSize )
+|| dsPointerBetween( endWrite, safeWritePointer, currentWritePointer, dsBufferSize ) ) {
+// We've strayed into the forbidden zone ... resync the read pointer.
+handle->xrun[0] = true;
+nextWritePointer = safeWritePointer + handle->dsPointerLeadTime[0] - bufferBytes;
+if ( nextWritePointer >= dsBufferSize ) nextWritePointer -= dsBufferSize;
+handle->bufferPointer[0] = nextWritePointer;
+endWrite = nextWritePointer + bufferBytes;
+}
+// Lock free space in the buffer
+result = dsBuffer->Lock( nextWritePointer, bufferBytes, &buffer1,
+&bufferSize1, &buffer2, &bufferSize2, 0 );
+if ( FAILED( result ) ) {
+errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking buffer during playback!";
+errorText_ = errorStream_.str();
+MUTEX_UNLOCK( &stream_.mutex );
+error( RtAudioError::SYSTEM_ERROR );
+return;
+}
+// Copy our buffer into the DS buffer
+CopyMemory( buffer1, buffer, bufferSize1 );
+if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 );
+// Update our buffer offset and unlock sound buffer
+dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
+if ( FAILED( result ) ) {
+errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!";
+errorText_ = errorStream_.str();
+MUTEX_UNLOCK( &stream_.mutex );
+error( RtAudioError::SYSTEM_ERROR );
+return;
+}
+nextWritePointer = ( nextWritePointer + bufferSize1 + bufferSize2 ) % dsBufferSize;
+handle->bufferPointer[0] = nextWritePointer;
+}
+// Don't bother draining input
+if ( handle->drainCounter ) {
+handle->drainCounter++;
+goto unlock;
+}
+if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+// Setup parameters.
+if ( stream_.doConvertBuffer[1] ) {
+buffer = stream_.deviceBuffer;
+bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[1];
+bufferBytes *= formatBytes( stream_.deviceFormat[1] );
+}
+else {
+buffer = stream_.userBuffer[1];
+bufferBytes = stream_.bufferSize * stream_.nUserChannels[1];
+bufferBytes *= formatBytes( stream_.userFormat );
+}
+LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+long nextReadPointer = handle->bufferPointer[1];
+DWORD dsBufferSize = handle->dsBufferSize[1];
+// Find out where the write and "safe read" pointers are.
+result = dsBuffer->GetCurrentPosition( &currentReadPointer, &safeReadPointer );
+if ( FAILED( result ) ) {
+errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
+errorText_ = errorStream_.str();
+MUTEX_UNLOCK( &stream_.mutex );
+error( RtAudioError::SYSTEM_ERROR );
+return;
+}
+if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset
+DWORD endRead = nextReadPointer + bufferBytes;
+// Handling depends on whether we are INPUT or DUPLEX.
+// If we're in INPUT mode then waiting is a good thing. If we're in DUPLEX mode,
+// then a wait here will drag the write pointers into the forbidden zone.
+//
+// In DUPLEX mode, rather than wait, we will back off the read pointer until
+// it's in a safe position. This causes dropouts, but it seems to be the only
+// practical way to sync up the read and write pointers reliably, given the
+// the very complex relationship between phase and increment of the read and write
+// pointers.
+//
+// In order to minimize audible dropouts in DUPLEX mode, we will
+// provide a pre-roll period of 0.5 seconds in which we return
+// zeros from the read buffer while the pointers sync up.
+if ( stream_.mode == DUPLEX ) {
+if ( safeReadPointer < endRead ) {
+if ( duplexPrerollBytes <= 0 ) {
+// Pre-roll time over. Be more agressive.
+int adjustment = endRead-safeReadPointer;
+handle->xrun[1] = true;
+// Two cases:
+//   - large adjustments: we've probably run out of CPU cycles, so just resync exactly,
+//     and perform fine adjustments later.
+//   - small adjustments: back off by twice as much.
+if ( adjustment >= 2*bufferBytes )
+nextReadPointer = safeReadPointer-2*bufferBytes;
+else
+nextReadPointer = safeReadPointer-bufferBytes-adjustment;
+if ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;
+}
+else {
+// In pre=roll time. Just do it.
+nextReadPointer = safeReadPointer - bufferBytes;
+while ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;
+}
+endRead = nextReadPointer + bufferBytes;
+}
+}
+else { // mode == INPUT
+while ( safeReadPointer < endRead && stream_.callbackInfo.isRunning ) {
+// See comments for playback.
+double millis = (endRead - safeReadPointer) * 1000.0;
+millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate);
+if ( millis < 1.0 ) millis = 1.0;
+Sleep( (DWORD) millis );
+// Wake up and find out where we are now.
+result = dsBuffer->GetCurrentPosition( &currentReadPointer, &safeReadPointer );
+if ( FAILED( result ) ) {
+errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
+errorText_ = errorStream_.str();
+MUTEX_UNLOCK( &stream_.mutex );
+error( RtAudioError::SYSTEM_ERROR );
+return;
+}
+if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset
+}
+}
+// Lock free space in the buffer
+result = dsBuffer->Lock( nextReadPointer, bufferBytes, &buffer1,
+&bufferSize1, &buffer2, &bufferSize2, 0 );
+if ( FAILED( result ) ) {
+errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking capture buffer!";
+errorText_ = errorStream_.str();
+MUTEX_UNLOCK( &stream_.mutex );
+error( RtAudioError::SYSTEM_ERROR );
+return;
+}
+if ( duplexPrerollBytes <= 0 ) {
+// Copy our buffer into the DS buffer
+CopyMemory( buffer, buffer1, bufferSize1 );
+if ( buffer2 != NULL ) CopyMemory( buffer+bufferSize1, buffer2, bufferSize2 );
+}
+else {
+memset( buffer, 0, bufferSize1 );
+if ( buffer2 != NULL ) memset( buffer + bufferSize1, 0, bufferSize2 );
+duplexPrerollBytes -= bufferSize1 + bufferSize2;
+}
+// Update our buffer offset and unlock sound buffer
+nextReadPointer = ( nextReadPointer + bufferSize1 + bufferSize2 ) % dsBufferSize;
+dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
+if ( FAILED( result ) ) {
+errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking capture buffer!";
+errorText_ = errorStream_.str();
+MUTEX_UNLOCK( &stream_.mutex );
+error( RtAudioError::SYSTEM_ERROR );
+return;
+}
+handle->bufferPointer[1] = nextReadPointer;
+// No byte swapping necessary in DirectSound implementation.
+// If necessary, convert 8-bit data from unsigned to signed.
+if ( stream_.deviceFormat[1] == RTAUDIO_SINT8 )
+for ( int j=0; j<bufferBytes; j++ ) buffer[j] = (signed char) ( buffer[j] - 128 );
+// Do buffer conversion if necessary.
+if ( stream_.doConvertBuffer[1] )
+convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
+}
+unlock:
+MUTEX_UNLOCK( &stream_.mutex );
+RtApi::tickStreamTime();
+}
+// Definitions for utility functions and callbacks
+// specific to the DirectSound implementation.
+static unsigned __stdcall callbackHandler( void *ptr )
+{
+CallbackInfo *info = (CallbackInfo *) ptr;
+RtApiDs *object = (RtApiDs *) info->object;
+bool* isRunning = &info->isRunning;
+while ( *isRunning == true ) {
+object->callbackEvent();
+}
+_endthreadex( 0 );
+return 0;
+}
+static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
+LPCTSTR description,
+LPCTSTR /*module*/,
+LPVOID lpContext )
+{
+struct DsProbeData& probeInfo = *(struct DsProbeData*) lpContext;
+std::vector<struct DsDevice>& dsDevices = *probeInfo.dsDevices;
+HRESULT hr;
+bool validDevice = false;
+if ( probeInfo.isInput == true ) {
+DSCCAPS caps;
+LPDIRECTSOUNDCAPTURE object;
+hr = DirectSoundCaptureCreate(  lpguid, &object,   NULL );
+if ( hr != DS_OK ) return TRUE;
+caps.dwSize = sizeof(caps);
+hr = object->GetCaps( &caps );
+if ( hr == DS_OK ) {
+if ( caps.dwChannels > 0 && caps.dwFormats > 0 )
+validDevice = true;
+}
+object->Release();
+}
+else {
+DSCAPS caps;
+LPDIRECTSOUND object;
+hr = DirectSoundCreate(  lpguid, &object,   NULL );
+if ( hr != DS_OK ) return TRUE;
+caps.dwSize = sizeof(caps);
+hr = object->GetCaps( &caps );
+if ( hr == DS_OK ) {
+if ( caps.dwFlags & DSCAPS_PRIMARYMONO || caps.dwFlags & DSCAPS_PRIMARYSTEREO )
+validDevice = true;
+}
+object->Release();
+}
+// If good device, then save its name and guid.
+std::string name = convertCharPointerToStdString( description );
+//if ( name == "Primary Sound Driver" || name == "Primary Sound Capture Driver" )
+if ( lpguid == NULL )
+name = "Default Device";
+if ( validDevice ) {
+for ( unsigned int i=0; i<dsDevices.size(); i++ ) {
+if ( dsDevices[i].name == name ) {
+dsDevices[i].found = true;
+if ( probeInfo.isInput ) {
+dsDevices[i].id[1] = lpguid;
+dsDevices[i].validId[1] = true;
+}
+else {
+dsDevices[i].id[0] = lpguid;
+dsDevices[i].validId[0] = true;
+}
+return TRUE;
+}
+}
+DsDevice device;
+device.name = name;
+device.found = true;
+if ( probeInfo.isInput ) {
+device.id[1] = lpguid;
+device.validId[1] = true;
+}
+else {
+device.id[0] = lpguid;
+device.validId[0] = true;
+}
+dsDevices.push_back( device );
+}
+return TRUE;
+}
+static const char* getErrorString( int code )
+{
+switch ( code ) {
+case DSERR_ALLOCATED:
+return "Already allocated";
+case DSERR_CONTROLUNAVAIL:
+return "Control unavailable";
+case DSERR_INVALIDPARAM:
+return "Invalid parameter";
+case DSERR_INVALIDCALL:
+return "Invalid call";
+case DSERR_GENERIC:
+return "Generic error";
+case DSERR_PRIOLEVELNEEDED:
+return "Priority level needed";
+case DSERR_OUTOFMEMORY:
+return "Out of memory";
+case DSERR_BADFORMAT:
+return "The sample rate or the channel format is not supported";
+case DSERR_UNSUPPORTED:
+return "Not supported";
+case DSERR_NODRIVER:
+return "No driver";
+case DSERR_ALREADYINITIALIZED:
+return "Already initialized";
+case DSERR_NOAGGREGATION:
+return "No aggregation";
+case DSERR_BUFFERLOST:
+return "Buffer lost";
+case DSERR_OTHERAPPHASPRIO:
+return "Another application already has priority";
+case DSERR_UNINITIALIZED:
+return "Uninitialized";
+default:
+return "DirectSound unknown error";
+}
+}
+//******************** End of __WINDOWS_DS__ *********************//
+#endif
+#if defined(__LINUX_ALSA__)
+#include <alsa/asoundlib.h>
+#include <unistd.h>
+// A structure to hold various information related to the ALSA API
+// implementation.
+struct AlsaHandle {
+snd_pcm_t *handles[2];
+bool synchronized;
+bool xrun[2];
+pthread_cond_t runnable_cv;
+bool runnable;
+AlsaHandle()
+#if _cplusplus >= 201103L
+:handles{nullptr, nullptr}, synchronized(false), runnable(false) { xrun[0] = false; xrun[1] = false; }
+#else
+: synchronized(false), runnable(false) { handles[0] = NULL; handles[1] = NULL; xrun[0] = false; xrun[1] = false; }
+#endif
+};
+static void *alsaCallbackHandler( void * ptr );
+RtApiAlsa :: RtApiAlsa()
+{
+// Nothing to do here.
+}
+RtApiAlsa :: ~RtApiAlsa()
+{
+if ( stream_.state != STREAM_CLOSED ) closeStream();
+}
+unsigned int RtApiAlsa :: getDeviceCount( void )
+{
+unsigned nDevices = 0;
+int result, subdevice, card;
+char name[64];
+snd_ctl_t *handle = 0;
+strcpy(name, "default");
+result = snd_ctl_open( &handle, "default", 0 );
+if (result == 0) {
+nDevices++;
+snd_ctl_close( handle );
+}
+// Count cards and devices
+card = -1;
+snd_card_next( &card );
+while ( card >= 0 ) {
+sprintf( name, "hw:%d", card );
+result = snd_ctl_open( &handle, name, 0 );
+if ( result < 0 ) {
+handle = 0;
+errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << ".";
+errorText_ = errorStream_.str();
+error( RtAudioError::WARNING );
+goto nextcard;
+}
+subdevice = -1;
+while( 1 ) {
+result = snd_ctl_pcm_next_device( handle, &subdevice );
+if ( result < 0 ) {
+errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
+errorText_ = errorStream_.str();
+error( RtAudioError::WARNING );
+break;
+}
+if ( subdevice < 0 )
+break;
+nDevices++;
+}
+nextcard:
+if ( handle )
+snd_ctl_close( handle );
+snd_card_next( &card );
+}
+return nDevices;
+}
+RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device )
+{
+RtAudio::DeviceInfo info;
+info.probed = false;
+unsigned nDevices = 0;
+int result=-1, subdevice=-1, card=-1;
+char name[64];
+snd_ctl_t *chandle = 0;
+result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
+if ( result == 0 ) {
+if ( nDevices++ == device ) {
+strcpy( name, "default" );
+goto foundDevice;
+}
+}
+if ( chandle )
+snd_ctl_close( chandle );
+// Count cards and devices
+snd_card_next( &card );
+while ( card >= 0 ) {
+sprintf( name, "hw:%d", card );
+result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
+if ( result < 0 ) {
+chandle = 0;
+errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << ".";
+errorText_ = errorStream_.str();
+error( RtAudioError::WARNING );
+goto nextcard;
+}
+subdevice = -1;
+while( 1 ) {
+result = snd_ctl_pcm_next_device( chandle, &subdevice );
+if ( result < 0 ) {
+errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
+errorText_ = errorStream_.str();
+error( RtAudioError::WARNING );
+break;
+}
+if ( subdevice < 0 ) break;
+if ( nDevices == device ) {
+sprintf( name, "hw:%d,%d", card, subdevice );
+goto foundDevice;
+}
+nDevices++;
+}
+nextcard:
+if ( chandle )
+snd_ctl_close( chandle );
+snd_card_next( &card );
+}
+if ( nDevices == 0 ) {
+errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!";
+error( RtAudioError::INVALID_USE );
+return info;
+}
+if ( device >= nDevices ) {
+errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!";
+error( RtAudioError::INVALID_USE );
+return info;
+}
+foundDevice:
+// If a stream is already open, we cannot probe the stream devices.
+// Thus, use the saved results.
+if ( stream_.state != STREAM_CLOSED &&
+( stream_.device[0] == device || stream_.device[1] == device ) ) {
+snd_ctl_close( chandle );
+if ( device >= devices_.size() ) {
+errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened.";
+error( RtAudioError::WARNING );
+return info;
+}
+return devices_[ device ];
+}
+int openMode = SND_PCM_ASYNC;
+snd_pcm_stream_t stream;
+snd_pcm_info_t *pcminfo;
+snd_pcm_info_alloca( &pcminfo );
+snd_pcm_t *phandle;
+snd_pcm_hw_params_t *params;
+snd_pcm_hw_params_alloca( &params );
+// First try for playback unless default device (which has subdev -1)
+stream = SND_PCM_STREAM_PLAYBACK;
+snd_pcm_info_set_stream( pcminfo, stream );
+if ( subdevice != -1 ) {
+snd_pcm_info_set_device( pcminfo, subdevice );
+snd_pcm_info_set_subdevice( pcminfo, 0 );
+result = snd_ctl_pcm_info( chandle, pcminfo );
+if ( result < 0 ) {
+// Device probably doesn't support playback.
+goto captureProbe;
+}
+}
+result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK );
+if ( result < 0 ) {
+errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
+errorText_ = errorStream_.str();
+error( RtAudioError::WARNING );
+goto captureProbe;
+}
+// The device is open ... fill the parameter structure.
+result = snd_pcm_hw_params_any( phandle, params );
+if ( result < 0 ) {
+snd_pcm_close( phandle );
+errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
+errorText_ = errorStream_.str();
+error( RtAudioError::WARNING );
+goto captureProbe;
+}
+// Get output channel information.
+unsigned int value;
+result = snd_pcm_hw_params_get_channels_max( params, &value );
+if ( result < 0 ) {
+snd_pcm_close( phandle );
+errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << ".";
+errorText_ = errorStream_.str();
+error( RtAudioError::WARNING );
+goto captureProbe;
+}
+info.outputChannels = value;
+snd_pcm_close( phandle );
+captureProbe:
+stream = SND_PCM_STREAM_CAPTURE;
+snd_pcm_info_set_stream( pcminfo, stream );
+// Now try for capture unless default device (with subdev = -1)
+if ( subdevice != -1 ) {
+result = snd_ctl_pcm_info( chandle, pcminfo );
+snd_ctl_close( chandle );
+if ( result < 0 ) {
+// Device probably doesn't support capture.
+if ( info.outputChannels == 0 ) return info;
+goto probeParameters;
+}
+}
+else
+snd_ctl_close( chandle );
+result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
+if ( result < 0 ) {
+errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
+errorText_ = errorStream_.str();
+error( RtAudioError::WARNING );
+if ( info.outputChannels == 0 ) return info;
+goto probeParameters;
+}
+// The device is open ... fill the parameter structure.
+result = snd_pcm_hw_params_any( phandle, params );
+if ( result < 0 ) {
+snd_pcm_close( phandle );
+errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
+errorText_ = errorStream_.str();
+error( RtAudioError::WARNING );
+if ( info.outputChannels == 0 ) return info;
+goto probeParameters;
+}
+result = snd_pcm_hw_params_get_channels_max( params, &value );
+if ( result < 0 ) {
+snd_pcm_close( phandle );
+errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << ".";
+errorText_ = errorStream_.str();
+error( RtAudioError::WARNING );
+if ( info.outputChannels == 0 ) return info;
+goto probeParameters;
+}
+info.inputChannels = value;
+snd_pcm_close( phandle );
+// If device opens for both playback and capture, we determine the channels.
+if ( info.outputChannels > 0 && info.inputChannels > 0 )
+info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
+// ALSA doesn't provide default devices so we'll use the first available one.
+if ( device == 0 && info.outputChannels > 0 )
+info.isDefaultOutput = true;
+if ( device == 0 && info.inputChannels > 0 )
+info.isDefaultInput = true;
+probeParameters:
+// At this point, we just need to figure out the supported data
+// formats and sample rates.  We'll proceed by opening the device in
+// the direction with the maximum number of channels, or playback if
+// they are equal.  This might limit our sample rate options, but so
+// be it.
+if ( info.outputChannels >= info.inputChannels )
+stream = SND_PCM_STREAM_PLAYBACK;
+else
+stream = SND_PCM_STREAM_CAPTURE;
+snd_pcm_info_set_stream( pcminfo, stream );
+result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
+if ( result < 0 ) {
+errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
+errorText_ = errorStream_.str();
+error( RtAudioError::WARNING );
+return info;
+}
+// The device is open ... fill the parameter structure.
+result = snd_pcm_hw_params_any( phandle, params );
+if ( result < 0 ) {
+snd_pcm_close( phandle );
+errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
+errorText_ = errorStream_.str();
+error( RtAudioError::WARNING );
+return info;
+}
+// Test our discrete set of sample rate values.
+info.sampleRates.clear();
+for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {
+if ( snd_pcm_hw_params_test_rate( phandle, params, SAMPLE_RATES[i], 0 ) == 0 ) {
+info.sampleRates.push_back( SAMPLE_RATES[i] );
+if ( !info.preferredSampleRate || ( SAMPLE_RATES[i] <= 48000 && SAMPLE_RATES[i] > info.preferredSampleRate ) )
+info.preferredSampleRate = SAMPLE_RATES[i];
+}
+}
+if ( info.sampleRates.size() == 0 ) {
+snd_pcm_close( phandle );
+errorStream_ << "RtApiAlsa::getDeviceInfo: no supported sample rates found for device (" << name << ").";
+errorText_ = errorStream_.str();
+error( RtAudioError::WARNING );
+return info;
+}
+// Probe the supported data formats ... we don't care about endian-ness just yet
+snd_pcm_format_t format;
+info.nativeFormats = 0;
+format = SND_PCM_FORMAT_S8;
+if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
+info.nativeFormats |= RTAUDIO_SINT8;
+format = SND_PCM_FORMAT_S16;
+if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
+info.nativeFormats |= RTAUDIO_SINT16;
+format = SND_PCM_FORMAT_S24;
+if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
+info.nativeFormats |= RTAUDIO_SINT24;
+format = SND_PCM_FORMAT_S32;
+if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
+info.nativeFormats |= RTAUDIO_SINT32;
+format = SND_PCM_FORMAT_FLOAT;
+if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
+info.nativeFormats |= RTAUDIO_FLOAT32;
+format = SND_PCM_FORMAT_FLOAT64;
+if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
+info.nativeFormats |= RTAUDIO_FLOAT64;
+// Check that we have at least one supported format
+if ( info.nativeFormats == 0 ) {
+snd_pcm_close( phandle );
+errorStream_ << "RtApiAlsa::getDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio.";
+errorText_ = errorStream_.str();
+error( RtAudioError::WARNING );
+return info;
+}
+// Get the device name
+if (strncmp(name, "default", 7)!=0) {
+char *cardname;
+result = snd_card_get_name( card, &cardname );
+if ( result >= 0 ) {
+sprintf( name, "hw:%s,%d", cardname, subdevice );
+free( cardname );
+}
+}
+info.name = name;
+// That's all ... close the device and return
+snd_pcm_close( phandle );
+info.probed = true;
+return info;
+}
+void RtApiAlsa :: saveDeviceInfo( void )
+{
+devices_.clear();
+unsigned int nDevices = getDeviceCount();
+devices_.resize( nDevices );
+for ( unsigned int i=0; i<nDevices; i++ )
+devices_[i] = getDeviceInfo( i );
+}
+bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+unsigned int firstChannel, unsigned int sampleRate,
+RtAudioFormat format, unsigned int *bufferSize,
+RtAudio::StreamOptions *options )
+{
+#if defined(__RTAUDIO_DEBUG__)
+struct SndOutputTdealloc {
+SndOutputTdealloc() : _out(NULL) { snd_output_stdio_attach(&_out, stderr, 0); }
+~SndOutputTdealloc() { snd_output_close(_out); }
+operator snd_output_t*() { return _out; }
+snd_output_t *_out;
+} out;
+#endif
+// I'm not using the "plug" interface ... too much inconsistent behavior.
+unsigned nDevices = 0;
+int result, subdevice, card;
+char name[64];
+snd_ctl_t *chandle;
+if ( device == 0
+|| (options && options->flags & RTAUDIO_ALSA_USE_DEFAULT) )
+{
+strcpy(name, "default");
+result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
+if ( result == 0 ) {
+if ( nDevices == device ) {
+strcpy( name, "default" );
+snd_ctl_close( chandle );
+goto foundDevice;
+}
+nDevices++;
+}
+}
+else {
+nDevices++;
+// Count cards and devices
+card = -1;
+snd_card_next( &card );
+while ( card >= 0 ) {
+sprintf( name, "hw:%d", card );
+result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
+if ( result < 0 ) {
+errorStream_ << "RtApiAlsa::probeDeviceOpen: control open, card = " << card << ", " << snd_strerror( result ) << ".";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+subdevice = -1;
+while( 1 ) {
+result = snd_ctl_pcm_next_device( chandle, &subdevice );
+if ( result < 0 ) break;
+if ( subdevice < 0 ) break;
+if ( nDevices == device ) {
+sprintf( name, "hw:%d,%d", card, subdevice );
+snd_ctl_close( chandle );
+goto foundDevice;
+}
+nDevices++;
+}
+snd_ctl_close( chandle );
+snd_card_next( &card );
+}
+if ( nDevices == 0 ) {
+// This should not happen because a check is made before this function is called.
+errorText_ = "RtApiAlsa::probeDeviceOpen: no devices found!";
+return FAILURE;
+}
+if ( device >= nDevices ) {
+// This should not happen because a check is made before this function is called.
+errorText_ = "RtApiAlsa::probeDeviceOpen: device ID is invalid!";
+return FAILURE;
+}
+}
+foundDevice:
+// The getDeviceInfo() function will not work for a device that is
+// already open.  Thus, we'll probe the system before opening a
+// stream and save the results for use by getDeviceInfo().
+if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) // only do once
+this->saveDeviceInfo();
+snd_pcm_stream_t stream;
+if ( mode == OUTPUT )
+stream = SND_PCM_STREAM_PLAYBACK;
+else
+stream = SND_PCM_STREAM_CAPTURE;
+snd_pcm_t *phandle;
+int openMode = SND_PCM_ASYNC;
+result = snd_pcm_open( &phandle, name, stream, openMode );
+if ( result < 0 ) {
+if ( mode == OUTPUT )
+errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for output.";
+else
+errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for input.";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+// Fill the parameter structure.
+snd_pcm_hw_params_t *hw_params;
+snd_pcm_hw_params_alloca( &hw_params );
+result = snd_pcm_hw_params_any( phandle, hw_params );
+if ( result < 0 ) {
+snd_pcm_close( phandle );
+errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") parameters, " << snd_strerror( result ) << ".";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+#if defined(__RTAUDIO_DEBUG__)
+fprintf( stderr, "\nRtApiAlsa: dump hardware params just after device open:\n\n" );
+snd_pcm_hw_params_dump( hw_params, out );
+#endif
+// Set access ... check user preference.
+if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) {
+stream_.userInterleaved = false;
+result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
+if ( result < 0 ) {
+result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
+stream_.deviceInterleaved[mode] =  true;
+}
+else
+stream_.deviceInterleaved[mode] = false;
+}
+else {
+stream_.userInterleaved = true;
+result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
+if ( result < 0 ) {
+result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
+stream_.deviceInterleaved[mode] =  false;
+}
+else
+stream_.deviceInterleaved[mode] =  true;
+}
+if ( result < 0 ) {
+snd_pcm_close( phandle );
+errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") access, " << snd_strerror( result ) << ".";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+// Determine how to set the device format.
+stream_.userFormat = format;
+snd_pcm_format_t deviceFormat = SND_PCM_FORMAT_UNKNOWN;
+if ( format == RTAUDIO_SINT8 )
+deviceFormat = SND_PCM_FORMAT_S8;
+else if ( format == RTAUDIO_SINT16 )
+deviceFormat = SND_PCM_FORMAT_S16;
+else if ( format == RTAUDIO_SINT24 )
+deviceFormat = SND_PCM_FORMAT_S24;
+else if ( format == RTAUDIO_SINT32 )
+deviceFormat = SND_PCM_FORMAT_S32;
+else if ( format == RTAUDIO_FLOAT32 )
+deviceFormat = SND_PCM_FORMAT_FLOAT;
+else if ( format == RTAUDIO_FLOAT64 )
+deviceFormat = SND_PCM_FORMAT_FLOAT64;
+if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat) == 0) {
+stream_.deviceFormat[mode] = format;
+goto setFormat;
+}
+// The user requested format is not natively supported by the device.
+deviceFormat = SND_PCM_FORMAT_FLOAT64;
+if ( snd_pcm_hw_params_test_format( phandle, hw_params, deviceFormat ) == 0 ) {
+stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
+goto setFormat;
+}
+deviceFormat = SND_PCM_FORMAT_FLOAT;
+if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
+stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
+goto setFormat;
+}
+deviceFormat = SND_PCM_FORMAT_S32;
+if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
+stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+goto setFormat;
+}
+deviceFormat = SND_PCM_FORMAT_S24;
+if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
+stream_.deviceFormat[mode] = RTAUDIO_SINT24;
+goto setFormat;
+}
+deviceFormat = SND_PCM_FORMAT_S16;
+if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
+stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+goto setFormat;
+}
+deviceFormat = SND_PCM_FORMAT_S8;
+if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
+stream_.deviceFormat[mode] = RTAUDIO_SINT8;
+goto setFormat;
+}
+// If we get here, no supported format was found.
+snd_pcm_close( phandle );
+errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device " << device << " data format not supported by RtAudio.";
+errorText_ = errorStream_.str();
+return FAILURE;
+setFormat:
+result = snd_pcm_hw_params_set_format( phandle, hw_params, deviceFormat );
+if ( result < 0 ) {
+snd_pcm_close( phandle );
+errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") data format, " << snd_strerror( result ) << ".";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+// Determine whether byte-swaping is necessary.
+stream_.doByteSwap[mode] = false;
+if ( deviceFormat != SND_PCM_FORMAT_S8 ) {
+result = snd_pcm_format_cpu_endian( deviceFormat );
+if ( result == 0 )
+stream_.doByteSwap[mode] = true;
+else if (result < 0) {
+snd_pcm_close( phandle );
+errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") endian-ness, " << snd_strerror( result ) << ".";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+}
+// Set the sample rate.
+result = snd_pcm_hw_params_set_rate_near( phandle, hw_params, (unsigned int*) &sampleRate, 0 );
+if ( result < 0 ) {
+snd_pcm_close( phandle );
+errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting sample rate on device (" << name << "), " << snd_strerror( result ) << ".";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+// Determine the number of channels for this device.  We support a possible
+// minimum device channel number > than the value requested by the user.
+stream_.nUserChannels[mode] = channels;
+unsigned int value;
+result = snd_pcm_hw_params_get_channels_max( hw_params, &value );
+unsigned int deviceChannels = value;
+if ( result < 0 || deviceChannels < channels + firstChannel ) {
+snd_pcm_close( phandle );
+errorStream_ << "RtApiAlsa::probeDeviceOpen: requested channel parameters not supported by device (" << name << "), " << snd_strerror( result ) << ".";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+result = snd_pcm_hw_params_get_channels_min( hw_params, &value );
+if ( result < 0 ) {
+snd_pcm_close( phandle );
+errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting minimum channels for device (" << name << "), " << snd_strerror( result ) << ".";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+deviceChannels = value;
+if ( deviceChannels < channels + firstChannel ) deviceChannels = channels + firstChannel;
+stream_.nDeviceChannels[mode] = deviceChannels;
+// Set the device channels.
+result = snd_pcm_hw_params_set_channels( phandle, hw_params, deviceChannels );
+if ( result < 0 ) {
+snd_pcm_close( phandle );
+errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting channels for device (" << name << "), " << snd_strerror( result ) << ".";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+// Set the buffer (or period) size.
+int dir = 0;
+snd_pcm_uframes_t periodSize = *bufferSize;
+result = snd_pcm_hw_params_set_period_size_near( phandle, hw_params, &periodSize, &dir );
+if ( result < 0 ) {
+snd_pcm_close( phandle );
+errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting period size for device (" << name << "), " << snd_strerror( result ) << ".";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+*bufferSize = periodSize;
+// Set the buffer number, which in ALSA is referred to as the "period".
+unsigned int periods = 0;
+if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2;
+if ( options && options->numberOfBuffers > 0 ) periods = options->numberOfBuffers;
+if ( periods < 2 ) periods = 4; // a fairly safe default value
+result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir );
+if ( result < 0 ) {
+snd_pcm_close( phandle );
+errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << ".";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+// If attempting to setup a duplex stream, the bufferSize parameter
+// MUST be the same in both directions!
+if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {
+snd_pcm_close( phandle );
+errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ").";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+stream_.bufferSize = *bufferSize;
+// Install the hardware configuration
+result = snd_pcm_hw_params( phandle, hw_params );
+if ( result < 0 ) {
+snd_pcm_close( phandle );
+errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing hardware configuration on device (" << name << "), " << snd_strerror( result ) << ".";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+#if defined(__RTAUDIO_DEBUG__)
+fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n");
+snd_pcm_hw_params_dump( hw_params, out );
+#endif
+// Set the software configuration to fill buffers with zeros and prevent device stopping on xruns.
+snd_pcm_sw_params_t *sw_params = NULL;
+snd_pcm_sw_params_alloca( &sw_params );
+snd_pcm_sw_params_current( phandle, sw_params );
+snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize );
+snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, ULONG_MAX );
+snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 );
+// The following two settings were suggested by Theo Veenker
+//snd_pcm_sw_params_set_avail_min( phandle, sw_params, *bufferSize );
+//snd_pcm_sw_params_set_xfer_align( phandle, sw_params, 1 );
+// here are two options for a fix
+//snd_pcm_sw_params_set_silence_size( phandle, sw_params, ULONG_MAX );
+snd_pcm_uframes_t val;
+snd_pcm_sw_params_get_boundary( sw_params, &val );
+snd_pcm_sw_params_set_silence_size( phandle, sw_params, val );
+result = snd_pcm_sw_params( phandle, sw_params );
+if ( result < 0 ) {
+snd_pcm_close( phandle );
+errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing software configuration on device (" << name << "), " << snd_strerror( result ) << ".";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+#if defined(__RTAUDIO_DEBUG__)
+fprintf(stderr, "\nRtApiAlsa: dump software params after installation:\n\n");
+snd_pcm_sw_params_dump( sw_params, out );
+#endif
+// Set flags for buffer conversion
+stream_.doConvertBuffer[mode] = false;
+if ( stream_.userFormat != stream_.deviceFormat[mode] )
+stream_.doConvertBuffer[mode] = true;
+if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
+stream_.doConvertBuffer[mode] = true;
+if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+stream_.nUserChannels[mode] > 1 )
+stream_.doConvertBuffer[mode] = true;
+// Allocate the ApiHandle if necessary and then save.
+AlsaHandle *apiInfo = 0;
+if ( stream_.apiHandle == 0 ) {
+try {
+apiInfo = (AlsaHandle *) new AlsaHandle;
+}
+catch ( std::bad_alloc& ) {
+errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating AlsaHandle memory.";
+goto error;
+}
+if ( pthread_cond_init( &apiInfo->runnable_cv, NULL ) ) {
+errorText_ = "RtApiAlsa::probeDeviceOpen: error initializing pthread condition variable.";
+goto error;
+}
+stream_.apiHandle = (void *) apiInfo;
+apiInfo->handles[0] = 0;
+apiInfo->handles[1] = 0;
+}
+else {
+apiInfo = (AlsaHandle *) stream_.apiHandle;
+}
+apiInfo->handles[mode] = phandle;
+phandle = 0;
+// Allocate necessary internal buffers.
+unsigned long bufferBytes;
+bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+if ( stream_.userBuffer[mode] == NULL ) {
+errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating user buffer memory.";
+goto error;
+}
+if ( stream_.doConvertBuffer[mode] ) {
+bool makeBuffer = true;
+bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
+if ( mode == INPUT ) {
+if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
+unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
+if ( bufferBytes <= bytesOut ) makeBuffer = false;
+}
+}
+if ( makeBuffer ) {
+bufferBytes *= *bufferSize;
+if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
+stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
+if ( stream_.deviceBuffer == NULL ) {
+errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating device buffer memory.";
+goto error;
+}
+}
+}
+stream_.sampleRate = sampleRate;
+stream_.nBuffers = periods;
+stream_.device[mode] = device;
+stream_.state = STREAM_STOPPED;
+// Setup the buffer conversion information structure.
+if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
+// Setup thread if necessary.
+if ( stream_.mode == OUTPUT && mode == INPUT ) {
+// We had already set up an output stream.
+stream_.mode = DUPLEX;
+// Link the streams if possible.
+apiInfo->synchronized = false;
+if ( snd_pcm_link( apiInfo->handles[0], apiInfo->handles[1] ) == 0 )
+apiInfo->synchronized = true;
+else {
+errorText_ = "RtApiAlsa::probeDeviceOpen: unable to synchronize input and output devices.";
+error( RtAudioError::WARNING );
+}
+}
+else {
+stream_.mode = mode;
+// Setup callback thread.
+stream_.callbackInfo.object = (void *) this;
+// Set the thread attributes for joinable and realtime scheduling
+// priority (optional).  The higher priority will only take affect
+// if the program is run as root or suid. Note, under Linux
+// processes with CAP_SYS_NICE privilege, a user can change
+// scheduling policy and priority (thus need not be root). See
+// POSIX "capabilities".
+pthread_attr_t attr;
+pthread_attr_init( &attr );
+pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
+#ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
+if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
+stream_.callbackInfo.doRealtime = true;
+struct sched_param param;
+int priority = options->priority;
+int min = sched_get_priority_min( SCHED_RR );
+int max = sched_get_priority_max( SCHED_RR );
+if ( priority < min ) priority = min;
+else if ( priority > max ) priority = max;
+param.sched_priority = priority;
+// Set the policy BEFORE the priority. Otherwise it fails.
+pthread_attr_setschedpolicy(&attr, SCHED_RR);
+pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM);
+// This is definitely required. Otherwise it fails.
+pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED);
+pthread_attr_setschedparam(&attr, &param);
+}
+else
+pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
+#else
+pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
+#endif
+stream_.callbackInfo.isRunning = true;
+result = pthread_create( &stream_.callbackInfo.thread, &attr, alsaCallbackHandler, &stream_.callbackInfo );
+pthread_attr_destroy( &attr );
+if ( result ) {
+// Failed. Try instead with default attributes.
+result = pthread_create( &stream_.callbackInfo.thread, NULL, alsaCallbackHandler, &stream_.callbackInfo );
+if ( result ) {
+stream_.callbackInfo.isRunning = false;
+errorText_ = "RtApiAlsa::error creating callback thread!";
+goto error;
+}
+}
+}
+return SUCCESS;
+error:
+if ( apiInfo ) {
+pthread_cond_destroy( &apiInfo->runnable_cv );
+if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
+if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
+delete apiInfo;
+stream_.apiHandle = 0;
+}
+if ( phandle) snd_pcm_close( phandle );
+for ( int i=0; i<2; i++ ) {
+if ( stream_.userBuffer[i] ) {
+free( stream_.userBuffer[i] );
+stream_.userBuffer[i] = 0;
+}
+}
+if ( stream_.deviceBuffer ) {
+free( stream_.deviceBuffer );
+stream_.deviceBuffer = 0;
+}
+stream_.state = STREAM_CLOSED;
+return FAILURE;
+}
+void RtApiAlsa :: closeStream()
+{
+if ( stream_.state == STREAM_CLOSED ) {
+errorText_ = "RtApiAlsa::closeStream(): no open stream to close!";
+error( RtAudioError::WARNING );
+return;
+}
+AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
+stream_.callbackInfo.isRunning = false;
+MUTEX_LOCK( &stream_.mutex );
+if ( stream_.state == STREAM_STOPPED ) {
+apiInfo->runnable = true;
+pthread_cond_signal( &apiInfo->runnable_cv );
+}
+MUTEX_UNLOCK( &stream_.mutex );
+pthread_join( stream_.callbackInfo.thread, NULL );
+if ( stream_.state == STREAM_RUNNING ) {
+stream_.state = STREAM_STOPPED;
+if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
+snd_pcm_drop( apiInfo->handles[0] );
+if ( stream_.mode == INPUT || stream_.mode == DUPLEX )
+snd_pcm_drop( apiInfo->handles[1] );
+}
+if ( apiInfo ) {
+pthread_cond_destroy( &apiInfo->runnable_cv );
+if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
+if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
+delete apiInfo;
+stream_.apiHandle = 0;
+}
+for ( int i=0; i<2; i++ ) {
+if ( stream_.userBuffer[i] ) {
+free( stream_.userBuffer[i] );
+stream_.userBuffer[i] = 0;
+}
+}
+if ( stream_.deviceBuffer ) {
+free( stream_.deviceBuffer );
+stream_.deviceBuffer = 0;
+}
+stream_.mode = UNINITIALIZED;
+stream_.state = STREAM_CLOSED;
+}
+void RtApiAlsa :: startStream()
+{
+// This method calls snd_pcm_prepare if the device isn't already in that state.
+verifyStream();
+if ( stream_.state == STREAM_RUNNING ) {
+errorText_ = "RtApiAlsa::startStream(): the stream is already running!";
+error( RtAudioError::WARNING );
+return;
+}
+MUTEX_LOCK( &stream_.mutex );
+#if defined( HAVE_GETTIMEOFDAY )
+gettimeofday( &stream_.lastTickTimestamp, NULL );
+#endif
+int result = 0;
+snd_pcm_state_t state;
+AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
+snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
+if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+state = snd_pcm_state( handle[0] );
+if ( state != SND_PCM_STATE_PREPARED ) {
+result = snd_pcm_prepare( handle[0] );
+if ( result < 0 ) {
+errorStream_ << "RtApiAlsa::startStream: error preparing output pcm device, " << snd_strerror( result ) << ".";
+errorText_ = errorStream_.str();
+goto unlock;
+}
+}
+}
+if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
+result = snd_pcm_drop(handle[1]); // fix to remove stale data received since device has been open
+state = snd_pcm_state( handle[1] );
+if ( state != SND_PCM_STATE_PREPARED ) {
+result = snd_pcm_prepare( handle[1] );
+if ( result < 0 ) {
+errorStream_ << "RtApiAlsa::startStream: error preparing input pcm device, " << snd_strerror( result ) << ".";
+errorText_ = errorStream_.str();
+goto unlock;
+}
+}
+}
+stream_.state = STREAM_RUNNING;
+unlock:
+apiInfo->runnable = true;
+pthread_cond_signal( &apiInfo->runnable_cv );
+MUTEX_UNLOCK( &stream_.mutex );
+if ( result >= 0 ) return;
+error( RtAudioError::SYSTEM_ERROR );
+}
+void RtApiAlsa :: stopStream()
+{
+verifyStream();
+if ( stream_.state == STREAM_STOPPED ) {
+errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!";
+error( RtAudioError::WARNING );
+return;
+}
+stream_.state = STREAM_STOPPED;
+MUTEX_LOCK( &stream_.mutex );
+int result = 0;
+AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
+snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
+if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+if ( apiInfo->synchronized )
+result = snd_pcm_drop( handle[0] );
+else
+result = snd_pcm_drain( handle[0] );
+if ( result < 0 ) {
+errorStream_ << "RtApiAlsa::stopStream: error draining output pcm device, " << snd_strerror( result ) << ".";
+errorText_ = errorStream_.str();
+goto unlock;
+}
+}
+if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
+result = snd_pcm_drop( handle[1] );
+if ( result < 0 ) {
+errorStream_ << "RtApiAlsa::stopStream: error stopping input pcm device, " << snd_strerror( result ) << ".";
+errorText_ = errorStream_.str();
+goto unlock;
+}
+}
+unlock:
+apiInfo->runnable = false; // fixes high CPU usage when stopped
+MUTEX_UNLOCK( &stream_.mutex );
+if ( result >= 0 ) return;
+error( RtAudioError::SYSTEM_ERROR );
+}
+void RtApiAlsa :: abortStream()
+{
+verifyStream();
+if ( stream_.state == STREAM_STOPPED ) {
+errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!";
+error( RtAudioError::WARNING );
+return;
+}
+stream_.state = STREAM_STOPPED;
+MUTEX_LOCK( &stream_.mutex );
+int result = 0;
+AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
+snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
+if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+result = snd_pcm_drop( handle[0] );
+if ( result < 0 ) {
+errorStream_ << "RtApiAlsa::abortStream: error aborting output pcm device, " << snd_strerror( result ) << ".";
+errorText_ = errorStream_.str();
+goto unlock;
+}
+}
+if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
+result = snd_pcm_drop( handle[1] );
+if ( result < 0 ) {
+errorStream_ << "RtApiAlsa::abortStream: error aborting input pcm device, " << snd_strerror( result ) << ".";
+errorText_ = errorStream_.str();
+goto unlock;
+}
+}
+unlock:
+apiInfo->runnable = false; // fixes high CPU usage when stopped
+MUTEX_UNLOCK( &stream_.mutex );
+if ( result >= 0 ) return;
+error( RtAudioError::SYSTEM_ERROR );
+}
+void RtApiAlsa :: callbackEvent()
+{
+AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
+if ( stream_.state == STREAM_STOPPED ) {
+MUTEX_LOCK( &stream_.mutex );
+while ( !apiInfo->runnable )
+pthread_cond_wait( &apiInfo->runnable_cv, &stream_.mutex );
+if ( stream_.state != STREAM_RUNNING ) {
+MUTEX_UNLOCK( &stream_.mutex );
+return;
+}
+MUTEX_UNLOCK( &stream_.mutex );
+}
+if ( stream_.state == STREAM_CLOSED ) {
+errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!";
+error( RtAudioError::WARNING );
+return;
+}
+int doStopStream = 0;
+RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
+double streamTime = getStreamTime();
+RtAudioStreamStatus status = 0;
+if ( stream_.mode != INPUT && apiInfo->xrun[0] == true ) {
+status |= RTAUDIO_OUTPUT_UNDERFLOW;
+apiInfo->xrun[0] = false;
+}
+if ( stream_.mode != OUTPUT && apiInfo->xrun[1] == true ) {
+status |= RTAUDIO_INPUT_OVERFLOW;
+apiInfo->xrun[1] = false;
+}
+doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
+if ( doStopStream == 2 ) {
+abortStream();
+return;
+}
+MUTEX_LOCK( &stream_.mutex );
+// The state might change while waiting on a mutex.
+if ( stream_.state == STREAM_STOPPED ) goto unlock;
+int result;
+char *buffer;
+int channels;
+snd_pcm_t **handle;
+snd_pcm_sframes_t frames;
+RtAudioFormat format;
+handle = (snd_pcm_t **) apiInfo->handles;
+if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+// Setup parameters.
+if ( stream_.doConvertBuffer[1] ) {
+buffer = stream_.deviceBuffer;
+channels = stream_.nDeviceChannels[1];
+format = stream_.deviceFormat[1];
+}
+else {
+buffer = stream_.userBuffer[1];
+channels = stream_.nUserChannels[1];
+format = stream_.userFormat;
+}
+// Read samples from device in interleaved/non-interleaved format.
+if ( stream_.deviceInterleaved[1] )
+result = snd_pcm_readi( handle[1], buffer, stream_.bufferSize );
+else {
+void *bufs[channels];
+size_t offset = stream_.bufferSize * formatBytes( format );
+for ( int i=0; i<channels; i++ )
+bufs[i] = (void *) (buffer + (i * offset));
+result = snd_pcm_readn( handle[1], bufs, stream_.bufferSize );
+}
+if ( result < (int) stream_.bufferSize ) {
+// Either an error or overrun occured.
+if ( result == -EPIPE ) {
+snd_pcm_state_t state = snd_pcm_state( handle[1] );
+if ( state == SND_PCM_STATE_XRUN ) {
+apiInfo->xrun[1] = true;
+result = snd_pcm_prepare( handle[1] );
+if ( result < 0 ) {
+errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after overrun, " << snd_strerror( result ) << ".";
+errorText_ = errorStream_.str();
+}
+}
+else {
+errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
+errorText_ = errorStream_.str();
+}
+}
+else {
+errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << ".";
+errorText_ = errorStream_.str();
+}
+error( RtAudioError::WARNING );
+goto tryOutput;
+}
+// Do byte swapping if necessary.
+if ( stream_.doByteSwap[1] )
+byteSwapBuffer( buffer, stream_.bufferSize * channels, format );
+// Do buffer conversion if necessary.
+if ( stream_.doConvertBuffer[1] )
+convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
+// Check stream latency
+result = snd_pcm_delay( handle[1], &frames );
+if ( result == 0 && frames > 0 ) stream_.latency[1] = frames;
+}
+tryOutput:
+if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+// Setup parameters and do buffer conversion if necessary.
+if ( stream_.doConvertBuffer[0] ) {
+buffer = stream_.deviceBuffer;
+convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
+channels = stream_.nDeviceChannels[0];
+format = stream_.deviceFormat[0];
+}
+else {
+buffer = stream_.userBuffer[0];
+channels = stream_.nUserChannels[0];
+format = stream_.userFormat;
+}
+// Do byte swapping if necessary.
+if ( stream_.doByteSwap[0] )
+byteSwapBuffer(buffer, stream_.bufferSize * channels, format);
+// Write samples to device in interleaved/non-interleaved format.
+if ( stream_.deviceInterleaved[0] )
+result = snd_pcm_writei( handle[0], buffer, stream_.bufferSize );
+else {
+void *bufs[channels];
+size_t offset = stream_.bufferSize * formatBytes( format );
+for ( int i=0; i<channels; i++ )
+bufs[i] = (void *) (buffer + (i * offset));
+result = snd_pcm_writen( handle[0], bufs, stream_.bufferSize );
+}
+if ( result < (int) stream_.bufferSize ) {
+// Either an error or underrun occured.
+if ( result == -EPIPE ) {
+snd_pcm_state_t state = snd_pcm_state( handle[0] );
+if ( state == SND_PCM_STATE_XRUN ) {
+apiInfo->xrun[0] = true;
+result = snd_pcm_prepare( handle[0] );
+if ( result < 0 ) {
+errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after underrun, " << snd_strerror( result ) << ".";
+errorText_ = errorStream_.str();
+}
+else
+errorText_ =  "RtApiAlsa::callbackEvent: audio write error, underrun.";
+}
+else {
+errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
+errorText_ = errorStream_.str();
+}
+}
+else {
+errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << ".";
+errorText_ = errorStream_.str();
+}
+error( RtAudioError::WARNING );
+goto unlock;
+}
+// Check stream latency
+result = snd_pcm_delay( handle[0], &frames );
+if ( result == 0 && frames > 0 ) stream_.latency[0] = frames;
+}
+unlock:
+MUTEX_UNLOCK( &stream_.mutex );
+RtApi::tickStreamTime();
+if ( doStopStream == 1 ) this->stopStream();
+}
+static void *alsaCallbackHandler( void *ptr )
+{
+CallbackInfo *info = (CallbackInfo *) ptr;
+RtApiAlsa *object = (RtApiAlsa *) info->object;
+bool *isRunning = &info->isRunning;
+#ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
+if ( info->doRealtime ) {
+std::cerr << "RtAudio alsa: " <<
+(sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") <<
+"running realtime scheduling" << std::endl;
+}
+#endif
+while ( *isRunning == true ) {
+pthread_testcancel();
+object->callbackEvent();
+}
+pthread_exit( NULL );
+}
+//******************** End of __LINUX_ALSA__ *********************//
+#endif
+#if defined(__LINUX_PULSE__)
+// Code written by Peter Meerwald, pmeerw@pmeerw.net
+// and Tristan Matthews.
+#include <pulse/error.h>
+#include <pulse/simple.h>
+#include <pulse/pulseaudio.h>
+#include <cstdio>
+static pa_mainloop_api *rt_pa_mainloop_api = NULL;
+struct PaDeviceInfo {
+PaDeviceInfo() : sink_index(-1), source_index(-1) {}
+int sink_index;
+int source_index;
+std::string sink_name;
+std::string source_name;
+RtAudio::DeviceInfo info;
+};
+static struct {
+std::vector<PaDeviceInfo> dev;
+std::string default_sink_name;
+std::string default_source_name;
+int default_rate;
+} rt_pa_info;
+static const unsigned int SUPPORTED_SAMPLERATES[] = { 8000, 16000, 22050, 32000,
+44100, 48000, 96000, 0};
+struct rtaudio_pa_format_mapping_t {
+RtAudioFormat rtaudio_format;
+pa_sample_format_t pa_format;
+};
+static const rtaudio_pa_format_mapping_t supported_sampleformats[] = {
+{RTAUDIO_SINT16, PA_SAMPLE_S16LE},
+{RTAUDIO_SINT24, PA_SAMPLE_S24LE},
+{RTAUDIO_SINT32, PA_SAMPLE_S32LE},
+{RTAUDIO_FLOAT32, PA_SAMPLE_FLOAT32LE},
+{0, PA_SAMPLE_INVALID}};
+struct PulseAudioHandle {
+pa_simple *s_play;
+pa_simple *s_rec;
+pthread_t thread;
+pthread_cond_t runnable_cv;
+bool runnable;
+PulseAudioHandle() : s_play(0), s_rec(0), runnable(false) { }
+};
+static void rt_pa_mainloop_api_quit(int ret) {
+rt_pa_mainloop_api->quit(rt_pa_mainloop_api, ret);
+}
+static void rt_pa_server_callback(pa_context *context, const pa_server_info *info, void *data){
+(void)context;
+(void)data;
+pa_sample_spec ss;
+if (!info)
+rt_pa_mainloop_api_quit(1);
+ss = info->sample_spec;
+rt_pa_info.default_rate = ss.rate;
+rt_pa_info.default_sink_name = info->default_sink_name;
+rt_pa_info.default_source_name = info->default_source_name;
+rt_pa_mainloop_api_quit(0);
+}
+static void rt_pa_sink_info_cb(pa_context * /*c*/, const pa_sink_info *i,
+int eol, void * /*userdata*/)
+{
+if (eol) return;
+PaDeviceInfo inf;
+inf.info.name = pa_proplist_gets(i->proplist, "device.description");
+inf.info.probed = true;
+inf.info.outputChannels = i->sample_spec.channels;
+inf.info.preferredSampleRate = i->sample_spec.rate;
+inf.info.isDefaultOutput = (rt_pa_info.default_sink_name == i->name);
+inf.sink_index = i->index;
+inf.sink_name = i->name;
+for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr )
+inf.info.sampleRates.push_back( *sr );
+for ( const rtaudio_pa_format_mapping_t *fm = supported_sampleformats;
+fm->rtaudio_format; ++fm )
+inf.info.nativeFormats |= fm->rtaudio_format;
+for (size_t i=0; i < rt_pa_info.dev.size(); i++)
+{
+/* Attempt to match up sink and source records by device description. */
+if (rt_pa_info.dev[i].info.name == inf.info.name) {
+rt_pa_info.dev[i].sink_index = inf.sink_index;
+rt_pa_info.dev[i].sink_name = inf.sink_name;
+rt_pa_info.dev[i].info.outputChannels = inf.info.outputChannels;
+rt_pa_info.dev[i].info.isDefaultOutput = inf.info.isDefaultOutput;
+/* Assume duplex channels are minimum of input and output channels. */
+/* Uncomment if we add support for DUPLEX
+if (rt_pa_info.dev[i].source_index > -1)
+(inf.info.outputChannels < rt_pa_info.dev[i].info.inputChannels)
+? inf.info.outputChannels : rt_pa_info.dev[i].info.inputChannels;
+*/
+return;
+}
+}
+/* try to ensure device #0 is the default */
+if (inf.info.isDefaultOutput)
+rt_pa_info.dev.insert(rt_pa_info.dev.begin(), inf);
+else
+rt_pa_info.dev.push_back(inf);
+}
+static void rt_pa_source_info_cb(pa_context * /*c*/, const pa_source_info *i,
+int eol, void * /*userdata*/)
+{
+if (eol) return;
+PaDeviceInfo inf;
+inf.info.name = pa_proplist_gets(i->proplist, "device.description");
+inf.info.probed = true;
+inf.info.inputChannels = i->sample_spec.channels;
+inf.info.preferredSampleRate = i->sample_spec.rate;
+inf.info.isDefaultInput = (rt_pa_info.default_source_name == i->name);
+inf.source_index = i->index;
+inf.source_name = i->name;
+for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr )
+inf.info.sampleRates.push_back( *sr );
+for ( const rtaudio_pa_format_mapping_t *fm = supported_sampleformats;
+fm->rtaudio_format; ++fm )
+inf.info.nativeFormats |= fm->rtaudio_format;
+for (size_t i=0; i < rt_pa_info.dev.size(); i++)
+{
+/* Attempt to match up sink and source records by device description. */
+if (rt_pa_info.dev[i].info.name == inf.info.name) {
+rt_pa_info.dev[i].source_index = inf.source_index;
+rt_pa_info.dev[i].source_name = inf.source_name;
+rt_pa_info.dev[i].info.inputChannels = inf.info.inputChannels;
+rt_pa_info.dev[i].info.isDefaultInput = inf.info.isDefaultInput;
+/* Assume duplex channels are minimum of input and output channels. */
+/* Uncomment if we add support for DUPLEX
+if (rt_pa_info.dev[i].sink_index > -1) {
+rt_pa_info.dev[i].info.duplexChannels =
+(inf.info.inputChannels < rt_pa_info.dev[i].info.outputChannels)
+? inf.info.inputChannels : rt_pa_info.dev[i].info.outputChannels;
+}
+*/
+return;
+}
+}
+/* try to ensure device #0 is the default */
+if (inf.info.isDefaultInput)
+rt_pa_info.dev.insert(rt_pa_info.dev.begin(), inf);
+else
+rt_pa_info.dev.push_back(inf);
+}
+static void rt_pa_context_state_callback(pa_context *context, void *userdata) {
+(void)userdata;
+switch (pa_context_get_state(context)) {
+case PA_CONTEXT_CONNECTING:
+case PA_CONTEXT_AUTHORIZING:
+case PA_CONTEXT_SETTING_NAME:
+break;
+case PA_CONTEXT_READY:
+rt_pa_info.dev.clear();
+pa_context_get_server_info(context, rt_pa_server_callback, NULL);
+pa_context_get_sink_info_list(context, rt_pa_sink_info_cb, NULL);
+pa_context_get_source_info_list(context, rt_pa_source_info_cb, NULL);
+break;
+case PA_CONTEXT_TERMINATED:
+rt_pa_mainloop_api_quit(0);
+break;
+case PA_CONTEXT_FAILED:
+default:
+rt_pa_mainloop_api_quit(1);
+}
+}
+RtApiPulse::~RtApiPulse()
+{
+if ( stream_.state != STREAM_CLOSED )
+closeStream();
+}
+void RtApiPulse::collectDeviceInfo( void )
+{
+pa_context *context = NULL;
+pa_mainloop *m = NULL;
+char *server = NULL;
+int ret = 1;
+if (!(m = pa_mainloop_new())) {
+errorStream_ << "RtApiPulse::DeviceInfo pa_mainloop_new() failed.";
+errorText_ = errorStream_.str();
+error( RtAudioError::WARNING );
+goto quit;
+}
+rt_pa_mainloop_api = pa_mainloop_get_api(m);
+if (!(context = pa_context_new_with_proplist(rt_pa_mainloop_api, NULL, NULL))) {
+errorStream_ << "pa_context_new() failed.";
+errorText_ = errorStream_.str();
+error( RtAudioError::WARNING );
+goto quit;
+}
+pa_context_set_state_callback(context, rt_pa_context_state_callback, NULL);
+if (pa_context_connect(context, server, PA_CONTEXT_NOFLAGS, NULL) < 0) {
+errorStream_ << "RtApiPulse::DeviceInfo pa_context_connect() failed: "
+<< pa_strerror(pa_context_errno(context));
+errorText_ = errorStream_.str();
+error( RtAudioError::WARNING );
+goto quit;
+}
+if (pa_mainloop_run(m, &ret) < 0) {
+errorStream_ << "pa_mainloop_run() failed.";
+errorText_ = errorStream_.str();
+error( RtAudioError::WARNING );
+goto quit;
+}
+quit:
+if (context)
+pa_context_unref(context);
+if (m) {
+pa_mainloop_free(m);
+}
+pa_xfree(server);
+}
+unsigned int RtApiPulse::getDeviceCount( void )
+{
+collectDeviceInfo();
+return rt_pa_info.dev.size();
+}
+RtAudio::DeviceInfo RtApiPulse::getDeviceInfo( unsigned int device )
+{
+if (rt_pa_info.dev.size()==0)
+collectDeviceInfo();
+if (device < rt_pa_info.dev.size())
+return rt_pa_info.dev[device].info;
+return RtAudio::DeviceInfo();
+}
+static void *pulseaudio_callback( void * user )
+{
+CallbackInfo *cbi = static_cast<CallbackInfo *>( user );
+RtApiPulse *context = static_cast<RtApiPulse *>( cbi->object );
+volatile bool *isRunning = &cbi->isRunning;
+#ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
+if (cbi->doRealtime) {
+std::cerr << "RtAudio pulse: " <<
+(sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") <<
+"running realtime scheduling" << std::endl;
+}
+#endif
+while ( *isRunning ) {
+pthread_testcancel();
+context->callbackEvent();
+}
+pthread_exit( NULL );
+}
+void RtApiPulse::closeStream( void )
+{
+PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
+stream_.callbackInfo.isRunning = false;
+if ( pah ) {
+MUTEX_LOCK( &stream_.mutex );
+if ( stream_.state == STREAM_STOPPED ) {
+pah->runnable = true;
+pthread_cond_signal( &pah->runnable_cv );
+}
+MUTEX_UNLOCK( &stream_.mutex );
+pthread_join( pah->thread, 0 );
+if ( pah->s_play ) {
+pa_simple_flush( pah->s_play, NULL );
+pa_simple_free( pah->s_play );
+}
+if ( pah->s_rec )
+pa_simple_free( pah->s_rec );
+pthread_cond_destroy( &pah->runnable_cv );
+delete pah;
+stream_.apiHandle = 0;
+}
+if ( stream_.userBuffer[0] ) {
+free( stream_.userBuffer[0] );
+stream_.userBuffer[0] = 0;
+}
+if ( stream_.userBuffer[1] ) {
+free( stream_.userBuffer[1] );
+stream_.userBuffer[1] = 0;
+}
+stream_.state = STREAM_CLOSED;
+stream_.mode = UNINITIALIZED;
+}
+void RtApiPulse::callbackEvent( void )
+{
+PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
+if ( stream_.state == STREAM_STOPPED ) {
+MUTEX_LOCK( &stream_.mutex );
+while ( !pah->runnable )
+pthread_cond_wait( &pah->runnable_cv, &stream_.mutex );
+if ( stream_.state != STREAM_RUNNING ) {
+MUTEX_UNLOCK( &stream_.mutex );
+return;
+}
+MUTEX_UNLOCK( &stream_.mutex );
+}
+if ( stream_.state == STREAM_CLOSED ) {
+errorText_ = "RtApiPulse::callbackEvent(): the stream is closed ... "
+"this shouldn't happen!";
+error( RtAudioError::WARNING );
+return;
+}
+RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
+double streamTime = getStreamTime();
+RtAudioStreamStatus status = 0;
+int doStopStream = callback( stream_.userBuffer[OUTPUT], stream_.userBuffer[INPUT],
+stream_.bufferSize, streamTime, status,
+stream_.callbackInfo.userData );
+if ( doStopStream == 2 ) {
+abortStream();
+return;
+}
+MUTEX_LOCK( &stream_.mutex );
+void *pulse_in = stream_.doConvertBuffer[INPUT] ? stream_.deviceBuffer : stream_.userBuffer[INPUT];
+void *pulse_out = stream_.doConvertBuffer[OUTPUT] ? stream_.deviceBuffer : stream_.userBuffer[OUTPUT];
+if ( stream_.state != STREAM_RUNNING )
+goto unlock;
+int pa_error;
+size_t bytes;
+if (stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+if ( stream_.doConvertBuffer[OUTPUT] ) {
+convertBuffer( stream_.deviceBuffer,
+stream_.userBuffer[OUTPUT],
+stream_.convertInfo[OUTPUT] );
+bytes = stream_.nDeviceChannels[OUTPUT] * stream_.bufferSize *
+formatBytes( stream_.deviceFormat[OUTPUT] );
+} else
+bytes = stream_.nUserChannels[OUTPUT] * stream_.bufferSize *
+formatBytes( stream_.userFormat );
+if ( pa_simple_write( pah->s_play, pulse_out, bytes, &pa_error ) < 0 ) {
+errorStream_ << "RtApiPulse::callbackEvent: audio write error, " <<
+pa_strerror( pa_error ) << ".";
+errorText_ = errorStream_.str();
+error( RtAudioError::WARNING );
+}
+}
+if ( stream_.mode == INPUT || stream_.mode == DUPLEX) {
+if ( stream_.doConvertBuffer[INPUT] )
+bytes = stream_.nDeviceChannels[INPUT] * stream_.bufferSize *
+formatBytes( stream_.deviceFormat[INPUT] );
+else
+bytes = stream_.nUserChannels[INPUT] * stream_.bufferSize *
+formatBytes( stream_.userFormat );
+if ( pa_simple_read( pah->s_rec, pulse_in, bytes, &pa_error ) < 0 ) {
+errorStream_ << "RtApiPulse::callbackEvent: audio read error, " <<
+pa_strerror( pa_error ) << ".";
+errorText_ = errorStream_.str();
+error( RtAudioError::WARNING );
+}
+if ( stream_.doConvertBuffer[INPUT] ) {
+convertBuffer( stream_.userBuffer[INPUT],
+stream_.deviceBuffer,
+stream_.convertInfo[INPUT] );
+}
+}
+unlock:
+MUTEX_UNLOCK( &stream_.mutex );
+RtApi::tickStreamTime();
+if ( doStopStream == 1 )
+stopStream();
+}
+void RtApiPulse::startStream( void )
+{
+PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
+if ( stream_.state == STREAM_CLOSED ) {
+errorText_ = "RtApiPulse::startStream(): the stream is not open!";
+error( RtAudioError::INVALID_USE );
+return;
+}
+if ( stream_.state == STREAM_RUNNING ) {
+errorText_ = "RtApiPulse::startStream(): the stream is already running!";
+error( RtAudioError::WARNING );
+return;
+}
+MUTEX_LOCK( &stream_.mutex );
+#if defined( HAVE_GETTIMEOFDAY )
+gettimeofday( &stream_.lastTickTimestamp, NULL );
+#endif
+stream_.state = STREAM_RUNNING;
+pah->runnable = true;
+pthread_cond_signal( &pah->runnable_cv );
+MUTEX_UNLOCK( &stream_.mutex );
+}
+void RtApiPulse::stopStream( void )
+{
+PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
+if ( stream_.state == STREAM_CLOSED ) {
+errorText_ = "RtApiPulse::stopStream(): the stream is not open!";
+error( RtAudioError::INVALID_USE );
+return;
+}
+if ( stream_.state == STREAM_STOPPED ) {
+errorText_ = "RtApiPulse::stopStream(): the stream is already stopped!";
+error( RtAudioError::WARNING );
+return;
+}
+stream_.state = STREAM_STOPPED;
+MUTEX_LOCK( &stream_.mutex );
+if ( pah ) {
+pah->runnable = false;
+if ( pah->s_play ) {
+int pa_error;
+if ( pa_simple_drain( pah->s_play, &pa_error ) < 0 ) {
+errorStream_ << "RtApiPulse::stopStream: error draining output device, " <<
+pa_strerror( pa_error ) << ".";
+errorText_ = errorStream_.str();
+MUTEX_UNLOCK( &stream_.mutex );
+error( RtAudioError::SYSTEM_ERROR );
+return;
+}
+}
+}
+stream_.state = STREAM_STOPPED;
+MUTEX_UNLOCK( &stream_.mutex );
+}
+void RtApiPulse::abortStream( void )
+{
+PulseAudioHandle *pah = static_cast<PulseAudioHandle*>( stream_.apiHandle );
+if ( stream_.state == STREAM_CLOSED ) {
+errorText_ = "RtApiPulse::abortStream(): the stream is not open!";
+error( RtAudioError::INVALID_USE );
+return;
+}
+if ( stream_.state == STREAM_STOPPED ) {
+errorText_ = "RtApiPulse::abortStream(): the stream is already stopped!";
+error( RtAudioError::WARNING );
+return;
+}
+stream_.state = STREAM_STOPPED;
+MUTEX_LOCK( &stream_.mutex );
+if ( pah ) {
+pah->runnable = false;
+if ( pah->s_play ) {
+int pa_error;
+if ( pa_simple_flush( pah->s_play, &pa_error ) < 0 ) {
+errorStream_ << "RtApiPulse::abortStream: error flushing output device, " <<
+pa_strerror( pa_error ) << ".";
+errorText_ = errorStream_.str();
+MUTEX_UNLOCK( &stream_.mutex );
+error( RtAudioError::SYSTEM_ERROR );
+return;
+}
+}
+}
+stream_.state = STREAM_STOPPED;
+MUTEX_UNLOCK( &stream_.mutex );
+}
+bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode,
+unsigned int channels, unsigned int firstChannel,
+unsigned int sampleRate, RtAudioFormat format,
+unsigned int *bufferSize, RtAudio::StreamOptions *options )
+{
+PulseAudioHandle *pah = 0;
+unsigned long bufferBytes = 0;
+pa_sample_spec ss;
+if ( device >= rt_pa_info.dev.size() ) return false;
+if ( firstChannel != 0 ) {
+errorText_ = "PulseAudio does not support channel offset mapping.";
+return false;
+}
+/* these may be NULL for default, but we've already got the names */
+const char *dev_input = NULL;
+const char *dev_output = NULL;
+if (!rt_pa_info.dev[device].source_name.empty())
+dev_input = rt_pa_info.dev[device].source_name.c_str();
+if (!rt_pa_info.dev[device].sink_name.empty())
+dev_output = rt_pa_info.dev[device].sink_name.c_str();
+if (mode==INPUT && rt_pa_info.dev[device].info.inputChannels == 0) {
+errorText_ = "PulseAudio device does not support input.";
+return false;
+}
+if (mode==OUTPUT && rt_pa_info.dev[device].info.outputChannels == 0) {
+errorText_ = "PulseAudio device does not support output.";
+return false;
+}
+if (mode==DUPLEX && rt_pa_info.dev[device].info.duplexChannels == 0) {
+/* Note: will always error, DUPLEX not yet supported */
+errorText_ = "PulseAudio device does not support duplex.";
+return false;
+}
+if (mode==INPUT && rt_pa_info.dev[device].info.inputChannels < channels) {
+errorText_ = "PulseAudio: unsupported number of input channels.";
+return false;
+}
+if (mode==OUTPUT && rt_pa_info.dev[device].info.outputChannels < channels) {
+errorText_ = "PulseAudio: unsupported number of output channels.";
+return false;
+}
+if (mode==DUPLEX && rt_pa_info.dev[device].info.duplexChannels < channels) {
+/* Note: will always error, DUPLEX not yet supported */
+errorText_ = "PulseAudio: unsupported number of duplex channels.";
+return false;
+}
+ss.channels = channels;
+bool sr_found = false;
+for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr ) {
+if ( sampleRate == *sr ) {
+sr_found = true;
+stream_.sampleRate = sampleRate;
+ss.rate = sampleRate;
+break;
+}
+}
+if ( !sr_found ) {
+errorText_ = "RtApiPulse::probeDeviceOpen: unsupported sample rate.";
+return false;
+}
+bool sf_found = 0;
+for ( const rtaudio_pa_format_mapping_t *sf = supported_sampleformats;
+sf->rtaudio_format && sf->pa_format != PA_SAMPLE_INVALID; ++sf ) {
+if ( format == sf->rtaudio_format ) {
+sf_found = true;
+stream_.userFormat = sf->rtaudio_format;
+stream_.deviceFormat[mode] = stream_.userFormat;
+ss.format = sf->pa_format;
+break;
+}
+}
+if ( !sf_found ) { // Use internal data format conversion.
+stream_.userFormat = format;
+stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
+ss.format = PA_SAMPLE_FLOAT32LE;
+}
+// Set other stream parameters.
+if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
+else stream_.userInterleaved = true;
+stream_.deviceInterleaved[mode] = true;
+stream_.nBuffers = options ? options->numberOfBuffers : 1;
+stream_.doByteSwap[mode] = false;
+stream_.nUserChannels[mode] = channels;
+stream_.nDeviceChannels[mode] = channels + firstChannel;
+stream_.channelOffset[mode] = 0;
+std::string streamName = "RtAudio";
+// Set flags for buffer conversion.
+stream_.doConvertBuffer[mode] = false;
+if ( stream_.userFormat != stream_.deviceFormat[mode] )
+stream_.doConvertBuffer[mode] = true;
+if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
+stream_.doConvertBuffer[mode] = true;
+if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] )
+stream_.doConvertBuffer[mode] = true;
+// Allocate necessary internal buffers.
+bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+if ( stream_.userBuffer[mode] == NULL ) {
+errorText_ = "RtApiPulse::probeDeviceOpen: error allocating user buffer memory.";
+goto error;
+}
+stream_.bufferSize = *bufferSize;
+if ( stream_.doConvertBuffer[mode] ) {
+bool makeBuffer = true;
+bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
+if ( mode == INPUT ) {
+if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
+unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
+if ( bufferBytes <= bytesOut ) makeBuffer = false;
+}
+}
+if ( makeBuffer ) {
+bufferBytes *= *bufferSize;
+if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
+stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
+if ( stream_.deviceBuffer == NULL ) {
+errorText_ = "RtApiPulse::probeDeviceOpen: error allocating device buffer memory.";
+goto error;
+}
+}
+}
+stream_.device[mode] = device;
+// Setup the buffer conversion information structure.
+if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
+if ( !stream_.apiHandle ) {
+PulseAudioHandle *pah = new PulseAudioHandle;
+if ( !pah ) {
+errorText_ = "RtApiPulse::probeDeviceOpen: error allocating memory for handle.";
+goto error;
+}
+stream_.apiHandle = pah;
+if ( pthread_cond_init( &pah->runnable_cv, NULL ) != 0 ) {
+errorText_ = "RtApiPulse::probeDeviceOpen: error creating condition variable.";
+goto error;
+}
+}
+pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
+int error;
+if ( options && !options->streamName.empty() ) streamName = options->streamName;
+switch ( mode ) {
+pa_buffer_attr buffer_attr;
+case INPUT:
+buffer_attr.fragsize = bufferBytes;
+buffer_attr.maxlength = -1;
+pah->s_rec = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_RECORD,
+dev_input, "Record", &ss, NULL, &buffer_attr, &error );
+if ( !pah->s_rec ) {
+errorText_ = "RtApiPulse::probeDeviceOpen: error connecting input to PulseAudio server.";
+goto error;
+}
+break;
+case OUTPUT: {
+pa_buffer_attr * attr_ptr;
+if ( options && options->numberOfBuffers > 0 ) {
+// pa_buffer_attr::fragsize is recording-only.
+// Hopefully PortAudio won't access uninitialized fields.
+buffer_attr.maxlength = bufferBytes * options->numberOfBuffers;
+buffer_attr.minreq = -1;
+buffer_attr.prebuf = -1;
+buffer_attr.tlength = -1;
+attr_ptr = &buffer_attr;
+} else {
+attr_ptr = nullptr;
+}
+pah->s_play = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_PLAYBACK,
+dev_output, "Playback", &ss, NULL, attr_ptr, &error );
+if ( !pah->s_play ) {
+errorText_ = "RtApiPulse::probeDeviceOpen: error connecting output to PulseAudio server.";
+goto error;
+}
+break;
+}
+case DUPLEX:
+/* Note: We could add DUPLEX by synchronizing multiple streams,
+but it would mean moving from Simple API to Asynchronous API:
+https://freedesktop.org/software/pulseaudio/doxygen/streams.html#sync_streams */
+errorText_ = "RtApiPulse::probeDeviceOpen: duplex not supported for PulseAudio.";
+goto error;
+default:
+goto error;
+}
+if ( stream_.mode == UNINITIALIZED )
+stream_.mode = mode;
+else if ( stream_.mode == mode )
+goto error;
+else
+stream_.mode = DUPLEX;
+if ( !stream_.callbackInfo.isRunning ) {
+stream_.callbackInfo.object = this;
+stream_.state = STREAM_STOPPED;
+// Set the thread attributes for joinable and realtime scheduling
+// priority (optional).  The higher priority will only take affect
+// if the program is run as root or suid. Note, under Linux
+// processes with CAP_SYS_NICE privilege, a user can change
+// scheduling policy and priority (thus need not be root). See
+// POSIX "capabilities".
+pthread_attr_t attr;
+pthread_attr_init( &attr );
+pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
+#ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
+if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
+stream_.callbackInfo.doRealtime = true;
+struct sched_param param;
+int priority = options->priority;
+int min = sched_get_priority_min( SCHED_RR );
+int max = sched_get_priority_max( SCHED_RR );
+if ( priority < min ) priority = min;
+else if ( priority > max ) priority = max;
+param.sched_priority = priority;
+// Set the policy BEFORE the priority. Otherwise it fails.
+pthread_attr_setschedpolicy(&attr, SCHED_RR);
+pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM);
+// This is definitely required. Otherwise it fails.
+pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED);
+pthread_attr_setschedparam(&attr, &param);
+}
+else
+pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
+#else
+pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
+#endif
+stream_.callbackInfo.isRunning = true;
+int result = pthread_create( &pah->thread, &attr, pulseaudio_callback, (void *)&stream_.callbackInfo);
+pthread_attr_destroy(&attr);
+if(result != 0) {
+// Failed. Try instead with default attributes.
+result = pthread_create( &pah->thread, NULL, pulseaudio_callback, (void *)&stream_.callbackInfo);
+if(result != 0) {
+stream_.callbackInfo.isRunning = false;
+errorText_ = "RtApiPulse::probeDeviceOpen: error creating thread.";
+goto error;
+}
+}
+}
+return SUCCESS;
+error:
+if ( pah && stream_.callbackInfo.isRunning ) {
+pthread_cond_destroy( &pah->runnable_cv );
+delete pah;
+stream_.apiHandle = 0;
+}
+for ( int i=0; i<2; i++ ) {
+if ( stream_.userBuffer[i] ) {
+free( stream_.userBuffer[i] );
+stream_.userBuffer[i] = 0;
+}
+}
+if ( stream_.deviceBuffer ) {
+free( stream_.deviceBuffer );
+stream_.deviceBuffer = 0;
+}
+stream_.state = STREAM_CLOSED;
+return FAILURE;
+}
+//******************** End of __LINUX_PULSE__ *********************//
+#endif
+#if defined(__LINUX_OSS__)
+#include <unistd.h>
+#include <sys/ioctl.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include <sys/soundcard.h>
+#include <errno.h>
+#include <math.h>
+static void *ossCallbackHandler(void * ptr);
+// A structure to hold various information related to the OSS API
+// implementation.
+struct OssHandle {
+int id[2];    // device ids
+bool xrun[2];
+bool triggered;
+pthread_cond_t runnable;
+OssHandle()
+:triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
+};
+RtApiOss :: RtApiOss()
+{
+// Nothing to do here.
+}
+RtApiOss :: ~RtApiOss()
+{
+if ( stream_.state != STREAM_CLOSED ) closeStream();
+}
+unsigned int RtApiOss :: getDeviceCount( void )
+{
+int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
+if ( mixerfd == -1 ) {
+errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'.";
+error( RtAudioError::WARNING );
+return 0;
+}
+oss_sysinfo sysinfo;
+if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) {
+close( mixerfd );
+errorText_ = "RtApiOss::getDeviceCount: error getting sysinfo, OSS version >= 4.0 is required.";
+error( RtAudioError::WARNING );
+return 0;
+}
+close( mixerfd );
+return sysinfo.numaudios;
+}
+RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device )
+{
+RtAudio::DeviceInfo info;
+info.probed = false;
+int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
+if ( mixerfd == -1 ) {
+errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'.";
+error( RtAudioError::WARNING );
+return info;
+}
+oss_sysinfo sysinfo;
+int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
+if ( result == -1 ) {
+close( mixerfd );
+errorText_ = "RtApiOss::getDeviceInfo: error getting sysinfo, OSS version >= 4.0 is required.";
+error( RtAudioError::WARNING );
+return info;
+}
+unsigned nDevices = sysinfo.numaudios;
+if ( nDevices == 0 ) {
+close( mixerfd );
+errorText_ = "RtApiOss::getDeviceInfo: no devices found!";
+error( RtAudioError::INVALID_USE );
+return info;
+}
+if ( device >= nDevices ) {
+close( mixerfd );
+errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!";
+error( RtAudioError::INVALID_USE );
+return info;
+}
+oss_audioinfo ainfo;
+ainfo.dev = device;
+result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
+close( mixerfd );
+if ( result == -1 ) {
+errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
+errorText_ = errorStream_.str();
+error( RtAudioError::WARNING );
+return info;
+}
+// Probe channels
+if ( ainfo.caps & PCM_CAP_OUTPUT ) info.outputChannels = ainfo.max_channels;
+if ( ainfo.caps & PCM_CAP_INPUT ) info.inputChannels = ainfo.max_channels;
+if ( ainfo.caps & PCM_CAP_DUPLEX ) {
+if ( info.outputChannels > 0 && info.inputChannels > 0 && ainfo.caps & PCM_CAP_DUPLEX )
+info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
+}
+// Probe data formats ... do for input
+unsigned long mask = ainfo.iformats;
+if ( mask & AFMT_S16_LE || mask & AFMT_S16_BE )
+info.nativeFormats |= RTAUDIO_SINT16;
+if ( mask & AFMT_S8 )
+info.nativeFormats |= RTAUDIO_SINT8;
+if ( mask & AFMT_S32_LE || mask & AFMT_S32_BE )
+info.nativeFormats |= RTAUDIO_SINT32;
+#ifdef AFMT_FLOAT
+if ( mask & AFMT_FLOAT )
+info.nativeFormats |= RTAUDIO_FLOAT32;
+#endif
+if ( mask & AFMT_S24_LE || mask & AFMT_S24_BE )
+info.nativeFormats |= RTAUDIO_SINT24;
+// Check that we have at least one supported format
+if ( info.nativeFormats == 0 ) {
+errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio.";
+errorText_ = errorStream_.str();
+error( RtAudioError::WARNING );
+return info;
+}
+// Probe the supported sample rates.
+info.sampleRates.clear();
+if ( ainfo.nrates ) {
+for ( unsigned int i=0; i<ainfo.nrates; i++ ) {
+for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
+if ( ainfo.rates[i] == SAMPLE_RATES[k] ) {
+info.sampleRates.push_back( SAMPLE_RATES[k] );
+if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
+info.preferredSampleRate = SAMPLE_RATES[k];
+break;
+}
+}
+}
+}
+else {
+// Check min and max rate values;
+for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
+if ( ainfo.min_rate <= (int) SAMPLE_RATES[k] && ainfo.max_rate >= (int) SAMPLE_RATES[k] ) {
+info.sampleRates.push_back( SAMPLE_RATES[k] );
+if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
+info.preferredSampleRate = SAMPLE_RATES[k];
+}
+}
+}
+if ( info.sampleRates.size() == 0 ) {
+errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ").";
+errorText_ = errorStream_.str();
+error( RtAudioError::WARNING );
+}
+else {
+info.probed = true;
+info.name = ainfo.name;
+}
+return info;
+}
+bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+unsigned int firstChannel, unsigned int sampleRate,
+RtAudioFormat format, unsigned int *bufferSize,
+RtAudio::StreamOptions *options )
+{
+int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
+if ( mixerfd == -1 ) {
+errorText_ = "RtApiOss::probeDeviceOpen: error opening '/dev/mixer'.";
+return FAILURE;
+}
+oss_sysinfo sysinfo;
+int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
+if ( result == -1 ) {
+close( mixerfd );
+errorText_ = "RtApiOss::probeDeviceOpen: error getting sysinfo, OSS version >= 4.0 is required.";
+return FAILURE;
+}
+unsigned nDevices = sysinfo.numaudios;
+if ( nDevices == 0 ) {
+// This should not happen because a check is made before this function is called.
+close( mixerfd );
+errorText_ = "RtApiOss::probeDeviceOpen: no devices found!";
+return FAILURE;
+}
+if ( device >= nDevices ) {
+// This should not happen because a check is made before this function is called.
+close( mixerfd );
+errorText_ = "RtApiOss::probeDeviceOpen: device ID is invalid!";
+return FAILURE;
+}
+oss_audioinfo ainfo;
+ainfo.dev = device;
+result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
+close( mixerfd );
+if ( result == -1 ) {
+errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+// Check if device supports input or output
+if ( ( mode == OUTPUT && !( ainfo.caps & PCM_CAP_OUTPUT ) ) ||
+( mode == INPUT && !( ainfo.caps & PCM_CAP_INPUT ) ) ) {
+if ( mode == OUTPUT )
+errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support output.";
+else
+errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support input.";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+int flags = 0;
+OssHandle *handle = (OssHandle *) stream_.apiHandle;
+if ( mode == OUTPUT )
+flags |= O_WRONLY;
+else { // mode == INPUT
+if (stream_.mode == OUTPUT && stream_.device[0] == device) {
+// We just set the same device for playback ... close and reopen for duplex (OSS only).
+close( handle->id[0] );
+handle->id[0] = 0;
+if ( !( ainfo.caps & PCM_CAP_DUPLEX ) ) {
+errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support duplex mode.";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+// Check that the number previously set channels is the same.
+if ( stream_.nUserChannels[0] != channels ) {
+errorStream_ << "RtApiOss::probeDeviceOpen: input/output channels must be equal for OSS duplex device (" << ainfo.name << ").";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+flags |= O_RDWR;
+}
+else
+flags |= O_RDONLY;
+}
+// Set exclusive access if specified.
+if ( options && options->flags & RTAUDIO_HOG_DEVICE ) flags |= O_EXCL;
+// Try to open the device.
+int fd;
+fd = open( ainfo.devnode, flags, 0 );
+if ( fd == -1 ) {
+if ( errno == EBUSY )
+errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") is busy.";
+else
+errorStream_ << "RtApiOss::probeDeviceOpen: error opening device (" << ainfo.name << ").";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+// For duplex operation, specifically set this mode (this doesn't seem to work).
+/*
+if ( flags | O_RDWR ) {
+result = ioctl( fd, SNDCTL_DSP_SETDUPLEX, NULL );
+if ( result == -1) {
+errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ").";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+}
+*/
+// Check the device channel support.
+stream_.nUserChannels[mode] = channels;
+if ( ainfo.max_channels < (int)(channels + firstChannel) ) {
+close( fd );
+errorStream_ << "RtApiOss::probeDeviceOpen: the device (" << ainfo.name << ") does not support requested channel parameters.";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+// Set the number of channels.
+int deviceChannels = channels + firstChannel;
+result = ioctl( fd, SNDCTL_DSP_CHANNELS, &deviceChannels );
+if ( result == -1 || deviceChannels < (int)(channels + firstChannel) ) {
+close( fd );
+errorStream_ << "RtApiOss::probeDeviceOpen: error setting channel parameters on device (" << ainfo.name << ").";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+stream_.nDeviceChannels[mode] = deviceChannels;
+// Get the data format mask
+int mask;
+result = ioctl( fd, SNDCTL_DSP_GETFMTS, &mask );
+if ( result == -1 ) {
+close( fd );
+errorStream_ << "RtApiOss::probeDeviceOpen: error getting device (" << ainfo.name << ") data formats.";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+// Determine how to set the device format.
+stream_.userFormat = format;
+int deviceFormat = -1;
+stream_.doByteSwap[mode] = false;
+if ( format == RTAUDIO_SINT8 ) {
+if ( mask & AFMT_S8 ) {
+deviceFormat = AFMT_S8;
+stream_.deviceFormat[mode] = RTAUDIO_SINT8;
+}
+}
+else if ( format == RTAUDIO_SINT16 ) {
+if ( mask & AFMT_S16_NE ) {
+deviceFormat = AFMT_S16_NE;
+stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+}
+else if ( mask & AFMT_S16_OE ) {
+deviceFormat = AFMT_S16_OE;
+stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+stream_.doByteSwap[mode] = true;
+}
+}
+else if ( format == RTAUDIO_SINT24 ) {
+if ( mask & AFMT_S24_NE ) {
+deviceFormat = AFMT_S24_NE;
+stream_.deviceFormat[mode] = RTAUDIO_SINT24;
+}
+else if ( mask & AFMT_S24_OE ) {
+deviceFormat = AFMT_S24_OE;
+stream_.deviceFormat[mode] = RTAUDIO_SINT24;
+stream_.doByteSwap[mode] = true;
+}
+}
+else if ( format == RTAUDIO_SINT32 ) {
+if ( mask & AFMT_S32_NE ) {
+deviceFormat = AFMT_S32_NE;
+stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+}
+else if ( mask & AFMT_S32_OE ) {
+deviceFormat = AFMT_S32_OE;
+stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+stream_.doByteSwap[mode] = true;
+}
+}
+if ( deviceFormat == -1 ) {
+// The user requested format is not natively supported by the device.
+if ( mask & AFMT_S16_NE ) {
+deviceFormat = AFMT_S16_NE;
+stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+}
+else if ( mask & AFMT_S32_NE ) {
+deviceFormat = AFMT_S32_NE;
+stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+}
+else if ( mask & AFMT_S24_NE ) {
+deviceFormat = AFMT_S24_NE;
+stream_.deviceFormat[mode] = RTAUDIO_SINT24;
+}
+else if ( mask & AFMT_S16_OE ) {
+deviceFormat = AFMT_S16_OE;
+stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+stream_.doByteSwap[mode] = true;
+}
+else if ( mask & AFMT_S32_OE ) {
+deviceFormat = AFMT_S32_OE;
+stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+stream_.doByteSwap[mode] = true;
+}
+else if ( mask & AFMT_S24_OE ) {
+deviceFormat = AFMT_S24_OE;
+stream_.deviceFormat[mode] = RTAUDIO_SINT24;
+stream_.doByteSwap[mode] = true;
+}
+else if ( mask & AFMT_S8) {
+deviceFormat = AFMT_S8;
+stream_.deviceFormat[mode] = RTAUDIO_SINT8;
+}
+}
+if ( stream_.deviceFormat[mode] == 0 ) {
+// This really shouldn't happen ...
+close( fd );
+errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") data format not supported by RtAudio.";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+// Set the data format.
+int temp = deviceFormat;
+result = ioctl( fd, SNDCTL_DSP_SETFMT, &deviceFormat );
+if ( result == -1 || deviceFormat != temp ) {
+close( fd );
+errorStream_ << "RtApiOss::probeDeviceOpen: error setting data format on device (" << ainfo.name << ").";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+// Attempt to set the buffer size.  According to OSS, the minimum
+// number of buffers is two.  The supposed minimum buffer size is 16
+// bytes, so that will be our lower bound.  The argument to this
+// call is in the form 0xMMMMSSSS (hex), where the buffer size (in
+// bytes) is given as 2^SSSS and the number of buffers as 2^MMMM.
+// We'll check the actual value used near the end of the setup
+// procedure.
+int ossBufferBytes = *bufferSize * formatBytes( stream_.deviceFormat[mode] ) * deviceChannels;
+if ( ossBufferBytes < 16 ) ossBufferBytes = 16;
+int buffers = 0;
+if ( options ) buffers = options->numberOfBuffers;
+if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) buffers = 2;
+if ( buffers < 2 ) buffers = 3;
+temp = ((int) buffers << 16) + (int)( log10( (double)ossBufferBytes ) / log10( 2.0 ) );
+result = ioctl( fd, SNDCTL_DSP_SETFRAGMENT, &temp );
+if ( result == -1 ) {
+close( fd );
+errorStream_ << "RtApiOss::probeDeviceOpen: error setting buffer size on device (" << ainfo.name << ").";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+stream_.nBuffers = buffers;
+// Save buffer size (in sample frames).
+*bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels );
+stream_.bufferSize = *bufferSize;
+// Set the sample rate.
+int srate = sampleRate;
+result = ioctl( fd, SNDCTL_DSP_SPEED, &srate );
+if ( result == -1 ) {
+close( fd );
+errorStream_ << "RtApiOss::probeDeviceOpen: error setting sample rate (" << sampleRate << ") on device (" << ainfo.name << ").";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+// Verify the sample rate setup worked.
+if ( abs( srate - (int)sampleRate ) > 100 ) {
+close( fd );
+errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ").";
+errorText_ = errorStream_.str();
+return FAILURE;
+}
+stream_.sampleRate = sampleRate;
+if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device) {
+// We're doing duplex setup here.
+stream_.deviceFormat[0] = stream_.deviceFormat[1];
+stream_.nDeviceChannels[0] = deviceChannels;
+}
+// Set interleaving parameters.
+stream_.userInterleaved = true;
+stream_.deviceInterleaved[mode] =  true;
+if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
+stream_.userInterleaved = false;
+// Set flags for buffer conversion
+stream_.doConvertBuffer[mode] = false;
+if ( stream_.userFormat != stream_.deviceFormat[mode] )
+stream_.doConvertBuffer[mode] = true;
+if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
+stream_.doConvertBuffer[mode] = true;
+if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+stream_.nUserChannels[mode] > 1 )
+stream_.doConvertBuffer[mode] = true;
+// Allocate the stream handles if necessary and then save.
+if ( stream_.apiHandle == 0 ) {
+try {
+handle = new OssHandle;
+}
+catch ( std::bad_alloc& ) {
+errorText_ = "RtApiOss::probeDeviceOpen: error allocating OssHandle memory.";
+goto error;
+}
+if ( pthread_cond_init( &handle->runnable, NULL ) ) {
+errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable.";
+goto error;
+}
+stream_.apiHandle = (void *) handle;
+}
+else {
+handle = (OssHandle *) stream_.apiHandle;
+}
+handle->id[mode] = fd;
+// Allocate necessary internal buffers.
+unsigned long bufferBytes;
+bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+if ( stream_.userBuffer[mode] == NULL ) {
+errorText_ = "RtApiOss::probeDeviceOpen: error allocating user buffer memory.";
+goto error;
+}
+if ( stream_.doConvertBuffer[mode] ) {
+bool makeBuffer = true;
+bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
+if ( mode == INPUT ) {
+if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
+unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
+if ( bufferBytes <= bytesOut ) makeBuffer = false;
+}
+}
+if ( makeBuffer ) {
+bufferBytes *= *bufferSize;
+if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
+stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
+if ( stream_.deviceBuffer == NULL ) {
+errorText_ = "RtApiOss::probeDeviceOpen: error allocating device buffer memory.";
+goto error;
+}
+}
+}
+stream_.device[mode] = device;
+stream_.state = STREAM_STOPPED;
+// Setup the buffer conversion information structure.
+if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
+// Setup thread if necessary.
+if ( stream_.mode == OUTPUT && mode == INPUT ) {
+// We had already set up an output stream.
+stream_.mode = DUPLEX;
+if ( stream_.device[0] == device ) handle->id[0] = fd;
+}
+else {
+stream_.mode = mode;
+// Setup callback thread.
+stream_.callbackInfo.object = (void *) this;
+// Set the thread attributes for joinable and realtime scheduling
+// priority.  The higher priority will only take affect if the
+// program is run as root or suid.
+pthread_attr_t attr;
+pthread_attr_init( &attr );
+pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
+#ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
+if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
+stream_.callbackInfo.doRealtime = true;
+struct sched_param param;
+int priority = options->priority;
+int min = sched_get_priority_min( SCHED_RR );
+int max = sched_get_priority_max( SCHED_RR );
+if ( priority < min ) priority = min;
+else if ( priority > max ) priority = max;
+param.sched_priority = priority;
+// Set the policy BEFORE the priority. Otherwise it fails.
+pthread_attr_setschedpolicy(&attr, SCHED_RR);
+pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM);
+// This is definitely required. Otherwise it fails.
+pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED);
+pthread_attr_setschedparam(&attr, &param);
+}
+else
+pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
+#else
+pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
+#endif
+stream_.callbackInfo.isRunning = true;
+result = pthread_create( &stream_.callbackInfo.thread, &attr, ossCallbackHandler, &stream_.callbackInfo );
+pthread_attr_destroy( &attr );
+if ( result ) {
+// Failed. Try instead with default attributes.
+result = pthread_create( &stream_.callbackInfo.thread, NULL, ossCallbackHandler, &stream_.callbackInfo );
+if ( result ) {
+stream_.callbackInfo.isRunning = false;
+errorText_ = "RtApiOss::error creating callback thread!";
+goto error;
+}
+}
+}
+return SUCCESS;
+error:
+if ( handle ) {
+pthread_cond_destroy( &handle->runnable );
+if ( handle->id[0] ) close( handle->id[0] );
+if ( handle->id[1] ) close( handle->id[1] );
+delete handle;
+stream_.apiHandle = 0;
+}
+for ( int i=0; i<2; i++ ) {
+if ( stream_.userBuffer[i] ) {
+free( stream_.userBuffer[i] );
+stream_.userBuffer[i] = 0;
+}
+}
+if ( stream_.deviceBuffer ) {
+free( stream_.deviceBuffer );
+stream_.deviceBuffer = 0;
+}
+stream_.state = STREAM_CLOSED;
+return FAILURE;
+}
+void RtApiOss :: closeStream()
+{
+if ( stream_.state == STREAM_CLOSED ) {
+errorText_ = "RtApiOss::closeStream(): no open stream to close!";
+error( RtAudioError::WARNING );
+return;
+}
+OssHandle *handle = (OssHandle *) stream_.apiHandle;
+stream_.callbackInfo.isRunning = false;
+MUTEX_LOCK( &stream_.mutex );
+if ( stream_.state == STREAM_STOPPED )
+pthread_cond_signal( &handle->runnable );
+MUTEX_UNLOCK( &stream_.mutex );
+pthread_join( stream_.callbackInfo.thread, NULL );
+if ( stream_.state == STREAM_RUNNING ) {
+if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
+ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
+else
+ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
+stream_.state = STREAM_STOPPED;
+}
+if ( handle ) {
+pthread_cond_destroy( &handle->runnable );
+if ( handle->id[0] ) close( handle->id[0] );
+if ( handle->id[1] ) close( handle->id[1] );
+delete handle;
+stream_.apiHandle = 0;
+}
+for ( int i=0; i<2; i++ ) {
+if ( stream_.userBuffer[i] ) {
+free( stream_.userBuffer[i] );
+stream_.userBuffer[i] = 0;
+}
+}
+if ( stream_.deviceBuffer ) {
+free( stream_.deviceBuffer );
+stream_.deviceBuffer = 0;
+}
+stream_.mode = UNINITIALIZED;
+stream_.state = STREAM_CLOSED;
+}
+void RtApiOss :: startStream()
+{
+verifyStream();
+if ( stream_.state == STREAM_RUNNING ) {
+errorText_ = "RtApiOss::startStream(): the stream is already running!";
+error( RtAudioError::WARNING );
+return;
+}
+MUTEX_LOCK( &stream_.mutex );
+#if defined( HAVE_GETTIMEOFDAY )
+gettimeofday( &stream_.lastTickTimestamp, NULL );
+#endif
+stream_.state = STREAM_RUNNING;
+// No need to do anything else here ... OSS automatically starts
+// when fed samples.
+MUTEX_UNLOCK( &stream_.mutex );
+OssHandle *handle = (OssHandle *) stream_.apiHandle;
+pthread_cond_signal( &handle->runnable );
+}
+void RtApiOss :: stopStream()
+{
+verifyStream();
+if ( stream_.state == STREAM_STOPPED ) {
+errorText_ = "RtApiOss::stopStream(): the stream is already stopped!";
+error( RtAudioError::WARNING );
+return;
+}
+MUTEX_LOCK( &stream_.mutex );
+// The state might change while waiting on a mutex.
+if ( stream_.state == STREAM_STOPPED ) {
+MUTEX_UNLOCK( &stream_.mutex );
+return;
+}
+int result = 0;
+OssHandle *handle = (OssHandle *) stream_.apiHandle;
+if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+// Flush the output with zeros a few times.
+char *buffer;
+int samples;
+RtAudioFormat format;
+if ( stream_.doConvertBuffer[0] ) {
+buffer = stream_.deviceBuffer;
+samples = stream_.bufferSize * stream_.nDeviceChannels[0];
+format = stream_.deviceFormat[0];
+}
+else {
+buffer = stream_.userBuffer[0];
+samples = stream_.bufferSize * stream_.nUserChannels[0];
+format = stream_.userFormat;
+}
+memset( buffer, 0, samples * formatBytes(format) );
+for ( unsigned int i=0; i<stream_.nBuffers+1; i++ ) {
+result = write( handle->id[0], buffer, samples * formatBytes(format) );
+if ( result == -1 ) {
+errorText_ = "RtApiOss::stopStream: audio write error.";
+error( RtAudioError::WARNING );
+}
+}
+result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
+if ( result == -1 ) {
+errorStream_ << "RtApiOss::stopStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
+errorText_ = errorStream_.str();
+goto unlock;
+}
+handle->triggered = false;
+}
+if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
+result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
+if ( result == -1 ) {
+errorStream_ << "RtApiOss::stopStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
+errorText_ = errorStream_.str();
+goto unlock;
+}
+}
+unlock:
+stream_.state = STREAM_STOPPED;
+MUTEX_UNLOCK( &stream_.mutex );
+if ( result != -1 ) return;
+error( RtAudioError::SYSTEM_ERROR );
+}
+void RtApiOss :: abortStream()
+{
+verifyStream();
+if ( stream_.state == STREAM_STOPPED ) {
+errorText_ = "RtApiOss::abortStream(): the stream is already stopped!";
+error( RtAudioError::WARNING );
+return;
+}
+MUTEX_LOCK( &stream_.mutex );
+// The state might change while waiting on a mutex.
+if ( stream_.state == STREAM_STOPPED ) {
+MUTEX_UNLOCK( &stream_.mutex );
+return;
+}
+int result = 0;
+OssHandle *handle = (OssHandle *) stream_.apiHandle;
+if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
+if ( result == -1 ) {
+errorStream_ << "RtApiOss::abortStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
+errorText_ = errorStream_.str();
+goto unlock;
+}
+handle->triggered = false;
+}
+if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
+result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
+if ( result == -1 ) {
+errorStream_ << "RtApiOss::abortStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
+errorText_ = errorStream_.str();
+goto unlock;
+}
+}
+unlock:
+stream_.state = STREAM_STOPPED;
+MUTEX_UNLOCK( &stream_.mutex );
+if ( result != -1 ) return;
+error( RtAudioError::SYSTEM_ERROR );
+}
+void RtApiOss :: callbackEvent()
+{
+OssHandle *handle = (OssHandle *) stream_.apiHandle;
+if ( stream_.state == STREAM_STOPPED ) {
+MUTEX_LOCK( &stream_.mutex );
+pthread_cond_wait( &handle->runnable, &stream_.mutex );
+if ( stream_.state != STREAM_RUNNING ) {
+MUTEX_UNLOCK( &stream_.mutex );
+return;
+}
+MUTEX_UNLOCK( &stream_.mutex );
+}
+if ( stream_.state == STREAM_CLOSED ) {
+errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!";
+error( RtAudioError::WARNING );
+return;
+}
+// Invoke user callback to get fresh output data.
+int doStopStream = 0;
+RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
+double streamTime = getStreamTime();
+RtAudioStreamStatus status = 0;
+if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
+status |= RTAUDIO_OUTPUT_UNDERFLOW;
+handle->xrun[0] = false;
+}
+if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
+status |= RTAUDIO_INPUT_OVERFLOW;
+handle->xrun[1] = false;
+}
+doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
+if ( doStopStream == 2 ) {
+this->abortStream();
+return;
+}
+MUTEX_LOCK( &stream_.mutex );
+// The state might change while waiting on a mutex.
+if ( stream_.state == STREAM_STOPPED ) goto unlock;
+int result;
+char *buffer;
+int samples;
+RtAudioFormat format;
+if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+// Setup parameters and do buffer conversion if necessary.
+if ( stream_.doConvertBuffer[0] ) {
+buffer = stream_.deviceBuffer;
+convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
+samples = stream_.bufferSize * stream_.nDeviceChannels[0];
+format = stream_.deviceFormat[0];
+}
+else {
+buffer = stream_.userBuffer[0];
+samples = stream_.bufferSize * stream_.nUserChannels[0];
+format = stream_.userFormat;
+}
+// Do byte swapping if necessary.
+if ( stream_.doByteSwap[0] )
+byteSwapBuffer( buffer, samples, format );
+if ( stream_.mode == DUPLEX && handle->triggered == false ) {
+int trig = 0;
+ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
+result = write( handle->id[0], buffer, samples * formatBytes(format) );
+trig = PCM_ENABLE_INPUT|PCM_ENABLE_OUTPUT;
+ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
+handle->triggered = true;
+}
+else
+// Write samples to device.
+result = write( handle->id[0], buffer, samples * formatBytes(format) );
+if ( result == -1 ) {
+// We'll assume this is an underrun, though there isn't a
+// specific means for determining that.
+handle->xrun[0] = true;
+errorText_ = "RtApiOss::callbackEvent: audio write error.";
+error( RtAudioError::WARNING );
+// Continue on to input section.
+}
+}
+if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+// Setup parameters.
+if ( stream_.doConvertBuffer[1] ) {
+buffer = stream_.deviceBuffer;
+samples = stream_.bufferSize * stream_.nDeviceChannels[1];
+format = stream_.deviceFormat[1];
+}
+else {
+buffer = stream_.userBuffer[1];
+samples = stream_.bufferSize * stream_.nUserChannels[1];
+format = stream_.userFormat;
+}
+// Read samples from device.
+result = read( handle->id[1], buffer, samples * formatBytes(format) );
+if ( result == -1 ) {
+// We'll assume this is an overrun, though there isn't a
+// specific means for determining that.
+handle->xrun[1] = true;
+errorText_ = "RtApiOss::callbackEvent: audio read error.";
+error( RtAudioError::WARNING );
+goto unlock;
+}
+// Do byte swapping if necessary.
+if ( stream_.doByteSwap[1] )
+byteSwapBuffer( buffer, samples, format );
+// Do buffer conversion if necessary.
+if ( stream_.doConvertBuffer[1] )
+convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
+}
+unlock:
+MUTEX_UNLOCK( &stream_.mutex );
+RtApi::tickStreamTime();
+if ( doStopStream == 1 ) this->stopStream();
+}
+static void *ossCallbackHandler( void *ptr )
+{
+CallbackInfo *info = (CallbackInfo *) ptr;
+RtApiOss *object = (RtApiOss *) info->object;
+bool *isRunning = &info->isRunning;
+#ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
+if (info->doRealtime) {
+std::cerr << "RtAudio oss: " <<
+(sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") <<
+"running realtime scheduling" << std::endl;
+}
+#endif
+while ( *isRunning == true ) {
+pthread_testcancel();
+object->callbackEvent();
+}
+pthread_exit( NULL );
+}
+//******************** End of __LINUX_OSS__ *********************//
+#endif
+// *************************************************** //
+//
+// Protected common (OS-independent) RtAudio methods.
+//
+// *************************************************** //
+// This method can be modified to control the behavior of error
+// message printing.
+void RtApi :: error( RtAudioError::Type type )
+{
+errorStream_.str(""); // clear the ostringstream
+RtAudioErrorCallback errorCallback = (RtAudioErrorCallback) stream_.callbackInfo.errorCallback;
+if ( errorCallback ) {
+// abortStream() can generate new error messages. Ignore them. Just keep original one.
+if ( firstErrorOccurred_ )
+return;
+firstErrorOccurred_ = true;
+const std::string errorMessage = errorText_;
+if ( type != RtAudioError::WARNING && stream_.state != STREAM_STOPPED) {
+stream_.callbackInfo.isRunning = false; // exit from the thread
+abortStream();
+}
+errorCallback( type, errorMessage );
+firstErrorOccurred_ = false;
+return;
+}
+if ( type == RtAudioError::WARNING && showWarnings_ == true )
+std::cerr << '\n' << errorText_ << "\n\n";
+else if ( type != RtAudioError::WARNING )
+throw( RtAudioError( errorText_, type ) );
+}
+void RtApi :: verifyStream()
+{
+if ( stream_.state == STREAM_CLOSED ) {
+errorText_ = "RtApi:: a stream is not open!";
+error( RtAudioError::INVALID_USE );
+}
+}
+void RtApi :: clearStreamInfo()
+{
+stream_.mode = UNINITIALIZED;
+stream_.state = STREAM_CLOSED;
+stream_.sampleRate = 0;
+stream_.bufferSize = 0;
+stream_.nBuffers = 0;
+stream_.userFormat = 0;
+stream_.userInterleaved = true;
+stream_.streamTime = 0.0;
+stream_.apiHandle = 0;
+stream_.deviceBuffer = 0;
+stream_.callbackInfo.callback = 0;
+stream_.callbackInfo.userData = 0;
+stream_.callbackInfo.isRunning = false;
+stream_.callbackInfo.errorCallback = 0;
+for ( int i=0; i<2; i++ ) {
+stream_.device[i] = 11111;
+stream_.doConvertBuffer[i] = false;
+stream_.deviceInterleaved[i] = true;
+stream_.doByteSwap[i] = false;
+stream_.nUserChannels[i] = 0;
+stream_.nDeviceChannels[i] = 0;
+stream_.channelOffset[i] = 0;
+stream_.deviceFormat[i] = 0;
+stream_.latency[i] = 0;
+stream_.userBuffer[i] = 0;
+stream_.convertInfo[i].channels = 0;
+stream_.convertInfo[i].inJump = 0;
+stream_.convertInfo[i].outJump = 0;
+stream_.convertInfo[i].inFormat = 0;
+stream_.convertInfo[i].outFormat = 0;
+stream_.convertInfo[i].inOffset.clear();
+stream_.convertInfo[i].outOffset.clear();
+}
+}
+unsigned int RtApi :: formatBytes( RtAudioFormat format )
+{
+if ( format == RTAUDIO_SINT16 )
+return 2;
+else if ( format == RTAUDIO_SINT32 || format == RTAUDIO_FLOAT32 )
+return 4;
+else if ( format == RTAUDIO_FLOAT64 )
+return 8;
+else if ( format == RTAUDIO_SINT24 )
+return 3;
+else if ( format == RTAUDIO_SINT8 )
+return 1;
+errorText_ = "RtApi::formatBytes: undefined format.";
+error( RtAudioError::WARNING );
+return 0;
+}
+void RtApi :: setConvertInfo( StreamMode mode, unsigned int firstChannel )
+{
+if ( mode == INPUT ) { // convert device to user buffer
+stream_.convertInfo[mode].inJump = stream_.nDeviceChannels[1];
+stream_.convertInfo[mode].outJump = stream_.nUserChannels[1];
+stream_.convertInfo[mode].inFormat = stream_.deviceFormat[1];
+stream_.convertInfo[mode].outFormat = stream_.userFormat;
+}
+else { // convert user to device buffer
+stream_.convertInfo[mode].inJump = stream_.nUserChannels[0];
+stream_.convertInfo[mode].outJump = stream_.nDeviceChannels[0];
+stream_.convertInfo[mode].inFormat = stream_.userFormat;
+stream_.convertInfo[mode].outFormat = stream_.deviceFormat[0];
+}
+if ( stream_.convertInfo[mode].inJump < stream_.convertInfo[mode].outJump )
+stream_.convertInfo[mode].channels = stream_.convertInfo[mode].inJump;
+else
+stream_.convertInfo[mode].channels = stream_.convertInfo[mode].outJump;
+// Set up the interleave/deinterleave offsets.
+if ( stream_.deviceInterleaved[mode] != stream_.userInterleaved ) {
+if ( ( mode == OUTPUT && stream_.deviceInterleaved[mode] ) ||
+( mode == INPUT && stream_.userInterleaved ) ) {
+for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
+stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
+stream_.convertInfo[mode].outOffset.push_back( k );
+stream_.convertInfo[mode].inJump = 1;
+}
+}
+else {
+for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
+stream_.convertInfo[mode].inOffset.push_back( k );
+stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
+stream_.convertInfo[mode].outJump = 1;
+}
+}
+}
+else { // no (de)interleaving
+if ( stream_.userInterleaved ) {
+for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
+stream_.convertInfo[mode].inOffset.push_back( k );
+stream_.convertInfo[mode].outOffset.push_back( k );
+}
+}
+else {
+for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
+stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
+stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
+stream_.convertInfo[mode].inJump = 1;
+stream_.convertInfo[mode].outJump = 1;
+}
+}
+}
+// Add channel offset.
+if ( firstChannel > 0 ) {
+if ( stream_.deviceInterleaved[mode] ) {
+if ( mode == OUTPUT ) {
+for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
+stream_.convertInfo[mode].outOffset[k] += firstChannel;
+}
+else {
+for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
+stream_.convertInfo[mode].inOffset[k] += firstChannel;
+}
+}
+else {
+if ( mode == OUTPUT ) {
+for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
+stream_.convertInfo[mode].outOffset[k] += ( firstChannel * stream_.bufferSize );
+}
+else {
+for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
+stream_.convertInfo[mode].inOffset[k] += ( firstChannel  * stream_.bufferSize );
+}
+}
+}
+}
+void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info )
+{
+// This function does format conversion, input/output channel compensation, and
+// data interleaving/deinterleaving.  24-bit integers are assumed to occupy
+// the lower three bytes of a 32-bit integer.
+// Clear our duplex device output buffer if there are more device outputs than user outputs
+if ( outBuffer == stream_.deviceBuffer && stream_.mode == DUPLEX && info.outJump > info.inJump )
+memset( outBuffer, 0, stream_.bufferSize * info.outJump * formatBytes( info.outFormat ) );
+int j;
+if (info.outFormat == RTAUDIO_FLOAT64) {
+Float64 *out = (Float64 *)outBuffer;
+if (info.inFormat == RTAUDIO_SINT8) {
+signed char *in = (signed char *)inBuffer;
+for (unsigned int i=0; i<stream_.bufferSize; i++) {
+for (j=0; j<info.channels; j++) {
+out[info.outOffset[j]] = (Float64) in[info.inOffset[j]] / 128.0;
+}
+in += info.inJump;
+out += info.outJump;
+}
+}
+else if (info.inFormat == RTAUDIO_SINT16) {
+Int16 *in = (Int16 *)inBuffer;
+for (unsigned int i=0; i<stream_.bufferSize; i++) {
+for (j=0; j<info.channels; j++) {
+out[info.outOffset[j]] = (Float64) in[info.inOffset[j]] / 32768.0;
+}
+in += info.inJump;
+out += info.outJump;
+}
+}
+else if (info.inFormat == RTAUDIO_SINT24) {
+Int24 *in = (Int24 *)inBuffer;
+for (unsigned int i=0; i<stream_.bufferSize; i++) {
+for (j=0; j<info.channels; j++) {
+out[info.outOffset[j]] = (Float64) in[info.inOffset[j]].asInt() / 8388608.0;
+}
+in += info.inJump;
+out += info.outJump;
+}
+}
+else if (info.inFormat == RTAUDIO_SINT32) {
+Int32 *in = (Int32 *)inBuffer;
+for (unsigned int i=0; i<stream_.bufferSize; i++) {
+for (j=0; j<info.channels; j++) {
+out[info.outOffset[j]] = (Float64) in[info.inOffset[j]] / 2147483648.0;
+}
+in += info.inJump;
+out += info.outJump;
+}
+}
+else if (info.inFormat == RTAUDIO_FLOAT32) {
+Float32 *in = (Float32 *)inBuffer;
+for (unsigned int i=0; i<stream_.bufferSize; i++) {
+for (j=0; j<info.channels; j++) {
+out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
+}
+in += info.inJump;
+out += info.outJump;
+}
+}
+else if (info.inFormat == RTAUDIO_FLOAT64) {
+// Channel compensation and/or (de)interleaving only.
+Float64 *in = (Float64 *)inBuffer;
+for (unsigned int i=0; i<stream_.bufferSize; i++) {
+for (j=0; j<info.channels; j++) {
+out[info.outOffset[j]] = in[info.inOffset[j]];
+}
+in += info.inJump;
+out += info.outJump;
+}
+}
+}
+else if (info.outFormat == RTAUDIO_FLOAT32) {
+Float32 *out = (Float32 *)outBuffer;
+if (info.inFormat == RTAUDIO_SINT8) {
+signed char *in = (signed char *)inBuffer;
+for (unsigned int i=0; i<stream_.bufferSize; i++) {
+for (j=0; j<info.channels; j++) {
+out[info.outOffset[j]] = (Float32) in[info.inOffset[j]] / 128.f;
+}
+in += info.inJump;
+out += info.outJump;
+}
+}
+else if (info.inFormat == RTAUDIO_SINT16) {
+Int16 *in = (Int16 *)inBuffer;
+for (unsigned int i=0; i<stream_.bufferSize; i++) {
+for (j=0; j<info.channels; j++) {
+out[info.outOffset[j]] = (Float32) in[info.inOffset[j]] / 32768.f;
+}
+in += info.inJump;
+out += info.outJump;
+}
+}
+else if (info.inFormat == RTAUDIO_SINT24) {
+Int24 *in = (Int24 *)inBuffer;
+for (unsigned int i=0; i<stream_.bufferSize; i++) {
+for (j=0; j<info.channels; j++) {
+out[info.outOffset[j]] = (Float32) in[info.inOffset[j]].asInt() / 8388608.f;
+}
+in += info.inJump;
+out += info.outJump;
+}
+}
+else if (info.inFormat == RTAUDIO_SINT32) {
+Int32 *in = (Int32 *)inBuffer;
+for (unsigned int i=0; i<stream_.bufferSize; i++) {
+for (j=0; j<info.channels; j++) {
+out[info.outOffset[j]] = (Float32) in[info.inOffset[j]] / 2147483648.f;
+}
+in += info.inJump;
+out += info.outJump;
+}
+}
+else if (info.inFormat == RTAUDIO_FLOAT32) {
+// Channel compensation and/or (de)interleaving only.
+Float32 *in = (Float32 *)inBuffer;
+for (unsigned int i=0; i<stream_.bufferSize; i++) {
+for (j=0; j<info.channels; j++) {
+out[info.outOffset[j]] = in[info.inOffset[j]];
+}
+in += info.inJump;
+out += info.outJump;
+}
+}
+else if (info.inFormat == RTAUDIO_FLOAT64) {
+Float64 *in = (Float64 *)inBuffer;
+for (unsigned int i=0; i<stream_.bufferSize; i++) {
+for (j=0; j<info.channels; j++) {
+out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
+}
+in += info.inJump;
+out += info.outJump;
+}
+}
+}
+else if (info.outFormat == RTAUDIO_SINT32) {
+Int32 *out = (Int32 *)outBuffer;
+if (info.inFormat == RTAUDIO_SINT8) {
+signed char *in = (signed char *)inBuffer;
+for (unsigned int i=0; i<stream_.bufferSize; i++) {
+for (j=0; j<info.channels; j++) {
+out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
+out[info.outOffset[j]] <<= 24;
+}
+in += info.inJump;
+out += info.outJump;
+}
+}
+else if (info.inFormat == RTAUDIO_SINT16) {
+Int16 *in = (Int16 *)inBuffer;
+for (unsigned int i=0; i<stream_.bufferSize; i++) {
+for (j=0; j<info.channels; j++) {
+out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
+out[info.outOffset[j]] <<= 16;
+}
+in += info.inJump;
+out += info.outJump;
+}
+}
+else if (info.inFormat == RTAUDIO_SINT24) {
+Int24 *in = (Int24 *)inBuffer;
+for (unsigned int i=0; i<stream_.bufferSize; i++) {
+for (j=0; j<info.channels; j++) {
+out[info.outOffset[j]] = (Int32) in[info.inOffset[j]].asInt();
+out[info.outOffset[j]] <<= 8;
+}
+in += info.inJump;
+out += info.outJump;
+}
+}
+else if (info.inFormat == RTAUDIO_SINT32) {
+// Channel compensation and/or (de)interleaving only.
+Int32 *in = (Int32 *)inBuffer;
+for (unsigned int i=0; i<stream_.bufferSize; i++) {
+for (j=0; j<info.channels; j++) {
+out[info.outOffset[j]] = in[info.inOffset[j]];
+}
+in += info.inJump;
+out += info.outJump;
+}
+}
+else if (info.inFormat == RTAUDIO_FLOAT32) {
+Float32 *in = (Float32 *)inBuffer;
+for (unsigned int i=0; i<stream_.bufferSize; i++) {
+for (j=0; j<info.channels; j++) {
+// Use llround() which returns `long long` which is guaranteed to be at least 64 bits.
+out[info.outOffset[j]] = (Int32) std::min(std::llround(in[info.inOffset[j]] * 2147483648.f), 2147483647LL);
+}
+in += info.inJump;
+out += info.outJump;
+}
+}
+else if (info.inFormat == RTAUDIO_FLOAT64) {
+Float64 *in = (Float64 *)inBuffer;
+for (unsigned int i=0; i<stream_.bufferSize; i++) {
+for (j=0; j<info.channels; j++) {
+out[info.outOffset[j]] = (Int32) std::min(std::llround(in[info.inOffset[j]] * 2147483648.0), 2147483647LL);
+}
+in += info.inJump;
+out += info.outJump;
+}
+}
+}
+else if (info.outFormat == RTAUDIO_SINT24) {
+Int24 *out = (Int24 *)outBuffer;
+if (info.inFormat == RTAUDIO_SINT8) {
+signed char *in = (signed char *)inBuffer;
+for (unsigned int i=0; i<stream_.bufferSize; i++) {
+for (j=0; j<info.channels; j++) {
+out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] << 16);
+//out[info.outOffset[j]] <<= 16;
+}
+in += info.inJump;
+out += info.outJump;
+}
+}
+else if (info.inFormat == RTAUDIO_SINT16) {
+Int16 *in = (Int16 *)inBuffer;
+for (unsigned int i=0; i<stream_.bufferSize; i++) {
+for (j=0; j<info.channels; j++) {
+out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] << 8);
+//out[info.outOffset[j]] <<= 8;
+}
+in += info.inJump;
+out += info.outJump;
+}
+}
+else if (info.inFormat == RTAUDIO_SINT24) {
+// Channel compensation and/or (de)interleaving only.
+Int24 *in = (Int24 *)inBuffer;
+for (unsigned int i=0; i<stream_.bufferSize; i++) {
+for (j=0; j<info.channels; j++) {
+out[info.outOffset[j]] = in[info.inOffset[j]];
+}
+in += info.inJump;
+out += info.outJump;
+}
+}
+else if (info.inFormat == RTAUDIO_SINT32) {
+Int32 *in = (Int32 *)inBuffer;
+for (unsigned int i=0; i<stream_.bufferSize; i++) {
+for (j=0; j<info.channels; j++) {
+out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] >> 8);
+//out[info.outOffset[j]] >>= 8;
+}
+in += info.inJump;
+out += info.outJump;
+}
+}
+else if (info.inFormat == RTAUDIO_FLOAT32) {
+Float32 *in = (Float32 *)inBuffer;
+for (unsigned int i=0; i<stream_.bufferSize; i++) {
+for (j=0; j<info.channels; j++) {
+out[info.outOffset[j]] = (Int32) std::min(std::llround(in[info.inOffset[j]] * 8388608.f), 8388607LL);
+}
+in += info.inJump;
+out += info.outJump;
+}
+}
+else if (info.inFormat == RTAUDIO_FLOAT64) {
+Float64 *in = (Float64 *)inBuffer;
+for (unsigned int i=0; i<stream_.bufferSize; i++) {
+for (j=0; j<info.channels; j++) {
+out[info.outOffset[j]] = (Int32) std::min(std::llround(in[info.inOffset[j]] * 8388608.0), 8388607LL);
+}
+in += info.inJump;
+out += info.outJump;
+}
+}
+}
+else if (info.outFormat == RTAUDIO_SINT16) {
+Int16 *out = (Int16 *)outBuffer;
+if (info.inFormat == RTAUDIO_SINT8) {
+signed char *in = (signed char *)inBuffer;
+for (unsigned int i=0; i<stream_.bufferSize; i++) {
+for (j=0; j<info.channels; j++) {
+out[info.outOffset[j]] = (Int16) in[info.inOffset[j]];
+out[info.outOffset[j]] <<= 8;
+}
+in += info.inJump;
+out += info.outJump;
+}
+}
+else if (info.inFormat == RTAUDIO_SINT16) {
+// Channel compensation and/or (de)interleaving only.
+Int16 *in = (Int16 *)inBuffer;
+for (unsigned int i=0; i<stream_.bufferSize; i++) {
+for (j=0; j<info.channels; j++) {
+out[info.outOffset[j]] = in[info.inOffset[j]];
+}
+in += info.inJump;
+out += info.outJump;
+}
+}
+else if (info.inFormat == RTAUDIO_SINT24) {
+Int24 *in = (Int24 *)inBuffer;
+for (unsigned int i=0; i<stream_.bufferSize; i++) {
+for (j=0; j<info.channels; j++) {
+out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]].asInt() >> 8);
+}
+in += info.inJump;
+out += info.outJump;
+}
+}
+else if (info.inFormat == RTAUDIO_SINT32) {
+Int32 *in = (Int32 *)inBuffer;
+for (unsigned int i=0; i<stream_.bufferSize; i++) {
+for (j=0; j<info.channels; j++) {
+out[info.outOffset[j]] = (Int16) ((in[info.inOffset[j]] >> 16) & 0x0000ffff);
+}
+in += info.inJump;
+out += info.outJump;
+}
+}
+else if (info.inFormat == RTAUDIO_FLOAT32) {
+Float32 *in = (Float32 *)inBuffer;
+for (unsigned int i=0; i<stream_.bufferSize; i++) {
+for (j=0; j<info.channels; j++) {
+out[info.outOffset[j]] = (Int16) std::min(std::llround(in[info.inOffset[j]] * 32768.f), 32767LL);
+}
+in += info.inJump;
+out += info.outJump;
+}
+}
+else if (info.inFormat == RTAUDIO_FLOAT64) {
+Float64 *in = (Float64 *)inBuffer;
+for (unsigned int i=0; i<stream_.bufferSize; i++) {
+for (j=0; j<info.channels; j++) {
+out[info.outOffset[j]] = (Int16) std::min(std::llround(in[info.inOffset[j]] * 32768.0), 32767LL);
+}
+in += info.inJump;
+out += info.outJump;
+}
+}
+}
+else if (info.outFormat == RTAUDIO_SINT8) {
+signed char *out = (signed char *)outBuffer;
+if (info.inFormat == RTAUDIO_SINT8) {
+// Channel compensation and/or (de)interleaving only.
+signed char *in = (signed char *)inBuffer;
+for (unsigned int i=0; i<stream_.bufferSize; i++) {
+for (j=0; j<info.channels; j++) {
+out[info.outOffset[j]] = in[info.inOffset[j]];
+}
+in += info.inJump;
+out += info.outJump;
+}
+}
+if (info.inFormat == RTAUDIO_SINT16) {
+Int16 *in = (Int16 *)inBuffer;
+for (unsigned int i=0; i<stream_.bufferSize; i++) {
+for (j=0; j<info.channels; j++) {
+out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 8) & 0x00ff);
+}
+in += info.inJump;
+out += info.outJump;
+}
+}
+else if (info.inFormat == RTAUDIO_SINT24) {
+Int24 *in = (Int24 *)inBuffer;
+for (unsigned int i=0; i<stream_.bufferSize; i++) {
+for (j=0; j<info.channels; j++) {
+out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]].asInt() >> 16);
+}
+in += info.inJump;
+out += info.outJump;
+}
+}
+else if (info.inFormat == RTAUDIO_SINT32) {
+Int32 *in = (Int32 *)inBuffer;
+for (unsigned int i=0; i<stream_.bufferSize; i++) {
+for (j=0; j<info.channels; j++) {
+out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 24) & 0x000000ff);
+}
+in += info.inJump;
+out += info.outJump;
+}
+}
+else if (info.inFormat == RTAUDIO_FLOAT32) {
+Float32 *in = (Float32 *)inBuffer;
+for (unsigned int i=0; i<stream_.bufferSize; i++) {
+for (j=0; j<info.channels; j++) {
+out[info.outOffset[j]] = (signed char) std::min(std::llround(in[info.inOffset[j]] * 128.f), 127LL);
+}
+in += info.inJump;
+out += info.outJump;
+}
+}
+else if (info.inFormat == RTAUDIO_FLOAT64) {
+Float64 *in = (Float64 *)inBuffer;
+for (unsigned int i=0; i<stream_.bufferSize; i++) {
+for (j=0; j<info.channels; j++) {
+out[info.outOffset[j]] = (signed char) std::min(std::llround(in[info.inOffset[j]] * 128.0), 127LL);
+}
+in += info.inJump;
+out += info.outJump;
+}
+}
+}
+}
+//static inline uint16_t bswap_16(uint16_t x) { return (x>>8) | (x<<8); }
+//static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); }
+//static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); }
+void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format )
+{
+char val;
+char *ptr;
+ptr = buffer;
+if ( format == RTAUDIO_SINT16 ) {
+for ( unsigned int i=0; i<samples; i++ ) {
+// Swap 1st and 2nd bytes.
+val = *(ptr);
+*(ptr) = *(ptr+1);
+*(ptr+1) = val;
+// Increment 2 bytes.
+ptr += 2;
+}
+}
+else if ( format == RTAUDIO_SINT32 ||
+format == RTAUDIO_FLOAT32 ) {
+for ( unsigned int i=0; i<samples; i++ ) {
+// Swap 1st and 4th bytes.
+val = *(ptr);
+*(ptr) = *(ptr+3);
+*(ptr+3) = val;
+// Swap 2nd and 3rd bytes.
+ptr += 1;
+val = *(ptr);
+*(ptr) = *(ptr+1);
+*(ptr+1) = val;
+// Increment 3 more bytes.
+ptr += 3;
+}
+}
+else if ( format == RTAUDIO_SINT24 ) {
+for ( unsigned int i=0; i<samples; i++ ) {
+// Swap 1st and 3rd bytes.
+val = *(ptr);
+*(ptr) = *(ptr+2);
+*(ptr+2) = val;
+// Increment 2 more bytes.
+ptr += 2;
+}
+}
+else if ( format == RTAUDIO_FLOAT64 ) {
+for ( unsigned int i=0; i<samples; i++ ) {
+// Swap 1st and 8th bytes
+val = *(ptr);
+*(ptr) = *(ptr+7);
+*(ptr+7) = val;
+// Swap 2nd and 7th bytes
+ptr += 1;
+val = *(ptr);
+*(ptr) = *(ptr+5);
+*(ptr+5) = val;
+// Swap 3rd and 6th bytes
+ptr += 1;
+val = *(ptr);
+*(ptr) = *(ptr+3);
+*(ptr+3) = val;
+// Swap 4th and 5th bytes
+ptr += 1;
+val = *(ptr);
+*(ptr) = *(ptr+1);
+*(ptr+1) = val;
+// Increment 5 more bytes.
+ptr += 5;
+}
+}
+}
+// Indentation settings for Vim and Emacs
+//
+// Local Variables:
+// c-basic-offset: 2
+// indent-tabs-mode: nil
+// End:
+//
+// vim: et sts=2 sw=2

Mercurial > hg > pub > prymula > com

comparison DPF-Prymula-audioplugins/dpf/distrho/src/jackbridge/rtaudio/RtAudio.cpp @ 3:84e66ea83026