-/************************************************************************/
+/************************************************************************/
/*! \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), SGI, Macintosh OS X (CoreAudio and Jack), and Windows
- (DirectSound and ASIO) operating systems.
+ and OSS), Macintosh OS X (CoreAudio and Jack), and Windows
+ (DirectSound, ASIO and WASAPI) operating systems.
RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/
RtAudio: realtime audio i/o C++ classes
- Copyright (c) 2001-2007 Gary P. Scavone
+ Copyright (c) 2001-2017 Gary P. Scavone
Permission is hereby granted, free of charge, to any person
obtaining a copy of this software and associated documentation files
*/
/************************************************************************/
-// RtAudio: Version 4.0
+// RtAudio: Version 5.0.0
#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;
32000, 44100, 48000, 88200, 96000, 176400, 192000
};
-#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__)
+#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)
-#elif defined(__LINUX_ALSA__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__)
+
+ #include "tchar.h"
+
+ static std::string convertCharPointerToStdString(const char *text)
+ {
+ return std::string(text);
+ }
+
+ static std::string convertCharPointerToStdString(const wchar_t *text)
+ {
+ int length = WideCharToMultiByte(CP_UTF8, 0, text, -1, NULL, 0, NULL, NULL);
+ std::string s( length-1, '\0' );
+ WideCharToMultiByte(CP_UTF8, 0, text, -1, &s[0], length, NULL, NULL);
+ return s;
+ }
+
+#elif defined(__LINUX_ALSA__) || defined(__LINUX_PULSE__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__)
// pthread API
#define MUTEX_INITIALIZE(A) pthread_mutex_init(A, NULL)
#define MUTEX_DESTROY(A) pthread_mutex_destroy(A)
//
// *************************************************** //
-void RtAudio :: getCompiledApi( std::vector<RtAudio::Api> &apis ) throw()
+std::string RtAudio :: getVersion( void )
+{
+ return RTAUDIO_VERSION;
+}
+
+void RtAudio :: getCompiledApi( std::vector<RtAudio::Api> &apis )
{
apis.clear();
#if defined(__UNIX_JACK__)
apis.push_back( UNIX_JACK );
#endif
+#if defined(__LINUX_PULSE__)
+ apis.push_back( LINUX_PULSE );
+#endif
#if defined(__LINUX_ALSA__)
apis.push_back( LINUX_ALSA );
#endif
#if defined(__WINDOWS_ASIO__)
apis.push_back( WINDOWS_ASIO );
#endif
+#if defined(__WINDOWS_WASAPI__)
+ apis.push_back( WINDOWS_WASAPI );
+#endif
#if defined(__WINDOWS_DS__)
apis.push_back( WINDOWS_DS );
#endif
void RtAudio :: openRtApi( RtAudio::Api api )
{
+ if ( rtapi_ )
+ delete rtapi_;
+ rtapi_ = 0;
+
#if defined(__UNIX_JACK__)
if ( api == UNIX_JACK )
rtapi_ = new RtApiJack();
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();
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
}
-RtAudio :: RtAudio( RtAudio::Api api ) throw()
+RtAudio :: RtAudio( RtAudio::Api api )
{
rtapi_ = 0;
getCompiledApi( apis );
for ( unsigned int i=0; i<apis.size(); i++ ) {
openRtApi( apis[i] );
- if ( rtapi_->getDeviceCount() ) break;
+ 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 print out an error message.
- std::cerr << "\nRtAudio: no compiled API support found ... critical error!!\n\n";
+ // 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() throw()
+RtAudio :: ~RtAudio()
{
- delete rtapi_;
+ if ( rtapi_ )
+ delete rtapi_;
}
void RtAudio :: openStream( RtAudio::StreamParameters *outputParameters,
RtAudioFormat format, unsigned int sampleRate,
unsigned int *bufferFrames,
RtAudioCallback callback, void *userData,
- RtAudio::StreamOptions *options )
+ RtAudio::StreamOptions *options,
+ RtAudioErrorCallback errorCallback )
{
return rtapi_->openStream( outputParameters, inputParameters, format,
sampleRate, bufferFrames, callback,
- userData, options );
+ userData, options, errorCallback );
}
// *************************************************** //
stream_.userBuffer[1] = 0;
MUTEX_INITIALIZE( &stream_.mutex );
showWarnings_ = true;
+ firstErrorOccurred_ = false;
}
RtApi :: ~RtApi()
RtAudioFormat format, unsigned int sampleRate,
unsigned int *bufferFrames,
RtAudioCallback callback, void *userData,
- RtAudio::StreamOptions *options )
+ RtAudio::StreamOptions *options,
+ RtAudioErrorCallback errorCallback )
{
if ( stream_.state != STREAM_CLOSED ) {
errorText_ = "RtApi::openStream: a stream is already open!";
- error( RtError::INVALID_USE );
+ 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( RtError::INVALID_USE );
+ 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( RtError::INVALID_USE );
+ error( RtAudioError::INVALID_USE );
+ return;
}
if ( oParams == NULL && iParams == NULL ) {
errorText_ = "RtApi::openStream: input and output StreamParameters structures are both NULL!";
- error( RtError::INVALID_USE );
+ error( RtAudioError::INVALID_USE );
+ return;
}
if ( formatBytes(format) == 0 ) {
errorText_ = "RtApi::openStream: 'format' parameter value is undefined.";
- error( RtError::INVALID_USE );
+ error( RtAudioError::INVALID_USE );
+ return;
}
unsigned int nDevices = getDeviceCount();
oChannels = oParams->nChannels;
if ( oParams->deviceId >= nDevices ) {
errorText_ = "RtApi::openStream: output device parameter value is invalid.";
- error( RtError::INVALID_USE );
+ error( RtAudioError::INVALID_USE );
+ return;
}
}
iChannels = iParams->nChannels;
if ( iParams->deviceId >= nDevices ) {
errorText_ = "RtApi::openStream: input device parameter value is invalid.";
- error( RtError::INVALID_USE );
+ error( RtAudioError::INVALID_USE );
+ return;
}
}
- clearStreamInfo();
bool result;
if ( oChannels > 0 ) {
result = probeDeviceOpen( oParams->deviceId, OUTPUT, oChannels, oParams->firstChannel,
sampleRate, format, bufferFrames, options );
- if ( result == false ) error( RtError::SYSTEM_ERROR );
+ if ( result == false ) {
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
}
if ( iChannels > 0 ) {
sampleRate, format, bufferFrames, options );
if ( result == false ) {
if ( oChannels > 0 ) closeStream();
- error( RtError::SYSTEM_ERROR );
+ 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;
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 )
+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;
#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;
+}
+
// *************************************************** //
//
// 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.
-//
-// The mechanism for querying and setting system parameters was
-// updated (and perhaps simplified) in OS-X version 10.4. However,
-// since 10.4 support is not necessarily available to all users, I've
-// decided not to update the respective code at this time. Perhaps
-// this will happen when Apple makes 10.4 free for everyone. :-)
// A structure to hold various information related to the CoreAudio API
// implementation.
struct CoreHandle {
AudioDeviceID id[2]; // device ids
- UInt32 iStream[2]; // device stream index (first for mono mode)
+#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;
bool internalDrain; // Indicates if stop is initiated from callback or not.
CoreHandle()
- :deviceBuffer(0), drainCounter(0), internalDrain(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
+ :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()
+RtApiCore:: RtApiCore()
{
- // Nothing to do here.
+#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()
{
// Find out how many audio devices there are, if any.
UInt32 dataSize;
- OSStatus result = AudioHardwareGetPropertyInfo( kAudioHardwarePropertyDevices, &dataSize, NULL );
+ AudioObjectPropertyAddress propertyAddress = { kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
+ OSStatus result = AudioObjectGetPropertyDataSize( kAudioObjectSystemObject, &propertyAddress, 0, NULL, &dataSize );
if ( result != noErr ) {
errorText_ = "RtApiCore::getDeviceCount: OS-X error getting device info!";
- error( RtError::WARNING );
+ error( RtAudioError::WARNING );
return 0;
}
AudioDeviceID id;
UInt32 dataSize = sizeof( AudioDeviceID );
- OSStatus result = AudioHardwareGetProperty( kAudioHardwarePropertyDefaultInputDevice,
- &dataSize, &id );
-
+ AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultInputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
+ OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id );
if ( result != noErr ) {
errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device.";
- error( RtError::WARNING );
+ error( RtAudioError::WARNING );
return 0;
}
dataSize *= nDevices;
AudioDeviceID deviceList[ nDevices ];
- result = AudioHardwareGetProperty( kAudioHardwarePropertyDevices, &dataSize, (void *) &deviceList );
+ property.mSelector = kAudioHardwarePropertyDevices;
+ result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList );
if ( result != noErr ) {
errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device IDs.";
- error( RtError::WARNING );
+ error( RtAudioError::WARNING );
return 0;
}
if ( id == deviceList[i] ) return i;
errorText_ = "RtApiCore::getDefaultInputDevice: No default device found!";
- error( RtError::WARNING );
+ error( RtAudioError::WARNING );
return 0;
}
AudioDeviceID id;
UInt32 dataSize = sizeof( AudioDeviceID );
- OSStatus result = AudioHardwareGetProperty( kAudioHardwarePropertyDefaultOutputDevice,
- &dataSize, &id );
-
+ 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( RtError::WARNING );
+ error( RtAudioError::WARNING );
return 0;
}
- dataSize *= nDevices;
+ dataSize = sizeof( AudioDeviceID ) * nDevices;
AudioDeviceID deviceList[ nDevices ];
- result = AudioHardwareGetProperty( kAudioHardwarePropertyDevices, &dataSize, (void *) &deviceList );
+ 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( RtError::WARNING );
+ error( RtAudioError::WARNING );
return 0;
}
if ( id == deviceList[i] ) return i;
errorText_ = "RtApiCore::getDefaultOutputDevice: No default device found!";
- error( RtError::WARNING );
+ error( RtAudioError::WARNING );
return 0;
}
unsigned int nDevices = getDeviceCount();
if ( nDevices == 0 ) {
errorText_ = "RtApiCore::getDeviceInfo: no devices found!";
- error( RtError::INVALID_USE );
+ error( RtAudioError::INVALID_USE );
+ return info;
}
if ( device >= nDevices ) {
errorText_ = "RtApiCore::getDeviceInfo: device ID is invalid!";
- error( RtError::INVALID_USE );
+ error( RtAudioError::INVALID_USE );
+ return info;
}
AudioDeviceID deviceList[ nDevices ];
UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;
- OSStatus result = AudioHardwareGetProperty( kAudioHardwarePropertyDevices, &dataSize, (void *) &deviceList );
+ AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
+ kAudioObjectPropertyScopeGlobal,
+ kAudioObjectPropertyElementMaster };
+ OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property,
+ 0, NULL, &dataSize, (void *) &deviceList );
if ( result != noErr ) {
errorText_ = "RtApiCore::getDeviceInfo: OS-X system error getting device IDs.";
- error( RtError::WARNING );
+ error( RtAudioError::WARNING );
return info;
}
// Get the device name.
info.name.erase();
- char name[256];
- dataSize = 256;
- result = AudioDeviceGetProperty( id, 0, false,
- kAudioDevicePropertyDeviceManufacturer,
- &dataSize, name );
-
+ CFStringRef cfname;
+ dataSize = sizeof( CFStringRef );
+ property.mSelector = kAudioObjectPropertyManufacturer;
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname );
if ( result != noErr ) {
errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device manufacturer.";
errorText_ = errorStream_.str();
- error( RtError::WARNING );
+ error( RtAudioError::WARNING );
return info;
}
- info.name.append( (const char *)name, strlen(name) );
+
+ //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);
- dataSize = 256;
- result = AudioDeviceGetProperty( id, 0, false,
- kAudioDevicePropertyDeviceName,
- &dataSize, name );
+ property.mSelector = kAudioObjectPropertyName;
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname );
if ( result != noErr ) {
errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device name.";
errorText_ = errorStream_.str();
- error( RtError::WARNING );
+ 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;
- result = AudioDeviceGetPropertyInfo( id, 0, false,
- kAudioDevicePropertyStreamConfiguration,
- &dataSize, NULL );
- if (result != noErr || dataSize == 0) {
+ property.mSelector = kAudioDevicePropertyStreamConfiguration;
+ property.mScope = kAudioDevicePropertyScopeOutput;
+ // property.mElement = kAudioObjectPropertyElementWildcard;
+ dataSize = 0;
+ result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
+ if ( result != noErr || dataSize == 0 ) {
errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration info for device (" << device << ").";
errorText_ = errorStream_.str();
- error( RtError::WARNING );
+ error( RtAudioError::WARNING );
return info;
}
bufferList = (AudioBufferList *) malloc( dataSize );
if ( bufferList == NULL ) {
errorText_ = "RtApiCore::getDeviceInfo: memory error allocating output AudioBufferList.";
- error( RtError::WARNING );
+ error( RtAudioError::WARNING );
return info;
}
- result = AudioDeviceGetProperty( id, 0, false,
- kAudioDevicePropertyStreamConfiguration,
- &dataSize, bufferList );
- if ( result != noErr ) {
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
+ if ( result != noErr || dataSize == 0 ) {
free( bufferList );
errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration for device (" << device << ").";
errorText_ = errorStream_.str();
- error( RtError::WARNING );
+ error( RtAudioError::WARNING );
return info;
}
free( bufferList );
// Get the input stream "configuration".
- result = AudioDeviceGetPropertyInfo( id, 0, true,
- kAudioDevicePropertyStreamConfiguration,
- &dataSize, NULL );
- if (result != noErr || dataSize == 0) {
+ property.mScope = kAudioDevicePropertyScopeInput;
+ result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
+ if ( result != noErr || dataSize == 0 ) {
errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration info for device (" << device << ").";
errorText_ = errorStream_.str();
- error( RtError::WARNING );
+ error( RtAudioError::WARNING );
return info;
}
bufferList = (AudioBufferList *) malloc( dataSize );
if ( bufferList == NULL ) {
errorText_ = "RtApiCore::getDeviceInfo: memory error allocating input AudioBufferList.";
- error( RtError::WARNING );
+ error( RtAudioError::WARNING );
return info;
}
- result = AudioDeviceGetProperty( id, 0, true,
- kAudioDevicePropertyStreamConfiguration,
- &dataSize, bufferList );
- if ( result != noErr ) {
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
+ if (result != noErr || dataSize == 0) {
free( bufferList );
errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration for device (" << device << ").";
errorText_ = errorStream_.str();
- error( RtError::WARNING );
+ error( RtAudioError::WARNING );
return info;
}
if ( info.outputChannels == 0 ) isInput = true;
// Determine the supported sample rates.
- result = AudioDeviceGetPropertyInfo( id, 0, isInput,
- kAudioDevicePropertyAvailableNominalSampleRates,
- &dataSize, NULL );
-
+ property.mSelector = kAudioDevicePropertyAvailableNominalSampleRates;
+ if ( isInput == false ) property.mScope = kAudioDevicePropertyScopeOutput;
+ result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
if ( result != kAudioHardwareNoError || dataSize == 0 ) {
errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rate info.";
errorText_ = errorStream_.str();
- error( RtError::WARNING );
+ error( RtAudioError::WARNING );
return info;
}
UInt32 nRanges = dataSize / sizeof( AudioValueRange );
AudioValueRange rangeList[ nRanges ];
- result = AudioDeviceGetProperty( id, 0, isInput,
- kAudioDevicePropertyAvailableNominalSampleRates,
- &dataSize, &rangeList );
-
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &rangeList );
if ( result != kAudioHardwareNoError ) {
errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rates.";
errorText_ = errorStream_.str();
- error( RtError::WARNING );
+ error( RtAudioError::WARNING );
return info;
}
- Float64 minimumRate = 100000000.0, maximumRate = 0.0;
+ // 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 < minimumRate ) minimumRate = rangeList[i].mMinimum;
- if ( rangeList[i].mMaximum > maximumRate ) maximumRate = rangeList[i].mMaximum;
+ 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;
+ }
}
- info.sampleRates.clear();
- 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 ( 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( RtError::WARNING );
+ error( RtAudioError::WARNING );
return info;
}
// no interest to the client.
info.nativeFormats = RTAUDIO_FLOAT32;
- if ( getDefaultOutputDevice() == device )
- info.isDefaultOutput = true;
- if ( getDefaultInputDevice() == device )
- info.isDefaultInput = true;
+ if ( info.outputChannels > 0 )
+ if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true;
+ if ( info.inputChannels > 0 )
+ if ( getDefaultInputDevice() == device ) info.isDefaultInput = true;
info.probed = true;
return info;
}
-OSStatus callbackHandler( AudioDeviceID inDevice,
- const AudioTimeStamp* inNow,
- const AudioBufferList* inInputData,
- const AudioTimeStamp* inInputTime,
- AudioBufferList* outOutputData,
- const AudioTimeStamp* inOutputTime,
- void* infoPointer )
+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;
return kAudioHardwareNoError;
}
-OSStatus deviceListener( AudioDeviceID inDevice,
- UInt32 channel,
- Boolean isInput,
- AudioDevicePropertyID propertyID,
- void* handlePointer )
+static OSStatus xrunListener( AudioObjectID /*inDevice*/,
+ UInt32 nAddresses,
+ const AudioObjectPropertyAddress properties[],
+ void* handlePointer )
{
CoreHandle *handle = (CoreHandle *) handlePointer;
- if ( propertyID == kAudioDeviceProcessorOverload ) {
- if ( isInput )
- handle->xrun[1] = true;
- else
- handle->xrun[0] = true;
+ 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 bool hasProperty( AudioDeviceID id, UInt32 channel, bool isInput, AudioDevicePropertyID property )
+static OSStatus rateListener( AudioObjectID inDevice,
+ UInt32 /*nAddresses*/,
+ const AudioObjectPropertyAddress /*properties*/[],
+ void* ratePointer )
{
- OSStatus result = AudioDeviceGetPropertyInfo( id, channel, isInput, property, NULL, NULL );
- return result == 0;
+ 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,
AudioDeviceID deviceList[ nDevices ];
UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;
- OSStatus result = AudioHardwareGetProperty( kAudioHardwarePropertyDevices, &dataSize, (void *) &deviceList );
+ 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;
// Setup for stream mode.
bool isInput = false;
- if ( mode == INPUT ) isInput = true;
-
- // Set or disable "hog" mode.
- dataSize = sizeof( UInt32 );
- UInt32 doHog = 0;
- if ( options && options->flags & RTAUDIO_HOG_DEVICE ) doHog = 1;
- result = AudioHardwareSetProperty( kAudioHardwarePropertyHogModeIsAllowed, dataSize, &doHog );
- if ( result != noErr ) {
- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting 'hog' state!";
- errorText_ = errorStream_.str();
- return FAILURE;
+ if ( mode == INPUT ) {
+ isInput = true;
+ property.mScope = kAudioDevicePropertyScopeInput;
}
+ else
+ property.mScope = kAudioDevicePropertyScopeOutput;
// Get the stream "configuration".
- AudioBufferList *bufferList;
- result = AudioDeviceGetPropertyInfo( id, 0, isInput,
- kAudioDevicePropertyStreamConfiguration,
- &dataSize, NULL );
- if (result != noErr || dataSize == 0) {
+ 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;
return FAILURE;
}
- result = AudioDeviceGetProperty( id, 0, isInput,
- kAudioDevicePropertyStreamConfiguration,
- &dataSize, bufferList );
- if ( result != noErr ) {
+ 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 a stream that contains the desired number of
+ // 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 currently only supports the use of one stream
- // of interleaved data or multiple consecutive single-channel
- // streams. Thus, our search below is limited to these two
- // contexts.
- unsigned int streamChannels = 0, nStreams = 0;
- UInt32 iChannel = 0, iStream = 0;
- unsigned int offsetCounter = firstChannel;
- stream_.deviceInterleaved[mode] = true;
- nStreams = bufferList->mNumberBuffers;
+ // 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 ) {
- iChannel += offsetCounter;
+ firstStream = iStream;
+ channelOffset = offsetCounter;
foundStream = true;
break;
}
if ( streamChannels > offsetCounter ) break;
offsetCounter -= streamChannels;
- iChannel += streamChannels;
}
- // If we didn't find a single stream above, see if we can meet
- // the channel specification in mono mode (i.e. using separate
- // non-interleaved buffers). This can only work if there are N
- // consecutive one-channel streams, where N is the number of
- // desired channels (+ channel offset).
+ // If we didn't find a single stream above, then we should be able
+ // to meet the channel specification with multiple streams.
if ( foundStream == false ) {
- unsigned int counter = 0;
+ monoMode = true;
offsetCounter = firstChannel;
- iChannel = 0;
for ( iStream=0; iStream<nStreams; iStream++ ) {
streamChannels = bufferList->mBuffers[iStream].mNumberChannels;
- if ( offsetCounter ) {
- if ( streamChannels > offsetCounter ) break;
- offsetCounter -= streamChannels;
- }
- else if ( streamChannels == 1 )
- counter++;
- else
- counter = 0;
- if ( counter == channels ) {
- iStream -= channels - 1;
- iChannel -= channels - 1;
- stream_.deviceInterleaved[mode] = false;
- foundStream = true;
- break;
- }
- iChannel += streamChannels;
+ if ( streamChannels > offsetCounter ) break;
+ offsetCounter -= streamChannels;
}
- }
- free( bufferList );
- if ( foundStream == false ) {
- errorStream_ << "RtApiCore::probeDeviceOpen: unable to find OS-X stream on device (" << device << ") for requested channels.";
- errorText_ = errorStream_.str();
- return FAILURE;
+ 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 );
- result = AudioDeviceGetProperty( id, 0, isInput,
- kAudioDevicePropertyBufferFrameSizeRange,
- &dataSize, &bufferRange );
+ 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();
else if ( bufferRange.mMaximum < *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMaximum;
if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) *bufferSize = (unsigned long) bufferRange.mMinimum;
- // Set the buffer size. For mono mode, I'm assuming we only need to
- // make this setting for the master channel.
+ // 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 );
- result = AudioDeviceSetProperty( id, NULL, 0, isInput,
- kAudioDevicePropertyBufferFrameSize,
- dataSize, &theSize );
+ 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 << ").";
stream_.bufferSize = *bufferSize;
stream_.nBuffers = 1;
- // Get the stream ID(s) so we can set the stream format. In mono
- // mode, we'll have to do this for each stream (channel).
- AudioStreamID streamIDs[ nStreams ];
- dataSize = nStreams * sizeof( AudioStreamID );
- result = AudioDeviceGetProperty( id, 0, isInput,
- kAudioDevicePropertyStreams,
- &dataSize, &streamIDs );
+ // 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 stream ID(s) for device (" << device << ").";
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting current sample rate.";
errorText_ = errorStream_.str();
return FAILURE;
}
- // Now set the stream format. Also, check the physical format of the
- // device and change that if necessary.
- AudioStreamBasicDescription description;
- dataSize = sizeof( AudioStreamBasicDescription );
- if ( stream_.deviceInterleaved[mode] ) nStreams = 1;
- else nStreams = channels;
-
- bool updateFormat;
- for ( unsigned int i=0; i<nStreams; i++ ) {
-
- result = AudioStreamGetProperty( streamIDs[iStream+i], 0,
- kAudioStreamPropertyVirtualFormat,
- &dataSize, &description );
+ // 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 ) << ") getting stream format for device (" << device << ").";
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate property listener 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.
- updateFormat = false;
- if ( fabs( description.mSampleRate - (double)sampleRate ) > 1.0 ) {
- description.mSampleRate = (double) sampleRate;
- updateFormat = true;
+ 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;
}
- if ( description.mFormatID != kAudioFormatLinearPCM ) {
- description.mFormatID = kAudioFormatLinearPCM;
- updateFormat = true;
+ // 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 );
}
- if ( updateFormat ) {
- result = AudioStreamSetProperty( streamIDs[iStream+i], NULL, 0,
- kAudioStreamPropertyVirtualFormat,
- dataSize, &description );
- if ( result != noErr ) {
- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate or data format for device (" << device << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
- }
+ // Remove the property listener.
+ AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate );
- // Now check the physical format.
- result = AudioStreamGetProperty( streamIDs[iStream+i], 0,
- kAudioStreamPropertyPhysicalFormat,
- &dataSize, &description );
- if ( result != noErr ) {
- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream physical format for device (" << device << ").";
+ if ( microCounter > 5000000 ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: timeout waiting for sample rate update for device (" << device << ").";
errorText_ = errorStream_.str();
return FAILURE;
}
+ }
- if ( description.mFormatID != kAudioFormatLinearPCM || description.mBitsPerChannel < 24 ) {
- description.mFormatID = kAudioFormatLinearPCM;
- AudioStreamBasicDescription testDescription = description;
- unsigned long formatFlags;
+ // 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;
+ }
- // We'll try higher bit rates first and then work our way down.
- testDescription.mBitsPerChannel = 32;
- formatFlags = description.mFormatFlags | kLinearPCMFormatFlagIsFloat & ~kLinearPCMFormatFlagIsSignedInteger;
- testDescription.mFormatFlags = formatFlags;
- result = AudioStreamSetProperty( streamIDs[iStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription );
- if ( result == noErr ) continue;
+ // 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;
+ }
- testDescription = description;
- testDescription.mBitsPerChannel = 32;
- formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger) & ~kLinearPCMFormatFlagIsFloat;
- testDescription.mFormatFlags = formatFlags;
- result = AudioStreamSetProperty( streamIDs[iStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription );
- if ( result == noErr ) continue;
+ if ( description.mFormatID != kAudioFormatLinearPCM ) {
+ description.mFormatID = kAudioFormatLinearPCM;
+ updateFormat = true;
+ }
- testDescription = description;
- testDescription.mBitsPerChannel = 24;
- testDescription.mFormatFlags = formatFlags;
- result = AudioStreamSetProperty( streamIDs[iStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription );
- if ( result == noErr ) continue;
+ 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;
+ }
+ }
- testDescription = description;
- testDescription.mBitsPerChannel = 16;
- testDescription.mFormatFlags = formatFlags;
- result = AudioStreamSetProperty( streamIDs[iStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription );
- if ( result == noErr ) continue;
+ // 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 = 8;
- testDescription.mFormatFlags = formatFlags;
- result = AudioStreamSetProperty( streamIDs[iStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription );
- if ( result != noErr ) {
- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting physical data format for device (" << device << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
+ 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;
}
}
- }
- // Get the stream latency. There can be latency in both the device
- // and the stream. First, attempt to get the device latency on the
- // master channel or the first open channel. Errors that might
- // occur here are not deemed critical.
- UInt32 latency, channel = 0;
+ 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 );
- AudioDevicePropertyID property = kAudioDevicePropertyLatency;
- for ( int i=0; i<2; i++ ) {
- if ( hasProperty( id, channel, isInput, property ) == true ) break;
- channel = iChannel + 1 + i;
- }
- if ( channel <= iChannel + 1 ) {
- result = AudioDeviceGetProperty( id, channel, isInput, property, &dataSize, &latency );
+ 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( RtError::WARNING );
+ error( RtAudioError::WARNING );
}
}
- // Now try to get the stream latency. For "mono" mode, I assume the
- // latency is equal for all single-channel streams.
- result = AudioStreamGetProperty( streamIDs[iStream], 0, property, &dataSize, &latency );
- if ( result == kAudioHardwareNoError ) stream_.latency[ mode ] += latency;
- else {
- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream latency for device (" << device << ").";
- errorText_ = errorStream_.str();
- error( RtError::WARNING );
- }
-
// 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_.userFormat = format;
stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
- if ( stream_.deviceInterleaved[mode] )
+ if ( streamCount == 1 )
stream_.nDeviceChannels[mode] = description.mChannelsPerFrame;
- else // mono mode
+ else // multiple streams
stream_.nDeviceChannels[mode] = channels;
stream_.nUserChannels[mode] = channels;
- stream_.channelOffset[mode] = iChannel; // offset within a CoreAudio stream
+ 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;
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 )
+ 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.
}
else
handle = (CoreHandle *) stream_.apiHandle;
- handle->iStream[mode] = iStream;
+ handle->iStream[mode] = firstStream;
+ handle->nStreams[mode] = streamCount;
handle->id[mode] = id;
// Allocate necessary internal buffers.
- unsigned long bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
- stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+ 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 the device
- // buffers are non-interleaved ("mono" mode).
- if ( !stream_.deviceInterleaved[mode] && stream_.doConvertBuffer[mode] ) {
+ // "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] );
errorText_ = "RtApiCore::probeDeviceOpen: error allocating device buffer memory.";
goto error;
}
-
- // Save a pointer to our own device buffer in the CoreHandle
- // structure because we may need to use the stream_.deviceBuffer
- // variable to point to the CoreAudio buffer before buffer
- // conversion (if we have a duplex stream with two different
- // conversion schemes).
- handle->deviceBuffer = stream_.deviceBuffer;
}
}
stream_.state = STREAM_STOPPED;
stream_.callbackInfo.object = (void *) this;
- // Setup the buffer conversion information structure. We override
- // the channel offset value and perform our own setting for that
- // here.
+ // Setup the buffer conversion information structure.
if ( stream_.doConvertBuffer[mode] ) {
- setConvertInfo( mode, 0 );
-
- // Add channel offset for interleaved channels.
- if ( firstChannel > 0 && 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;
- }
- }
+ 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();
}
// Setup the device property listener for over/underload.
- result = AudioDeviceAddPropertyListener( id, 0, isInput,
- kAudioDeviceProcessorOverload,
- deviceListener, (void *) handle );
+ property.mSelector = kAudioDeviceProcessorOverload;
+ property.mScope = kAudioObjectPropertyScopeGlobal;
+ result = AudioObjectAddPropertyListener( id, &property, xrunListener, (void *) handle );
return SUCCESS;
stream_.deviceBuffer = 0;
}
+ stream_.state = STREAM_CLOSED;
return FAILURE;
}
{
if ( stream_.state == STREAM_CLOSED ) {
errorText_ = "RtApiCore::closeStream(): no open stream to close!";
- error( RtError::WARNING );
+ error( RtAudioError::WARNING );
return;
}
CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ if (handle) {
+ AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
+ kAudioObjectPropertyScopeGlobal,
+ kAudioObjectPropertyElementMaster };
+
+ property.mSelector = kAudioDeviceProcessorOverload;
+ property.mScope = kAudioObjectPropertyScopeGlobal;
+ if (AudioObjectRemovePropertyListener( handle->id[0], &property, xrunListener, (void *) handle ) != noErr) {
+ errorText_ = "RtApiCore::closeStream(): error removing property listener!";
+ error( RtAudioError::WARNING );
+ }
+ }
if ( stream_.state == STREAM_RUNNING )
AudioDeviceStop( handle->id[0], callbackHandler );
+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
+ AudioDeviceDestroyIOProcID( handle->id[0], handle->procId[0] );
+#else
+ // deprecated in favor of AudioDeviceDestroyIOProcID()
AudioDeviceRemoveIOProc( handle->id[0], callbackHandler );
+#endif
}
if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
+ if (handle) {
+ AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
+ kAudioObjectPropertyScopeGlobal,
+ kAudioObjectPropertyElementMaster };
+
+ property.mSelector = kAudioDeviceProcessorOverload;
+ property.mScope = kAudioObjectPropertyScopeGlobal;
+ if (AudioObjectRemovePropertyListener( handle->id[1], &property, xrunListener, (void *) handle ) != noErr) {
+ errorText_ = "RtApiCore::closeStream(): error removing property listener!";
+ error( RtAudioError::WARNING );
+ }
+ }
if ( stream_.state == STREAM_RUNNING )
AudioDeviceStop( handle->id[1], callbackHandler );
+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
+ AudioDeviceDestroyIOProcID( handle->id[1], handle->procId[1] );
+#else
+ // deprecated in favor of AudioDeviceDestroyIOProcID()
AudioDeviceRemoveIOProc( handle->id[1], callbackHandler );
+#endif
}
for ( int i=0; i<2; i++ ) {
}
}
- if ( handle->deviceBuffer ) {
- free( handle->deviceBuffer );
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
stream_.deviceBuffer = 0;
}
verifyStream();
if ( stream_.state == STREAM_RUNNING ) {
errorText_ = "RtApiCore::startStream(): the stream is already running!";
- error( RtError::WARNING );
+ error( RtAudioError::WARNING );
return;
}
- MUTEX_LOCK( &stream_.mutex );
-
OSStatus result = noErr;
CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
stream_.state = STREAM_RUNNING;
unlock:
- MUTEX_UNLOCK( &stream_.mutex );
-
if ( result == noErr ) return;
- error( RtError::SYSTEM_ERROR );
+ error( RtAudioError::SYSTEM_ERROR );
}
void RtApiCore :: stopStream( void )
verifyStream();
if ( stream_.state == STREAM_STOPPED ) {
errorText_ = "RtApiCore::stopStream(): the stream is already stopped!";
- error( RtError::WARNING );
+ error( RtAudioError::WARNING );
return;
}
- MUTEX_LOCK( &stream_.mutex );
-
OSStatus result = noErr;
CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
if ( handle->drainCounter == 0 ) {
- handle->drainCounter = 1;
+ handle->drainCounter = 2;
pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
}
}
}
- unlock:
- MUTEX_UNLOCK( &stream_.mutex );
-
stream_.state = STREAM_STOPPED;
+
+ unlock:
if ( result == noErr ) return;
- error( RtError::SYSTEM_ERROR );
+ error( RtAudioError::SYSTEM_ERROR );
}
void RtApiCore :: abortStream( void )
verifyStream();
if ( stream_.state == STREAM_STOPPED ) {
errorText_ = "RtApiCore::abortStream(): the stream is already stopped!";
- error( RtError::WARNING );
+ error( RtAudioError::WARNING );
return;
}
CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
- handle->drainCounter = 1;
+ 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 ) return SUCCESS;
+ 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( RtError::WARNING );
+ error( RtAudioError::WARNING );
return FAILURE;
}
// Check if we were draining the stream and signal is finished.
if ( handle->drainCounter > 3 ) {
- if ( handle->internalDrain == false )
+ 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 );
- else
- stopStream();
return SUCCESS;
}
- MUTEX_LOCK( &stream_.mutex );
-
AudioDeviceID outputDevice = handle->id[0];
// Invoke user callback to get fresh output data UNLESS we are
status |= RTAUDIO_INPUT_OVERFLOW;
handle->xrun[1] = false;
}
- handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1],
- stream_.bufferSize, streamTime, status, info->userData );
- if ( handle->drainCounter == 2 ) {
- MUTEX_UNLOCK( &stream_.mutex );
+
+ 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 ( handle->drainCounter == 1 )
+ 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 ( stream_.deviceInterleaved[0] ) {
+ if ( handle->nStreams[0] == 1 ) {
memset( outBufferList->mBuffers[handle->iStream[0]].mData,
0,
outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );
}
- else {
- for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
+ 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 ( stream_.doConvertBuffer[0] ) {
-
- if ( stream_.deviceInterleaved[0] )
- stream_.deviceBuffer = (char *) outBufferList->mBuffers[handle->iStream[0]].mData;
- else
- stream_.deviceBuffer = handle->deviceBuffer;
-
- convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
-
- if ( !stream_.deviceInterleaved[0] ) {
- UInt32 bufferBytes = outBufferList->mBuffers[handle->iStream[0]].mDataByteSize;
- for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
- memcpy( outBufferList->mBuffers[handle->iStream[0]+i].mData,
- &stream_.deviceBuffer[i*bufferBytes], bufferBytes );
- }
+ 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 {
- if ( stream_.deviceInterleaved[0] ) {
+ else { // copy from user buffer
memcpy( outBufferList->mBuffers[handle->iStream[0]].mData,
stream_.userBuffer[0],
outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );
}
- else {
+ }
+ 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_.nDeviceChannels[0]; i++ ) {
+ for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
memcpy( outBufferList->mBuffers[handle->iStream[0]+i].mData,
- &stream_.userBuffer[0][i*bufferBytes], bufferBytes );
+ (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 ( handle->drainCounter ) {
- handle->drainCounter++;
- goto unlock;
+ 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;
+ }
+ }
}
}
- AudioDeviceID inputDevice = handle->id[1];
+ // 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 ( stream_.doConvertBuffer[1] ) {
+ 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] )
- stream_.deviceBuffer = (char *) inBufferList->mBuffers[handle->iStream[1]].mData;
- else {
- stream_.deviceBuffer = (char *) handle->deviceBuffer;
+ if ( stream_.deviceInterleaved[1] == false ) { // mono mode
UInt32 bufferBytes = inBufferList->mBuffers[handle->iStream[1]].mDataByteSize;
- for ( unsigned int i=0; i<stream_.nDeviceChannels[1]; i++ ) {
- memcpy( &stream_.deviceBuffer[i*bufferBytes],
+ 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];
+ }
- convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[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;
+ }
- }
- else {
- memcpy( stream_.userBuffer[1],
- inBufferList->mBuffers[handle->iStream[1]].mData,
- inBufferList->mBuffers[handle->iStream[1]].mDataByteSize );
+ // 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 );
+ //MUTEX_UNLOCK( &stream_.mutex );
RtApi::tickStreamTime();
return SUCCESS;
const char* RtApiCore :: getErrorCode( OSStatus code )
{
- switch( code ) {
+ switch( code ) {
case kAudioHardwareNotRunningError:
return "kAudioHardwareNotRunningError";
default:
return "CoreAudio unknown error";
- }
+ }
}
-//******************** End of __MACOSX_CORE__ *********************//
+ //******************** End of __MACOSX_CORE__ *********************//
#endif
#if defined(__UNIX_JACK__)
#include <jack/jack.h>
#include <unistd.h>
+#include <cstdio>
// A structure to hold various information related to the Jack API
// implementation.
: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()
unsigned int RtApiJack :: getDeviceCount( void )
{
// See if we can become a jack client.
- jack_client_t *client = jack_client_new( "RtApiJackCount" );
+ jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption;
+ jack_status_t *status = NULL;
+ jack_client_t *client = jack_client_open( "RtApiJackCount", options, status );
if ( client == 0 ) return 0;
const char **ports;
std::string port, previousPort;
unsigned int nChannels = 0, nDevices = 0;
- ports = jack_get_ports( client, NULL, NULL, 0 );
+ ports = jack_get_ports( client, NULL, JACK_DEFAULT_AUDIO_TYPE, 0 );
if ( ports ) {
// Parse the port names up to the first colon (:).
- unsigned int iColon = 0;
+ size_t iColon = 0;
do {
port = (char *) ports[ nChannels ];
iColon = port.find(":");
RtAudio::DeviceInfo info;
info.probed = false;
- jack_client_t *client = jack_client_new( "RtApiJackInfo" );
+ jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption
+ jack_status_t *status = NULL;
+ jack_client_t *client = jack_client_open( "RtApiJackInfo", options, status );
if ( client == 0 ) {
errorText_ = "RtApiJack::getDeviceInfo: Jack server not found or connection error!";
- error( RtError::WARNING );
+ error( RtAudioError::WARNING );
return info;
}
const char **ports;
std::string port, previousPort;
unsigned int nPorts = 0, nDevices = 0;
- ports = jack_get_ports( client, NULL, NULL, 0 );
+ ports = jack_get_ports( client, NULL, JACK_DEFAULT_AUDIO_TYPE, 0 );
if ( ports ) {
// Parse the port names up to the first colon (:).
- unsigned int iColon = 0;
+ size_t iColon = 0;
do {
port = (char *) ports[ nPorts ];
iColon = port.find(":");
}
if ( device >= nDevices ) {
+ jack_client_close( client );
errorText_ = "RtApiJack::getDeviceInfo: device ID is invalid!";
- error( RtError::INVALID_USE );
+ error( RtAudioError::INVALID_USE );
+ return info;
}
// Get the current jack server sample rate.
info.sampleRates.clear();
- info.sampleRates.push_back( jack_get_sample_rate( client ) );
+
+ info.preferredSampleRate = jack_get_sample_rate( client );
+ info.sampleRates.push_back( info.preferredSampleRate );
// Count the available ports containing the client name as device
// channels. Jack "input ports" equal RtAudio output channels.
unsigned int nChannels = 0;
- ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsInput );
+ ports = jack_get_ports( client, info.name.c_str(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput );
if ( ports ) {
while ( ports[ nChannels ] ) nChannels++;
free( ports );
// Jack "output ports" equal RtAudio input channels.
nChannels = 0;
- ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsOutput );
+ ports = jack_get_ports( client, info.name.c_str(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput );
if ( ports ) {
while ( ports[ nChannels ] ) nChannels++;
free( ports );
if ( info.outputChannels == 0 && info.inputChannels == 0 ) {
jack_client_close(client);
errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!";
- error( RtError::WARNING );
+ error( RtAudioError::WARNING );
return info;
}
return info;
}
-int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer )
+static int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer )
{
CallbackInfo *info = (CallbackInfo *) infoPointer;
return 0;
}
-void jackShutdown( void *infoPointer )
+// 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;
// other problem occurred and we should close the stream.
if ( object->isStreamRunning() == false ) return;
- object->closeStream();
+ 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;
}
-int jackXrun( void *infoPointer )
+static int jackXrun( void *infoPointer )
{
- JackHandle *handle = (JackHandle *) infoPointer;
+ JackHandle *handle = *((JackHandle **) infoPointer);
if ( handle->ports[0] ) handle->xrun[0] = true;
if ( handle->ports[1] ) handle->xrun[1] = true;
// Look for jack server and try to become a client (only do once per stream).
jack_client_t *client = 0;
if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) {
+ jack_options_t jackoptions = (jack_options_t) ( JackNoStartServer ); //JackNullOption;
+ jack_status_t *status = NULL;
if ( options && !options->streamName.empty() )
- client = jack_client_new( options->streamName.c_str() );
+ client = jack_client_open( options->streamName.c_str(), jackoptions, status );
else
- client = jack_client_new( "RtApiJack" );
+ client = jack_client_open( "RtApiJack", jackoptions, status );
if ( client == 0 ) {
errorText_ = "RtApiJack::probeDeviceOpen: Jack server not found or connection error!";
- error( RtError::WARNING );
+ error( RtAudioError::WARNING );
return FAILURE;
}
}
const char **ports;
std::string port, previousPort, deviceName;
unsigned int nPorts = 0, nDevices = 0;
- ports = jack_get_ports( client, NULL, NULL, 0 );
+ ports = jack_get_ports( client, NULL, JACK_DEFAULT_AUDIO_TYPE, 0 );
if ( ports ) {
// Parse the port names up to the first colon (:).
- unsigned int iColon = 0;
+ size_t iColon = 0;
do {
port = (char *) ports[ nPorts ];
iColon = port.find(":");
return FAILURE;
}
- // Count the available ports containing the client name as device
- // channels. Jack "input ports" equal RtAudio output channels.
- unsigned int nChannels = 0;
- unsigned long flag = JackPortIsOutput;
- if ( mode == INPUT ) flag = JackPortIsInput;
- ports = jack_get_ports( client, deviceName.c_str(), NULL, flag );
- if ( ports ) {
- while ( ports[ nChannels ] ) nChannels++;
- free( ports );
- }
+ unsigned long flag = JackPortIsInput;
+ if ( mode == INPUT ) flag = JackPortIsOutput;
- // 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;
+ 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.
stream_.sampleRate = jackRate;
// Get the latency of the JACK port.
- ports = jack_get_ports( client, deviceName.c_str(), NULL, flag );
- if ( ports[ firstChannel ] )
- stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) );
+ 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.
else {
stream_.mode = mode;
jack_set_process_callback( handle->client, jackCallbackHandler, (void *) &stream_.callbackInfo );
- jack_set_xrun_callback( handle->client, jackXrun, (void *) &handle );
+ jack_set_xrun_callback( handle->client, jackXrun, (void *) &stream_.apiHandle );
jack_on_shutdown( handle->client, jackShutdown, (void *) &stream_.callbackInfo );
}
// here.
if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );
+ if ( options && options->flags & RTAUDIO_JACK_DONT_CONNECT ) shouldAutoconnect_ = false;
+
return SUCCESS;
error:
{
if ( stream_.state == STREAM_CLOSED ) {
errorText_ = "RtApiJack::closeStream(): no open stream to close!";
- error( RtError::WARNING );
+ error( RtAudioError::WARNING );
return;
}
verifyStream();
if ( stream_.state == STREAM_RUNNING ) {
errorText_ = "RtApiJack::startStream(): the stream is already running!";
- error( RtError::WARNING );
+ error( RtAudioError::WARNING );
return;
}
- MUTEX_LOCK(&stream_.mutex);
-
JackHandle *handle = (JackHandle *) stream_.apiHandle;
int result = jack_activate( handle->client );
if ( result ) {
const char **ports;
// Get the list of available ports.
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ if ( shouldAutoconnect_ && (stream_.mode == OUTPUT || stream_.mode == DUPLEX) ) {
result = 1;
- ports = jack_get_ports( handle->client, handle->deviceName[0].c_str(), NULL, JackPortIsInput);
+ 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;
free(ports);
}
- if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+ if ( shouldAutoconnect_ && (stream_.mode == INPUT || stream_.mode == DUPLEX) ) {
result = 1;
- ports = jack_get_ports( handle->client, handle->deviceName[1].c_str(), NULL, JackPortIsOutput );
+ 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;
stream_.state = STREAM_RUNNING;
unlock:
- MUTEX_UNLOCK(&stream_.mutex);
-
if ( result == 0 ) return;
- error( RtError::SYSTEM_ERROR );
+ error( RtAudioError::SYSTEM_ERROR );
}
void RtApiJack :: stopStream( void )
verifyStream();
if ( stream_.state == STREAM_STOPPED ) {
errorText_ = "RtApiJack::stopStream(): the stream is already stopped!";
- error( RtError::WARNING );
+ error( RtAudioError::WARNING );
return;
}
- MUTEX_LOCK( &stream_.mutex );
-
JackHandle *handle = (JackHandle *) stream_.apiHandle;
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
if ( handle->drainCounter == 0 ) {
- handle->drainCounter = 1;
+ handle->drainCounter = 2;
pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
}
}
jack_deactivate( handle->client );
stream_.state = STREAM_STOPPED;
-
- MUTEX_UNLOCK( &stream_.mutex );
}
void RtApiJack :: abortStream( void )
verifyStream();
if ( stream_.state == STREAM_STOPPED ) {
errorText_ = "RtApiJack::abortStream(): the stream is already stopped!";
- error( RtError::WARNING );
+ error( RtAudioError::WARNING );
return;
}
JackHandle *handle = (JackHandle *) stream_.apiHandle;
- handle->drainCounter = 1;
+ 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 ) return SUCCESS;
+ 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( RtError::WARNING );
+ error( RtAudioError::WARNING );
return FAILURE;
}
if ( stream_.bufferSize != nframes ) {
errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!";
- error( RtError::WARNING );
+ error( RtAudioError::WARNING );
return FAILURE;
}
// Check if we were draining the stream and signal is finished.
if ( handle->drainCounter > 3 ) {
- if ( handle->internalDrain == false )
- pthread_cond_signal( &handle->condition );
+ ThreadHandle threadId;
+
+ stream_.state = STREAM_STOPPING;
+ if ( handle->internalDrain == true )
+ pthread_create( &threadId, NULL, jackStopStream, info );
else
- stopStream();
+ pthread_cond_signal( &handle->condition );
return SUCCESS;
}
- MUTEX_LOCK( &stream_.mutex );
-
// Invoke user callback first, to get fresh output data.
if ( handle->drainCounter == 0 ) {
RtAudioCallback callback = (RtAudioCallback) info->callback;
status |= RTAUDIO_INPUT_OVERFLOW;
handle->xrun[1] = false;
}
- handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1],
- stream_.bufferSize, streamTime, status, info->userData );
- if ( handle->drainCounter == 2 ) {
- MUTEX_UNLOCK( &stream_.mutex );
- abortStream();
+ 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 ( handle->drainCounter == 1 )
+ 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 > 0 ) { // write zeros to the output stream
+ 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 );
memcpy( jackbuffer, &stream_.userBuffer[0][i*bufferBytes], bufferBytes );
}
}
+ }
- if ( handle->drainCounter ) {
- handle->drainCounter++;
- goto unlock;
- }
+ // Don't bother draining input
+ if ( handle->drainCounter ) {
+ handle->drainCounter++;
+ goto unlock;
}
if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
}
unlock:
- MUTEX_UNLOCK(&stream_.mutex);
-
RtApi::tickStreamTime();
return SUCCESS;
}
-//******************** End of __UNIX_JACK__ *********************//
+ //******************** End of __UNIX_JACK__ *********************//
#endif
#if defined(__WINDOWS_ASIO__) // ASIO API on Windows
// on information found in
// http://www.cs.wustl.edu/~schmidt/win32-cv-1.html.
-#include "asio/asiosys.h"
-#include "asio/asio.h"
-#include "asio/iasiothiscallresolver.h"
-#include "asio/asiodrivers.h"
+#include "asiosys.h"
+#include "asio.h"
+#include "iasiothiscallresolver.h"
+#include "asiodrivers.h"
#include <cmath>
-AsioDrivers drivers;
-ASIOCallbacks asioCallbacks;
-ASIODriverInfo driverInfo;
-CallbackInfo *asioCallbackInfo;
-bool asioXRun;
+static AsioDrivers drivers;
+static ASIOCallbacks asioCallbacks;
+static ASIODriverInfo driverInfo;
+static CallbackInfo *asioCallbackInfo;
+static bool asioXRun;
struct AsioHandle {
int drainCounter; // Tracks callback counts when draining
// Function declarations (definitions at end of section)
static const char* getAsioErrorString( ASIOError result );
-void sampleRateChanged( ASIOSampleRate sRate );
-long asioMessages( long selector, long value, void* message, double* opt );
+static void sampleRateChanged( ASIOSampleRate sRate );
+static long asioMessages( long selector, long value, void* message, double* opt );
RtApiAsio :: RtApiAsio()
{
HRESULT hr = CoInitialize( NULL );
if ( FAILED(hr) ) {
errorText_ = "RtApiAsio::ASIO requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)";
- error( RtError::WARNING );
+ error( RtAudioError::WARNING );
}
coInitialized_ = true;
unsigned int nDevices = getDeviceCount();
if ( nDevices == 0 ) {
errorText_ = "RtApiAsio::getDeviceInfo: no devices found!";
- error( RtError::INVALID_USE );
+ error( RtAudioError::INVALID_USE );
+ return info;
}
if ( device >= nDevices ) {
errorText_ = "RtApiAsio::getDeviceInfo: device ID is invalid!";
- error( RtError::INVALID_USE );
+ error( RtAudioError::INVALID_USE );
+ return info;
}
- // Don't probe if a stream is already open.
+ // If a stream is already open, we cannot probe other devices. Thus, use the saved results.
if ( stream_.state != STREAM_CLOSED ) {
- errorText_ = "RtApiAsio::getDeviceInfo: unable to probe driver while a stream is open.";
- error( RtError::WARNING );
- return info;
+ 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];
if ( result != ASE_OK ) {
errorStream_ << "RtApiAsio::getDeviceInfo: unable to get driver name (" << getAsioErrorString( result ) << ").";
errorText_ = errorStream_.str();
- error( RtError::WARNING );
+ error( RtAudioError::WARNING );
return info;
}
if ( !drivers.loadDriver( driverName ) ) {
errorStream_ << "RtApiAsio::getDeviceInfo: unable to load driver (" << driverName << ").";
errorText_ = errorStream_.str();
- error( RtError::WARNING );
+ error( RtAudioError::WARNING );
return info;
}
if ( result != ASE_OK ) {
errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
errorText_ = errorStream_.str();
- error( RtError::WARNING );
+ error( RtAudioError::WARNING );
return info;
}
drivers.removeCurrentDriver();
errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
errorText_ = errorStream_.str();
- error( RtError::WARNING );
+ error( RtAudioError::WARNING );
return info;
}
info.sampleRates.clear();
for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {
result = ASIOCanSampleRate( (ASIOSampleRate) SAMPLE_RATES[i] );
- if ( result == ASE_OK )
+ 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.
drivers.removeCurrentDriver();
errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting driver channel info (" << driverName << ").";
errorText_ = errorStream_.str();
- error( RtError::WARNING );
+ error( RtAudioError::WARNING );
return info;
}
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 ( getDefaultOutputDevice() == device )
- info.isDefaultOutput = true;
- if ( getDefaultInputDevice() == device )
- info.isDefaultInput = true;
+ 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;
}
-void bufferSwitch( long index, ASIOBool processNow )
+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 ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] != device ) {
+ if ( isDuplexInput && stream_.device[0] != device ) {
errorText_ = "RtApiAsio::probeDeviceOpen: an ASIO duplex stream must use the same device for input and output!";
return FAILURE;
}
}
// Only load the driver once for duplex stream.
- if ( mode != INPUT || stream_.mode != OUTPUT ) {
+ 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();
}
}
+ // 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 ) {
- drivers.removeCurrentDriver();
errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
errorText_ = errorStream_.str();
- return FAILURE;
+ goto error;
}
if ( ( mode == OUTPUT && (channels+firstChannel) > (unsigned int) outputChannels) ||
( mode == INPUT && (channels+firstChannel) > (unsigned int) inputChannels) ) {
- drivers.removeCurrentDriver();
errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested channel count (" << channels << ") + offset (" << firstChannel << ").";
errorText_ = errorStream_.str();
- return FAILURE;
+ goto error;
}
stream_.nDeviceChannels[mode] = channels;
stream_.nUserChannels[mode] = channels;
// Verify the sample rate is supported.
result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate );
if ( result != ASE_OK ) {
- drivers.removeCurrentDriver();
errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested sample rate (" << sampleRate << ").";
errorText_ = errorStream_.str();
- return FAILURE;
+ goto error;
}
- // Set the sample rate.
- result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate );
+ // Get the current sample rate
+ ASIOSampleRate currentRate;
+ result = ASIOGetSampleRate( ¤tRate );
if ( result != ASE_OK ) {
- drivers.removeCurrentDriver();
- errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ").";
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error getting sample rate.";
errorText_ = errorStream_.str();
- return FAILURE;
+ 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.
else channelInfo.isInput = true;
result = ASIOGetChannelInfo( &channelInfo );
if ( result != ASE_OK ) {
- drivers.removeCurrentDriver();
errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting data format.";
errorText_ = errorStream_.str();
- return FAILURE;
+ goto error;
}
// Assuming WINDOWS host is always little-endian.
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 ) {
- drivers.removeCurrentDriver();
errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") data format not supported by RtAudio.";
errorText_ = errorStream_.str();
- return FAILURE;
+ goto error;
}
// Set the buffer size. For a duplex stream, this will end up
long minSize, maxSize, preferSize, granularity;
result = ASIOGetBufferSize( &minSize, &maxSize, &preferSize, &granularity );
if ( result != ASE_OK ) {
- drivers.removeCurrentDriver();
errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting buffer size.";
errorText_ = errorStream_.str();
- return FAILURE;
+ goto error;
}
- if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;
- else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;
- else if ( granularity == -1 ) {
- // Make sure bufferSize is a power of two.
- double power = std::log10( (double) *bufferSize ) / log10( 2.0 );
- *bufferSize = (int) pow( 2.0, floor(power+0.5) );
- if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;
+ 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 *bufferSize = preferSize;
- }
- else if ( granularity != 0 ) {
- // Set to an even multiple of granularity, rounding up.
- *bufferSize = (*bufferSize + granularity-1) / granularity * granularity;
+ else if ( granularity == -1 ) {
+ // Make sure bufferSize is a power of two.
+ int log2_of_min_size = 0;
+ int log2_of_max_size = 0;
+
+ for ( unsigned int i = 0; i < sizeof(long) * 8; i++ ) {
+ if ( minSize & ((long)1 << i) ) log2_of_min_size = i;
+ if ( maxSize & ((long)1 << i) ) log2_of_max_size = i;
+ }
+
+ long min_delta = std::abs( (long)*bufferSize - ((long)1 << log2_of_min_size) );
+ int min_delta_num = log2_of_min_size;
+
+ for (int i = log2_of_min_size + 1; i <= log2_of_max_size; i++) {
+ long current_delta = std::abs( (long)*bufferSize - ((long)1 << i) );
+ if (current_delta < min_delta) {
+ min_delta = current_delta;
+ min_delta_num = i;
+ }
+ }
+
+ *bufferSize = ( (unsigned int)1 << min_delta_num );
+ if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;
+ else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;
+ }
+ else if ( granularity != 0 ) {
+ // Set to an even multiple of granularity, rounding up.
+ *bufferSize = (*bufferSize + granularity-1) / granularity * granularity;
+ }
}
- if ( mode == INPUT && stream_.mode == OUTPUT && stream_.bufferSize != *bufferSize ) {
- drivers.removeCurrentDriver();
+ /*
+ // 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!";
- return FAILURE;
+ goto error;
}
+ */
stream_.bufferSize = *bufferSize;
stream_.nBuffers = 2;
stream_.deviceInterleaved[mode] = false;
// Allocate, if necessary, our AsioHandle structure for the stream.
- AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
if ( handle == 0 ) {
try {
handle = new AsioHandle;
}
catch ( std::bad_alloc& ) {
- //if ( handle == NULL ) {
- drivers.removeCurrentDriver();
errorText_ = "RtApiAsio::probeDeviceOpen: error allocating AsioHandle memory.";
- return FAILURE;
+ goto error;
}
handle->bufferInfos = 0;
// Create the ASIO internal buffers. Since RtAudio sets up input
// and output separately, we'll have to dispose of previously
// created output buffers for a duplex stream.
- long inputLatency, outputLatency;
if ( mode == INPUT && stream_.mode == OUTPUT ) {
ASIODisposeBuffers();
if ( handle->bufferInfos ) free( handle->bufferInfos );
}
// Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure.
- bool buffersAllocated = false;
- unsigned int i, nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
+ 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 << ").";
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;
+ buffersAllocated = true;
+ stream_.state = STREAM_STOPPED;
// Set flags for buffer conversion.
stream_.doConvertBuffer[mode] = false;
stream_.doConvertBuffer[mode] = true;
// Allocate necessary internal buffers
- unsigned long bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+ 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.";
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 ( isDuplexInput && stream_.deviceBuffer ) {
+ unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
+ if ( bufferBytes <= bytesOut ) makeBuffer = false;
}
if ( makeBuffer ) {
}
}
- stream_.sampleRate = sampleRate;
- stream_.device[mode] = device;
- stream_.state = STREAM_STOPPED;
- asioCallbackInfo = &stream_.callbackInfo;
- stream_.callbackInfo.object = (void *) this;
- if ( stream_.mode == OUTPUT && mode == INPUT )
- // We had already set up an output stream.
- stream_.mode = DUPLEX;
- else
- stream_.mode = mode;
-
// Determine device latencies
+ 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( RtError::WARNING); // warn but don't fail
+ error( RtAudioError::WARNING); // warn but don't fail
}
else {
stream_.latency[0] = outputLatency;
return SUCCESS;
error:
- if ( buffersAllocated )
- ASIODisposeBuffers();
- drivers.removeCurrentDriver();
+ if ( !isDuplexInput ) {
+ // the cleanup for error in the duplex input, is done by RtApi::openStream
+ // So we clean up for single channel only
- if ( handle ) {
- CloseHandle( handle->condition );
- if ( handle->bufferInfos )
- free( handle->bufferInfos );
- delete handle;
- stream_.apiHandle = 0;
- }
+ if ( buffersAllocated )
+ ASIODisposeBuffers();
- for ( int i=0; i<2; i++ ) {
- if ( stream_.userBuffer[i] ) {
- free( stream_.userBuffer[i] );
- stream_.userBuffer[i] = 0;
+ drivers.removeCurrentDriver();
+
+ if ( handle ) {
+ CloseHandle( handle->condition );
+ if ( handle->bufferInfos )
+ free( handle->bufferInfos );
+
+ delete handle;
+ stream_.apiHandle = 0;
}
- }
- if ( stream_.deviceBuffer ) {
- free( stream_.deviceBuffer );
- stream_.deviceBuffer = 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( RtError::WARNING );
+ error( RtAudioError::WARNING );
return;
}
stream_.state = STREAM_CLOSED;
}
+bool stopThreadCalled = false;
+
void RtApiAsio :: startStream()
{
verifyStream();
if ( stream_.state == STREAM_RUNNING ) {
errorText_ = "RtApiAsio::startStream(): the stream is already running!";
- error( RtError::WARNING );
+ error( RtAudioError::WARNING );
return;
}
- MUTEX_LOCK( &stream_.mutex );
-
AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
ASIOError result = ASIOStart();
if ( result != ASE_OK ) {
handle->drainCounter = 0;
handle->internalDrain = false;
+ ResetEvent( handle->condition );
stream_.state = STREAM_RUNNING;
asioXRun = false;
unlock:
- MUTEX_UNLOCK( &stream_.mutex );
+ stopThreadCalled = false;
if ( result == ASE_OK ) return;
- error( RtError::SYSTEM_ERROR );
+ error( RtAudioError::SYSTEM_ERROR );
}
void RtApiAsio :: stopStream()
verifyStream();
if ( stream_.state == STREAM_STOPPED ) {
errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!";
- error( RtError::WARNING );
+ error( RtAudioError::WARNING );
return;
}
- MUTEX_LOCK( &stream_.mutex );
-
AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
if ( handle->drainCounter == 0 ) {
- handle->drainCounter = 1;
- MUTEX_UNLOCK( &stream_.mutex );
- WaitForMultipleObjects( 1, &handle->condition, FALSE, INFINITE ); // block until signaled
- ResetEvent( handle->condition );
- MUTEX_LOCK( &stream_.mutex );
+ 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();
}
- stream_.state = STREAM_STOPPED;
- MUTEX_UNLOCK( &stream_.mutex );
-
if ( result == ASE_OK ) return;
- error( RtError::SYSTEM_ERROR );
+ error( RtAudioError::SYSTEM_ERROR );
}
void RtApiAsio :: abortStream()
verifyStream();
if ( stream_.state == STREAM_STOPPED ) {
errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!";
- error( RtError::WARNING );
+ 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 completed
+ // 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 = 1;
+ // 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 ) return SUCCESS;
+ 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( RtError::WARNING );
+ error( RtAudioError::WARNING );
return FAILURE;
}
CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
- // Check if we were draining the stream and signal is finished.
+ // 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
- stopStream();
+ else { // spawn a thread to stop the stream
+ unsigned threadId;
+ stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream,
+ &stream_.callbackInfo, 0, &threadId );
+ }
return SUCCESS;
}
- MUTEX_LOCK( &stream_.mutex );
-
- // The state might change while waiting on a mutex.
- if ( stream_.state == STREAM_STOPPED ) goto unlock;
-
// Invoke user callback to get fresh output data UNLESS we are
// draining stream.
if ( handle->drainCounter == 0 ) {
status |= RTAUDIO_INPUT_OVERFLOW;
asioXRun = false;
}
- handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+ int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
stream_.bufferSize, streamTime, status, info->userData );
- if ( handle->drainCounter == 2 ) {
- MUTEX_UNLOCK( &stream_.mutex );
- abortStream();
+ 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 ( handle->drainCounter == 1 )
+ else if ( cbReturnValue == 1 ) {
+ handle->drainCounter = 1;
handle->internalDrain = true;
+ }
}
- unsigned int bufferBytes, i, j;
- unsigned int nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
+ 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 ) {
- handle->drainCounter++;
- goto unlock;
- }
+ // Don't bother draining input
+ if ( handle->drainCounter ) {
+ handle->drainCounter++;
+ goto unlock;
}
if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
// drivers apparently do not function correctly without it.
ASIOOutputReady();
- MUTEX_UNLOCK( &stream_.mutex );
-
RtApi::tickStreamTime();
return SUCCESS;
}
-void sampleRateChanged( ASIOSampleRate sRate )
+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,
try {
object->stopStream();
}
- catch ( RtError &exception ) {
+ 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;
}
-long asioMessages( long selector, long value, void* message, double* opt )
+static long asioMessages( long selector, long value, void* /*message*/, double* /*opt*/ )
{
long ret = 0;
const char*message;
};
- static Messages m[] =
- {
- { ASE_NotPresent, "Hardware input or output is not present or available." },
- { ASE_HWMalfunction, "Hardware is malfunctioning." },
- { ASE_InvalidParameter, "Invalid input parameter." },
- { ASE_InvalidMode, "Invalid mode." },
- { ASE_SPNotAdvancing, "Sample position not advancing." },
- { ASE_NoClock, "Sample clock or rate cannot be determined or is not present." },
- { ASE_NoMemory, "Not enough memory to complete the request." }
- };
+ 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_DS__) // Windows DirectSound API
+#if defined(__WINDOWS_WASAPI__) // Windows WASAPI API
-// Modified by Robin Davies, October 2005
-// - Improvements to DirectX pointer chasing.
-// - Backdoor RtDsStatistics hook provides DirectX performance information.
-// - Bug fix for non-power-of-two Asio granularity used by Edirol PCR-A30.
-// - Auto-call CoInitialize for DSOUND and ASIO platforms.
-// Various revisions for RtAudio 4.0 by Gary Scavone, April 2007
+// 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
-#include <dsound.h>
-#include <assert.h>
+#ifndef INITGUID
+ #define INITGUID
+#endif
+#include <audioclient.h>
+#include <avrt.h>
+#include <mmdeviceapi.h>
+#include <functiondiscoverykeys_devpkey.h>
+#include <sstream>
+
+//=============================================================================
+
+#define SAFE_RELEASE( objectPtr )\
+if ( objectPtr )\
+{\
+ objectPtr->Release();\
+ objectPtr = NULL;\
+}
-#define MINIMUM_DEVICE_BUFFER_SIZE 32768
+typedef HANDLE ( __stdcall *TAvSetMmThreadCharacteristicsPtr )( LPCWSTR TaskName, LPDWORD TaskIndex );
-#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 dsPointerDifference( DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize )
+// 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
{
- if (laterPointer > earlierPointer)
- return laterPointer - earlierPointer;
- else
- return laterPointer - earlierPointer + bufferSize;
-}
+public:
+ WasapiBuffer()
+ : buffer_( NULL ),
+ bufferSize_( 0 ),
+ inIndex_( 0 ),
+ outIndex_( 0 ) {}
-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;
-}
+ ~WasapiBuffer() {
+ free( buffer_ );
+ }
-// 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;
+ // sets the length of the internal ring buffer
+ void setBufferSize( unsigned int bufferSize, unsigned int formatBytes ) {
+ free( buffer_ );
- DsHandle()
- :drainCounter(0), internalDrain(false) { id[0] = 0, id[1] = 0; xrun[0] = false; xrun[1] = false; bufferPointer[0] = 0; bufferPointer[1] = 0; }
-};
+ buffer_ = ( char* ) calloc( bufferSize, formatBytes );
-/*
-RtApiDs::RtDsStatistics RtApiDs::statistics;
+ bufferSize_ = bufferSize;
+ inIndex_ = 0;
+ outIndex_ = 0;
+ }
-// Provides a backdoor hook to monitor for DirectSound read overruns and write underruns.
-RtApiDs::RtDsStatistics RtApiDs::getDsStatistics()
-{
- RtDsStatistics s = statistics;
+ // 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;
+ }
- // update the calculated fields.
- if ( s.inputFrameSize != 0 )
- s.latency += s.readDeviceSafeLeadBytes * 1.0 / s.inputFrameSize / s.sampleRate;
+ unsigned int relOutIndex = outIndex_;
+ unsigned int inIndexEnd = inIndex_ + bufferSize;
+ if ( relOutIndex < inIndex_ && inIndexEnd >= bufferSize_ ) {
+ relOutIndex += bufferSize_;
+ }
- if ( s.outputFrameSize != 0 )
- s.latency += (s.writeDeviceSafeLeadBytes + s.writeDeviceBufferLeadBytes) * 1.0 / s.outputFrameSize / s.sampleRate;
+ // "in" index can end on the "out" index but cannot begin at it
+ if ( inIndex_ <= relOutIndex && inIndexEnd > relOutIndex ) {
+ return false; // not enough space between "in" index and "out" index
+ }
- return s;
-}
-*/
+ // copy buffer from external to internal
+ int fromZeroSize = inIndex_ + bufferSize - bufferSize_;
+ fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize;
+ int fromInSize = bufferSize - fromZeroSize;
-// Declarations for utility functions, callbacks, and structures
-// specific to the DirectSound implementation.
-static bool CALLBACK deviceCountCallback( LPGUID lpguid,
- LPCTSTR description,
- LPCTSTR module,
- LPVOID lpContext );
+ 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;
+ }
-static char* getErrorString( int code );
+ // update "in" index
+ inIndex_ += bufferSize;
+ inIndex_ %= bufferSize_;
-extern "C" unsigned __stdcall callbackHandler( void *ptr );
+ return true;
+ }
-struct EnumInfo {
- bool isInput;
- bool getDefault;
- bool findIndex;
- unsigned int counter;
- unsigned int index;
- LPGUID id;
- std::string name;
+ // attempt to pull a buffer from the ring buffer from the current "out" index
+ bool pullBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format )
+ {
+ if ( !buffer || // incoming buffer is NULL
+ bufferSize == 0 || // incoming buffer has no data
+ bufferSize > bufferSize_ ) // incoming buffer too large
+ {
+ return false;
+ }
+
+ unsigned int relInIndex = inIndex_;
+ unsigned int outIndexEnd = outIndex_ + bufferSize;
+ if ( relInIndex < outIndex_ && outIndexEnd >= bufferSize_ ) {
+ relInIndex += bufferSize_;
+ }
+
+ // "out" index can begin at and end on the "in" index
+ if ( outIndex_ < relInIndex && outIndexEnd > relInIndex ) {
+ return false; // not enough space between "out" index and "in" index
+ }
+
+ // copy buffer from internal to external
+ int fromZeroSize = outIndex_ + bufferSize - bufferSize_;
+ fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize;
+ int fromOutSize = bufferSize - fromZeroSize;
+
+ switch( format )
+ {
+ case RTAUDIO_SINT8:
+ memcpy( buffer, &( ( char* ) buffer_ )[outIndex_], fromOutSize * sizeof( char ) );
+ memcpy( &( ( char* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( char ) );
+ break;
+ case RTAUDIO_SINT16:
+ memcpy( buffer, &( ( short* ) buffer_ )[outIndex_], fromOutSize * sizeof( short ) );
+ memcpy( &( ( short* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( short ) );
+ break;
+ case RTAUDIO_SINT24:
+ memcpy( buffer, &( ( S24* ) buffer_ )[outIndex_], fromOutSize * sizeof( S24 ) );
+ memcpy( &( ( S24* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( S24 ) );
+ break;
+ case RTAUDIO_SINT32:
+ memcpy( buffer, &( ( int* ) buffer_ )[outIndex_], fromOutSize * sizeof( int ) );
+ memcpy( &( ( int* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( int ) );
+ break;
+ case RTAUDIO_FLOAT32:
+ memcpy( buffer, &( ( float* ) buffer_ )[outIndex_], fromOutSize * sizeof( float ) );
+ memcpy( &( ( float* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( float ) );
+ break;
+ case RTAUDIO_FLOAT64:
+ memcpy( buffer, &( ( double* ) buffer_ )[outIndex_], fromOutSize * sizeof( double ) );
+ memcpy( &( ( double* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( double ) );
+ break;
+ }
+
+ // update "out" index
+ outIndex_ += bufferSize;
+ outIndex_ %= bufferSize_;
- EnumInfo()
- : isInput(false), getDefault(false), findIndex(false), counter(0), index(0) {}
+ return true;
+ }
+
+private:
+ char* buffer_;
+ unsigned int bufferSize_;
+ unsigned int inIndex_;
+ unsigned int outIndex_;
};
-RtApiDs :: RtApiDs()
+//-----------------------------------------------------------------------------
+
+// A structure to hold various information related to the WASAPI implementation.
+struct WasapiHandle
{
- // 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;
-}
+ 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 ) {}
+};
-RtApiDs :: ~RtApiDs()
+//=============================================================================
+
+RtApiWasapi::RtApiWasapi()
+ : coInitialized_( false ), deviceEnumerator_( NULL )
{
- if ( coInitialized_ ) CoUninitialize(); // balanced call.
- if ( stream_.state != STREAM_CLOSED ) closeStream();
+ // WASAPI can run either apartment or multi-threaded
+ HRESULT hr = CoInitialize( NULL );
+ if ( !FAILED( hr ) )
+ coInitialized_ = true;
+
+ // Instantiate device enumerator
+ hr = CoCreateInstance( __uuidof( MMDeviceEnumerator ), NULL,
+ CLSCTX_ALL, __uuidof( IMMDeviceEnumerator ),
+ ( void** ) &deviceEnumerator_ );
+
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::RtApiWasapi: Unable to instantiate device enumerator";
+ error( RtAudioError::DRIVER_ERROR );
+ }
}
-unsigned int RtApiDs :: getDefaultInputDevice( void )
+//-----------------------------------------------------------------------------
+
+RtApiWasapi::~RtApiWasapi()
{
- // Count output devices.
- EnumInfo info;
- HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceCountCallback, &info );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::getDefaultOutputDevice: error (" << getErrorString( result ) << ") counting output devices!";
- errorText_ = errorStream_.str();
- error( RtError::WARNING );
- return 0;
- }
+ if ( stream_.state != STREAM_CLOSED )
+ closeStream();
- // Now enumerate input devices until we find the id = NULL.
- info.isInput = true;
- info.getDefault = true;
- result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceCountCallback, &info );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::getDefaultInputDevice: error (" << getErrorString( result ) << ") enumerating input devices!";
- errorText_ = errorStream_.str();
- error( RtError::WARNING );
- return 0;
- }
+ SAFE_RELEASE( deviceEnumerator_ );
- if ( info.counter > 0 ) return info.counter - 1;
- return 0;
+ // If this object previously called CoInitialize()
+ if ( coInitialized_ )
+ CoUninitialize();
}
-unsigned int RtApiDs :: getDefaultOutputDevice( void )
+//=============================================================================
+
+unsigned int RtApiWasapi::getDeviceCount( void )
{
- // Enumerate output devices until we find the id = NULL.
- EnumInfo info;
- info.getDefault = true;
- HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceCountCallback, &info );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::getDefaultOutputDevice: error (" << getErrorString( result ) << ") enumerating output devices!";
- errorText_ = errorStream_.str();
- error( RtError::WARNING );
- return 0;
+ unsigned int captureDeviceCount = 0;
+ unsigned int renderDeviceCount = 0;
+
+ IMMDeviceCollection* captureDevices = NULL;
+ IMMDeviceCollection* renderDevices = NULL;
+
+ // Count capture devices
+ errorText_.clear();
+ HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device collection.";
+ goto Exit;
}
- if ( info.counter > 0 ) return info.counter - 1;
- return 0;
-}
+ hr = captureDevices->GetCount( &captureDeviceCount );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device count.";
+ goto Exit;
+ }
-unsigned int RtApiDs :: getDeviceCount( void )
-{
- // Count DirectSound devices.
- EnumInfo info;
- HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceCountCallback, &info );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!";
- errorText_ = errorStream_.str();
- error( RtError::WARNING );
+ // 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;
}
- // Count DirectSoundCapture devices.
- info.isInput = true;
- result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceCountCallback, &info );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!";
- errorText_ = errorStream_.str();
- error( RtError::WARNING );
+ hr = renderDevices->GetCount( &renderDeviceCount );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device count.";
+ goto Exit;
}
- return info.counter;
+Exit:
+ // release all references
+ SAFE_RELEASE( captureDevices );
+ SAFE_RELEASE( renderDevices );
+
+ if ( errorText_.empty() )
+ return captureDeviceCount + renderDeviceCount;
+
+ error( RtAudioError::DRIVER_ERROR );
+ return 0;
}
-RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device )
-{
- // Because DirectSound always enumerates input and output devices
- // separately (and because we don't attempt to combine devices
- // internally), none of our "devices" will ever be duplex.
+//-----------------------------------------------------------------------------
+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;
- // Enumerate through devices to find the id (if it exists). Note
- // that we have to do the output enumeration first, even if this is
- // an input device, in order for the device counter to be correct.
- EnumInfo dsinfo;
- dsinfo.findIndex = true;
- dsinfo.index = device;
- HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceCountCallback, &dsinfo );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") enumerating output devices!";
- errorText_ = errorStream_.str();
- error( RtError::WARNING );
+ // 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;
}
- if ( dsinfo.name.empty() ) goto probeInput;
+ hr = captureDevices->GetCount( &captureDeviceCount );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device count.";
+ goto Exit;
+ }
- LPDIRECTSOUND output;
- DSCAPS outCaps;
- result = DirectSoundCreate( dsinfo.id, &output, NULL );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening output device (" << dsinfo.name << ")!";
- errorText_ = errorStream_.str();
- error( RtError::WARNING );
- return info;
+ // 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;
}
- outCaps.dwSize = sizeof( outCaps );
- result = output->GetCaps( &outCaps );
- if ( FAILED( result ) ) {
- output->Release();
- errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting capabilities!";
- errorText_ = errorStream_.str();
- error( RtError::WARNING );
- return info;
+ hr = renderDevices->GetCount( &renderDeviceCount );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device count.";
+ goto Exit;
}
- // Get output channel information.
- info.outputChannels = ( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1;
+ // validate device index
+ if ( device >= captureDeviceCount + renderDeviceCount ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Invalid device index.";
+ errorType = RtAudioError::INVALID_USE;
+ goto Exit;
+ }
- // 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] );
+ // 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 format information.
- if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT ) info.nativeFormats |= RTAUDIO_SINT16;
- if ( outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) info.nativeFormats |= RTAUDIO_SINT8;
+ // 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;
+ }
+ }
- output->Release();
+ hr = defaultDevicePtr->OpenPropertyStore( STGM_READ, &defaultDevicePropStore );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open default device property store.";
+ goto Exit;
+ }
+ PropVariantInit( &defaultDeviceNameProp );
- if ( getDefaultOutputDevice() == device )
- info.isDefaultOutput = true;
+ hr = defaultDevicePropStore->GetValue( PKEY_Device_FriendlyName, &defaultDeviceNameProp );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default device property: PKEY_Device_FriendlyName.";
+ goto Exit;
+ }
- // Copy name and return.
- info.name = dsinfo.name;
+ defaultDeviceName = convertCharPointerToStdString(defaultDeviceNameProp.pwszVal);
- info.probed = true;
- return info;
+ // name
+ hr = devicePtr->OpenPropertyStore( STGM_READ, &devicePropStore );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open device property store.";
+ goto Exit;
+ }
- probeInput:
+ PropVariantInit( &deviceNameProp );
- dsinfo.isInput = true;
- result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceCountCallback, &dsinfo );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") enumerating input devices!";
- errorText_ = errorStream_.str();
- error( RtError::WARNING );
+ hr = devicePropStore->GetValue( PKEY_Device_FriendlyName, &deviceNameProp );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device property: PKEY_Device_FriendlyName.";
+ goto Exit;
}
- if ( dsinfo.name.empty() ) return info;
+ info.name =convertCharPointerToStdString(deviceNameProp.pwszVal);
- LPDIRECTSOUNDCAPTURE input;
- result = DirectSoundCaptureCreate( dsinfo.id, &input, NULL );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsinfo.name << ")!";
- errorText_ = errorStream_.str();
- error( RtError::WARNING );
- return info;
+ // is default
+ if ( isCaptureDevice ) {
+ info.isDefaultInput = info.name == defaultDeviceName;
+ info.isDefaultOutput = false;
+ }
+ else {
+ info.isDefaultInput = false;
+ info.isDefaultOutput = info.name == defaultDeviceName;
}
- DSCCAPS inCaps;
- inCaps.dwSize = sizeof( inCaps );
- result = input->GetCaps( &inCaps );
- if ( FAILED( result ) ) {
- input->Release();
- errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsinfo.name << ")!";
- errorText_ = errorStream_.str();
- error( RtError::WARNING );
- return info;
+ // 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;
}
- // Get input channel information.
- info.inputChannels = inCaps.dwChannels;
+ hr = audioClient->GetMixFormat( &deviceFormat );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device mix format.";
+ goto Exit;
+ }
- // Get sample rate and format information.
- 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 ( isCaptureDevice ) {
+ info.inputChannels = deviceFormat->nChannels;
+ info.outputChannels = 0;
+ info.duplexChannels = 0;
+ }
+ else {
+ info.inputChannels = 0;
+ info.outputChannels = deviceFormat->nChannels;
+ info.duplexChannels = 0;
+ }
- if ( info.nativeFormats & RTAUDIO_SINT16 ) {
- if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.sampleRates.push_back( 11025 );
- if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.sampleRates.push_back( 22050 );
- if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.sampleRates.push_back( 44100 );
- if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.sampleRates.push_back( 96000 );
+ // sample rates (WASAPI only supports the one native sample rate)
+ info.preferredSampleRate = deviceFormat->nSamplesPerSec;
+
+ info.sampleRates.clear();
+ info.sampleRates.push_back( 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 ( info.nativeFormats & RTAUDIO_SINT8 ) {
- if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.sampleRates.push_back( 11025 );
- if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.sampleRates.push_back( 22050 );
- if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.sampleRates.push_back( 44100 );
- if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.sampleRates.push_back( 44100 );
+ else if ( deviceFormat->wBitsPerSample == 64 ) {
+ info.nativeFormats |= RTAUDIO_FLOAT64;
}
}
- else if ( inCaps.dwChannels == 1 ) {
- if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.nativeFormats |= RTAUDIO_SINT16;
- if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.nativeFormats |= RTAUDIO_SINT16;
- if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.nativeFormats |= RTAUDIO_SINT16;
- if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.nativeFormats |= RTAUDIO_SINT16;
- if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.nativeFormats |= RTAUDIO_SINT8;
- if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.nativeFormats |= RTAUDIO_SINT8;
- if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.nativeFormats |= RTAUDIO_SINT8;
- if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.nativeFormats |= RTAUDIO_SINT8;
-
- if ( info.nativeFormats & RTAUDIO_SINT16 ) {
- if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.sampleRates.push_back( 11025 );
- if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.sampleRates.push_back( 22050 );
- if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.sampleRates.push_back( 44100 );
- if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.sampleRates.push_back( 96000 );
+ else if ( 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 ( info.nativeFormats & RTAUDIO_SINT8 ) {
- if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.sampleRates.push_back( 11025 );
- if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.sampleRates.push_back( 22050 );
- if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.sampleRates.push_back( 44100 );
- if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.sampleRates.push_back( 96000 );
+ else 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;
}
}
- else info.inputChannels = 0; // technically, this would be an error
- input->Release();
+ // probed
+ info.probed = true;
- if ( info.inputChannels == 0 ) return info;
+Exit:
+ // release all references
+ PropVariantClear( &deviceNameProp );
+ PropVariantClear( &defaultDeviceNameProp );
- if ( getDefaultInputDevice() == device )
- info.isDefaultInput = true;
+ SAFE_RELEASE( captureDevices );
+ SAFE_RELEASE( renderDevices );
+ SAFE_RELEASE( devicePtr );
+ SAFE_RELEASE( defaultDevicePtr );
+ SAFE_RELEASE( audioClient );
+ SAFE_RELEASE( devicePropStore );
+ SAFE_RELEASE( defaultDevicePropStore );
- // Copy name and return.
- info.name = dsinfo.name;
- info.probed = true;
+ CoTaskMemFree( deviceFormat );
+ CoTaskMemFree( closestMatchFormat );
+
+ if ( !errorText_.empty() )
+ error( errorType );
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 )
+//-----------------------------------------------------------------------------
+
+unsigned int RtApiWasapi::getDefaultOutputDevice( void )
{
- if ( channels + firstChannel > 2 ) {
- errorText_ = "RtApiDs::probeDeviceOpen: DirectSound does not support more than 2 channels per device.";
- return FAILURE;
+ for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {
+ if ( getDeviceInfo( i ).isDefaultOutput ) {
+ return i;
+ }
}
- // Enumerate through devices to find the id (if it exists). Note
- // that we have to do the output enumeration first, even if this is
- // an input device, in order for the device counter to be correct.
- EnumInfo dsinfo;
- dsinfo.findIndex = true;
- dsinfo.index = device;
- HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceCountCallback, &dsinfo );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") enumerating output devices!";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
+ return 0;
+}
- if ( mode == OUTPUT ) {
- if ( dsinfo.name.empty() ) {
- errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support output!";
- errorText_ = errorStream_.str();
- return FAILURE;
+//-----------------------------------------------------------------------------
+
+unsigned int RtApiWasapi::getDefaultInputDevice( void )
+{
+ for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {
+ if ( getDeviceInfo( i ).isDefaultInput ) {
+ return i;
}
}
- else { // mode == INPUT
- dsinfo.isInput = true;
- HRESULT result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceCountCallback, &dsinfo );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") enumerating input devices!";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
- if ( dsinfo.name.empty() ) {
- errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support input!";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
+
+ return 0;
+}
+
+//-----------------------------------------------------------------------------
+
+void RtApiWasapi::closeStream( void )
+{
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiWasapi::closeStream: No open stream to close.";
+ error( RtAudioError::WARNING );
+ return;
}
- // According to a note in PortAudio, using GetDesktopWindow()
- // instead of GetForegroundWindow() is supposed to avoid problems
- // that occur when the application's window is not the foreground
- // window. Also, if the application window closes before the
- // DirectSound buffer, DirectSound can crash. However, for console
- // applications, no sound was produced when using GetDesktopWindow().
- HWND hWnd = GetForegroundWindow();
+ if ( stream_.state != STREAM_STOPPED )
+ stopStream();
- // 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;
+ // clean up stream memory
+ SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient )
+ SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient )
- // 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;
+ SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureClient )
+ SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderClient )
- // Determine the device buffer size. By default, 32k, but we will
- // grow it to make allowances for very large software buffer sizes.
- DWORD dsBufferSize = 0;
- DWORD dsPointerLeadTime = 0;
- long bufferBytes = MINIMUM_DEVICE_BUFFER_SIZE; // sound cards will always *knock wood* support this
+ if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent )
+ CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent );
- void *ohandle = 0, *bhandle = 0;
- if ( mode == OUTPUT ) {
+ if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent )
+ CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent );
- LPDIRECTSOUND output;
- result = DirectSoundCreate( dsinfo.id, &output, NULL );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsinfo.name << ")!";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
+ delete ( WasapiHandle* ) stream_.apiHandle;
+ stream_.apiHandle = NULL;
- DSCAPS outCaps;
- outCaps.dwSize = sizeof( outCaps );
- result = output->GetCaps( &outCaps );
- if ( FAILED( result ) ) {
- output->Release();
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting capabilities (" << dsinfo.name << ")!";
- errorText_ = errorStream_.str();
- return FAILURE;
+ for ( int i = 0; i < 2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
}
+ }
- // Check channel information.
- if ( channels + firstChannel == 2 && !( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ) {
- errorStream_ << "RtApiDs::getDeviceInfo: the output device (" << dsinfo.name << ") does not support stereo playback.";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
- // 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 stream state
+ stream_.state = STREAM_CLOSED;
+}
- // Update wave format structure and buffer information.
- waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
- waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
- dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
+//-----------------------------------------------------------------------------
- // If the user wants an even bigger buffer, increase the device buffer size accordingly.
- while ( dsPointerLeadTime * 2U > (DWORD) bufferBytes )
- bufferBytes *= 2;
+void RtApiWasapi::startStream( void )
+{
+ verifyStream();
- // Set cooperative level to DSSCL_EXCLUSIVE
- result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE );
- if ( FAILED( result ) ) {
- output->Release();
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting cooperative level (" << dsinfo.name << ")!";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
+ if ( stream_.state == STREAM_RUNNING ) {
+ errorText_ = "RtApiWasapi::startStream: The stream is already running.";
+ error( RtAudioError::WARNING );
+ return;
+ }
- // 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;
+ // update stream state
+ stream_.state = STREAM_RUNNING;
- // 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 (" << dsinfo.name << ")!";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
+ // create WASAPI stream thread
+ stream_.callbackInfo.thread = ( ThreadHandle ) CreateThread( NULL, 0, runWasapiThread, this, CREATE_SUSPENDED, NULL );
- // 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 (" << dsinfo.name << ")!";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
+ 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 );
+ }
+}
- // Setup the secondary DS buffer description.
- dsBufferSize = (DWORD) bufferBytes;
- ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
- bufferDescription.dwSize = sizeof( DSBUFFERDESC );
- bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
- DSBCAPS_GETCURRENTPOSITION2 |
- DSBCAPS_LOCHARDWARE ); // Force hardware mixing
- bufferDescription.dwBufferBytes = bufferBytes;
- bufferDescription.lpwfxFormat = &waveFormat;
+//-----------------------------------------------------------------------------
- // Try to create the secondary DS buffer. If that doesn't work,
- // try to use software mixing. Otherwise, there's a problem.
- result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
- if ( FAILED( result ) ) {
- bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
- 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 (" << dsinfo.name << ")!";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
+void RtApiWasapi::stopStream( void )
+{
+ verifyStream();
+
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiWasapi::stopStream: The stream is already stopped.";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ // inform stream thread by setting stream state to STREAM_STOPPING
+ stream_.state = STREAM_STOPPING;
+
+ // wait until stream thread is stopped
+ while( stream_.state != STREAM_STOPPED ) {
+ Sleep( 1 );
+ }
+
+ // Wait for the last buffer to play before stopping.
+ Sleep( 1000 * stream_.bufferSize / stream_.sampleRate );
+
+ // stop capture client if applicable
+ if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) {
+ HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop();
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::stopStream: Unable to stop capture stream.";
+ error( RtAudioError::DRIVER_ERROR );
+ return;
}
+ }
- // 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 (" << dsinfo.name << ")!";
- errorText_ = errorStream_.str();
- return FAILURE;
+ // stop render client if applicable
+ if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) {
+ HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop();
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::stopStream: Unable to stop render stream.";
+ error( RtAudioError::DRIVER_ERROR );
+ return;
}
+ }
- bufferBytes = dsbcaps.dwBufferBytes;
+ // 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;
+ }
- // Lock the DS buffer
- LPVOID audioPtr;
- DWORD dataLen;
- result = buffer->Lock( 0, bufferBytes, &audioPtr, &dataLen, NULL, NULL, 0 );
- if ( FAILED( result ) ) {
- output->Release();
- buffer->Release();
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking buffer (" << dsinfo.name << ")!";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
+ stream_.callbackInfo.thread = (ThreadHandle) NULL;
+}
- // 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 (" << dsinfo.name << ")!";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
+void RtApiWasapi::abortStream( void )
+{
+ verifyStream();
- dsBufferSize = bufferBytes;
- ohandle = (void *) output;
- bhandle = (void *) buffer;
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiWasapi::abortStream: The stream is already stopped.";
+ error( RtAudioError::WARNING );
+ return;
}
- if ( mode == INPUT ) {
+ // inform stream thread by setting stream state to STREAM_STOPPING
+ stream_.state = STREAM_STOPPING;
- LPDIRECTSOUNDCAPTURE input;
- result = DirectSoundCaptureCreate( dsinfo.id, &input, NULL );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsinfo.name << ")!";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
+ // wait until stream thread is stopped
+ while ( stream_.state != STREAM_STOPPED ) {
+ Sleep( 1 );
+ }
- DSCCAPS inCaps;
- inCaps.dwSize = sizeof( inCaps );
- result = input->GetCaps( &inCaps );
- if ( FAILED( result ) ) {
- input->Release();
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsinfo.name << ")!";
- errorText_ = errorStream_.str();
- return FAILURE;
+ // stop capture client if applicable
+ if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) {
+ HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop();
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::abortStream: Unable to stop capture stream.";
+ error( RtAudioError::DRIVER_ERROR );
+ return;
}
+ }
- // Check channel information.
- if ( inCaps.dwChannels < channels + firstChannel ) {
- errorText_ = "RtApiDs::getDeviceInfo: the input device does not support requested input channels.";
- return FAILURE;
+ // stop render client if applicable
+ if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) {
+ HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop();
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::abortStream: Unable to stop render stream.";
+ error( RtAudioError::DRIVER_ERROR );
+ return;
}
+ }
- // 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;
- }
+ // 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;
+ RtAudio::DeviceInfo deviceInfo;
+
+ // 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;
+ }
+
+ deviceInfo = getDeviceInfo( device );
+
+ // validate sample rate
+ if ( sampleRate != deviceInfo.preferredSampleRate )
+ {
+ errorType = RtAudioError::INVALID_USE;
+ std::stringstream ss;
+ ss << "RtApiWasapi::probeDeviceOpen: " << sampleRate
+ << "Hz sample rate not supported. This device only supports "
+ << deviceInfo.preferredSampleRate << "Hz.";
+ errorText_ = ss.str();
+ goto Exit;
+ }
+
+ // determine whether index falls within capture or render devices
+ if ( device >= renderDeviceCount ) {
+ if ( mode != INPUT ) {
+ errorType = RtAudioError::INVALID_USE;
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Capture device selected as output device.";
+ goto Exit;
}
- 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;
- }
+
+ // 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;
}
- stream_.userFormat = format;
- // Update wave format structure and buffer information.
- waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
- waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
+ hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
+ NULL, ( void** ) &captureAudioClient );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client.";
+ goto Exit;
+ }
- // Setup the secondary DS buffer description.
- dsBufferSize = bufferBytes;
- DSCBUFFERDESC bufferDescription;
- ZeroMemory( &bufferDescription, sizeof( DSCBUFFERDESC ) );
- bufferDescription.dwSize = sizeof( DSCBUFFERDESC );
- bufferDescription.dwFlags = 0;
- bufferDescription.dwReserved = 0;
- bufferDescription.dwBufferBytes = bufferBytes;
- bufferDescription.lpwfxFormat = &waveFormat;
+ hr = captureAudioClient->GetMixFormat( &deviceFormat );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format.";
+ goto Exit;
+ }
- // 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 (" << dsinfo.name << ")!";
- errorText_ = errorStream_.str();
- return FAILURE;
+ stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
+ captureAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
+ }
+ else {
+ if ( mode != OUTPUT ) {
+ errorType = RtAudioError::INVALID_USE;
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Render device selected as input device.";
+ goto Exit;
}
- // Lock the capture buffer
- LPVOID audioPtr;
- DWORD dataLen;
- result = buffer->Lock( 0, bufferBytes, &audioPtr, &dataLen, NULL, NULL, 0 );
- if ( FAILED( result ) ) {
- input->Release();
- buffer->Release();
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking input buffer (" << dsinfo.name << ")!";
- errorText_ = errorStream_.str();
- return FAILURE;
+ // retrieve renderAudioClient from devicePtr
+ IAudioClient*& renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
+
+ hr = renderDevices->Item( device, &devicePtr );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device handle.";
+ goto Exit;
}
- // Zero the buffer
- ZeroMemory( audioPtr, dataLen );
+ hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
+ NULL, ( void** ) &renderAudioClient );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client.";
+ goto Exit;
+ }
- // 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 (" << dsinfo.name << ")!";
- errorText_ = errorStream_.str();
- return FAILURE;
+ hr = renderAudioClient->GetMixFormat( &deviceFormat );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format.";
+ goto Exit;
}
- dsBufferSize = bufferBytes;
- ohandle = (void *) input;
- bhandle = (void *) buffer;
+ stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
+ renderAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
}
- // Set various stream parameters
- stream_.nDeviceChannels[mode] = channels + firstChannel;
- stream_.nUserChannels[mode] = channels;
+ // 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] = deviceInfo.nativeFormats;
+
+ if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
+ stream_.userInterleaved = false;
+ else
+ stream_.userInterleaved = true;
stream_.deviceInterleaved[mode] = true;
- if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
- else stream_.userInterleaved = true;
- // Set flag for buffer conversion
+ // Set flags for buffer conversion.
stream_.doConvertBuffer[mode] = false;
- if (stream_.nUserChannels[mode] != stream_.nDeviceChannels[mode])
- stream_.doConvertBuffer[mode] = true;
- if (stream_.userFormat != stream_.deviceFormat[mode])
+ if ( stream_.userFormat != stream_.deviceFormat[mode] ||
+ stream_.nUserChannels != stream_.nDeviceChannels )
stream_.doConvertBuffer[mode] = true;
- if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
- stream_.nUserChannels[mode] > 1 )
+ else if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+ stream_.nUserChannels[mode] > 1 )
stream_.doConvertBuffer[mode] = true;
+ if ( stream_.doConvertBuffer[mode] )
+ setConvertInfo( mode, 0 );
+
// Allocate necessary internal buffers
- bufferBytes = stream_.nUserChannels[mode] * *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;
+ 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 ( stream_.doConvertBuffer[mode] ) {
+ if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME )
+ stream_.callbackInfo.priority = 15;
+ else
+ stream_.callbackInfo.priority = 0;
- 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;
+ ///! 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;
+ WasapiBuffer captureBuffer;
+ WasapiBuffer renderBuffer;
+
+ // declare local stream variables
+ RtAudioCallback callback = ( RtAudioCallback ) stream_.callbackInfo.callback;
+ BYTE* streamBuffer = NULL;
+ unsigned long captureFlags = 0;
+ unsigned int bufferFrameCount = 0;
+ unsigned int numFramesPadding = 0;
+ bool callbackPushed = false;
+ bool callbackPulled = false;
+ bool callbackStopped = false;
+ int callbackResult = 0;
+
+ unsigned int deviceBuffSize = 0;
+
+ errorText_.clear();
+ RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
+
+ // Attempt to assign "Pro Audio" characteristic to thread
+ HMODULE AvrtDll = LoadLibrary( (LPCTSTR) "AVRT.dll" );
+ if ( AvrtDll ) {
+ DWORD taskIndex = 0;
+ TAvSetMmThreadCharacteristicsPtr AvSetMmThreadCharacteristicsPtr = ( TAvSetMmThreadCharacteristicsPtr ) GetProcAddress( AvrtDll, "AvSetMmThreadCharacteristicsW" );
+ AvSetMmThreadCharacteristicsPtr( L"Pro Audio", &taskIndex );
+ FreeLibrary( AvrtDll );
+ }
+
+ // start capture stream if applicable
+ if ( captureAudioClient ) {
+ hr = captureAudioClient->GetMixFormat( &captureFormat );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
+ goto Exit;
+ }
+
+ // initialize capture stream according to desire buffer size
+ REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) stream_.bufferSize * 10000000 / captureFormat->nSamplesPerSec );
+
+ if ( !captureClient ) {
+ hr = captureAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
+ AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
+ desiredBufferPeriod,
+ desiredBufferPeriod,
+ captureFormat,
+ NULL );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize capture audio client.";
+ goto Exit;
}
- }
- 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;
+ hr = captureAudioClient->GetService( __uuidof( IAudioCaptureClient ),
+ ( void** ) &captureClient );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture client handle.";
+ goto Exit;
}
+
+ // configure captureEvent to trigger on every available capture buffer
+ captureEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
+ if ( !captureEvent ) {
+ errorType = RtAudioError::SYSTEM_ERROR;
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to create capture event.";
+ goto Exit;
+ }
+
+ hr = captureAudioClient->SetEventHandle( captureEvent );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to set capture event handle.";
+ goto Exit;
+ }
+
+ ( ( WasapiHandle* ) stream_.apiHandle )->captureClient = captureClient;
+ ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent = captureEvent;
}
- }
- // Allocate our DsHandle structures for the stream.
- DsHandle *handle;
- if ( stream_.apiHandle == 0 ) {
- try {
- handle = new DsHandle;
+ unsigned int inBufferSize = 0;
+ hr = captureAudioClient->GetBufferSize( &inBufferSize );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to get capture buffer size.";
+ goto Exit;
}
- catch ( std::bad_alloc& ) {
- errorText_ = "RtApiDs::probeDeviceOpen: error allocating AsioHandle memory.";
- goto error;
+
+ // scale outBufferSize according to stream->user sample rate ratio
+ unsigned int outBufferSize = ( unsigned int ) stream_.bufferSize * stream_.nDeviceChannels[INPUT];
+ inBufferSize *= stream_.nDeviceChannels[INPUT];
+
+ // set captureBuffer size
+ captureBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[INPUT] ) );
+
+ // reset the capture stream
+ hr = captureAudioClient->Reset();
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to reset capture stream.";
+ goto Exit;
}
- // Create a manual-reset event.
- handle->condition = CreateEvent( NULL, // no security
- TRUE, // manual-reset
- FALSE, // non-signaled initially
- NULL ); // unnamed
- stream_.apiHandle = (void *) handle;
+ // start the capture stream
+ hr = captureAudioClient->Start();
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to start capture stream.";
+ goto Exit;
+ }
}
- 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;
+ // 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;
+ }
- // Setup the buffer conversion information structure.
- if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
+ // initialize render stream according to desire buffer size
+ REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) stream_.bufferSize * 10000000 / renderFormat->nSamplesPerSec );
- // Setup the callback thread.
- unsigned threadId;
- stream_.callbackInfo.object = (void *) this;
- stream_.callbackInfo.isRunning = true;
- stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &callbackHandler,
- &stream_.callbackInfo, 0, &threadId );
- if ( stream_.callbackInfo.thread == 0 ) {
- errorText_ = "RtApiDs::probeDeviceOpen: error creating callback thread!";
- goto error;
- }
+ if ( !renderClient ) {
+ hr = renderAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
+ AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
+ desiredBufferPeriod,
+ desiredBufferPeriod,
+ renderFormat,
+ NULL );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize render audio client.";
+ goto Exit;
+ }
- return SUCCESS;
+ hr = renderAudioClient->GetService( __uuidof( IAudioRenderClient ),
+ ( void** ) &renderClient );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render client handle.";
+ goto Exit;
+ }
- 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();
+ // 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;
}
- if ( handle->buffer[1] ) {
- LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
- LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
- if ( buffer ) buffer->Release();
- object->Release();
+
+ unsigned int outBufferSize = 0;
+ hr = renderAudioClient->GetBufferSize( &outBufferSize );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to get render buffer size.";
+ goto Exit;
}
- 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;
+ // scale inBufferSize according to user->stream sample rate ratio
+ unsigned int inBufferSize = ( unsigned int ) stream_.bufferSize * stream_.nDeviceChannels[OUTPUT];
+ outBufferSize *= stream_.nDeviceChannels[OUTPUT];
+
+ // set renderBuffer size
+ renderBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[OUTPUT] ) );
+
+ // reset the render stream
+ hr = renderAudioClient->Reset();
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to reset render stream.";
+ goto Exit;
+ }
+
+ // start the render stream
+ hr = renderAudioClient->Start();
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to start render stream.";
+ goto Exit;
}
}
- if ( stream_.deviceBuffer ) {
- free( stream_.deviceBuffer );
- stream_.deviceBuffer = 0;
+ if ( stream_.mode == INPUT ) {
+ using namespace std; // for roundf
+ deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
+ }
+ else if ( stream_.mode == OUTPUT ) {
+ deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
+ }
+ else if ( stream_.mode == DUPLEX ) {
+ deviceBuffSize = std::max( stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
+ stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
}
- return FAILURE;
-}
+ stream_.deviceBuffer = ( char* ) malloc( deviceBuffSize );
+ if ( !stream_.deviceBuffer ) {
+ errorType = RtAudioError::MEMORY_ERROR;
+ errorText_ = "RtApiWasapi::wasapiThread: Error allocating device buffer memory.";
+ goto Exit;
+ }
-void RtApiDs :: closeStream()
-{
- if ( stream_.state == STREAM_CLOSED ) {
- errorText_ = "RtApiDs::closeStream(): no open stream to close!";
- error( RtError::WARNING );
- return;
- }
+ // 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 format
- // Stop the callback thread.
- stream_.callbackInfo.isRunning = false;
- WaitForSingleObject( (HANDLE) stream_.callbackInfo.thread, INFINITE );
- CloseHandle( (HANDLE) stream_.callbackInfo.thread );
+ if ( captureAudioClient ) {
+ // Pull callback buffer from inputBuffer
+ callbackPulled = captureBuffer.pullBuffer( stream_.deviceBuffer,
+ ( unsigned int ) stream_.bufferSize * stream_.nDeviceChannels[INPUT],
+ stream_.deviceFormat[INPUT] );
- 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();
+ 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 ) );
+ }
+ }
}
- object->Release();
- }
- if ( handle->buffer[1] ) {
- LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
- LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
- if ( buffer ) {
- buffer->Stop();
- buffer->Release();
+ else {
+ // if there is no capture stream, set callbackPulled flag
+ callbackPulled = true;
}
- object->Release();
- }
- CloseHandle( handle->condition );
- delete handle;
- stream_.apiHandle = 0;
- }
-
- for ( int i=0; i<2; i++ ) {
- if ( stream_.userBuffer[i] ) {
- free( stream_.userBuffer[i] );
- stream_.userBuffer[i] = 0;
- }
- }
-
- if ( stream_.deviceBuffer ) {
- free( stream_.deviceBuffer );
- stream_.deviceBuffer = 0;
- }
-
- stream_.mode = UNINITIALIZED;
- stream_.state = STREAM_CLOSED;
-}
-
-void RtApiDs :: startStream()
-{
- verifyStream();
- if ( stream_.state == STREAM_RUNNING ) {
- errorText_ = "RtApiDs::startStream(): the stream is already running!";
- error( RtError::WARNING );
- return;
- }
-
- // Increase scheduler frequency on lesser windows (a side-effect of
- // increasing timer accuracy). On greater windows (Win2K or later),
- // this is already in effect.
- MUTEX_LOCK( &stream_.mutex );
-
- DsHandle *handle = (DsHandle *) stream_.apiHandle;
+ // Execute Callback
+ // ================
+ // 1. Execute user callback method
+ // 2. Handle return value from callback
+
+ // if callback has not requested the stream to stop
+ if ( callbackPulled && !callbackStopped ) {
+ // Execute user callback method
+ callbackResult = callback( stream_.userBuffer[OUTPUT],
+ stream_.userBuffer[INPUT],
+ stream_.bufferSize,
+ getStreamTime(),
+ captureFlags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY ? RTAUDIO_INPUT_OVERFLOW : 0,
+ stream_.callbackInfo.userData );
+
+ // Handle return value from callback
+ if ( callbackResult == 1 ) {
+ // instantiate a thread to stop this thread
+ HANDLE threadHandle = CreateThread( NULL, 0, stopWasapiThread, this, 0, NULL );
+ if ( !threadHandle ) {
+ errorType = RtAudioError::THREAD_ERROR;
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream stop thread.";
+ goto Exit;
+ }
+ else if ( !CloseHandle( threadHandle ) ) {
+ errorType = RtAudioError::THREAD_ERROR;
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream stop thread handle.";
+ goto Exit;
+ }
- timeBeginPeriod( 1 );
+ 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;
+ }
- /*
- memset( &statistics, 0, sizeof( statistics ) );
- statistics.sampleRate = stream_.sampleRate;
- statistics.writeDeviceBufferLeadBytes = handle->dsPointerLeadTime[0];
- */
+ callbackStopped = true;
+ }
+ }
+ }
- buffersRolling = false;
- duplexPrerollBytes = 0;
+ // Callback Output
+ // ===============
+ // 1. Convert callback buffer to stream format
+ // 2. Push callback buffer into outputBuffer
- 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] );
- }
+ if ( renderAudioClient && callbackPulled ) {
+ if ( stream_.doConvertBuffer[OUTPUT] ) {
+ // Convert callback buffer to stream format
+ convertBuffer( stream_.deviceBuffer,
+ stream_.userBuffer[OUTPUT],
+ stream_.convertInfo[OUTPUT] );
- HRESULT result;
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
- //statistics.outputFrameSize = formatBytes( stream_.deviceFormat[0] ) * stream_.nDeviceChannels[0];
+ }
- 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;
+ // Push callback buffer into outputBuffer
+ callbackPushed = renderBuffer.pushBuffer( stream_.deviceBuffer,
+ stream_.bufferSize * stream_.nDeviceChannels[OUTPUT],
+ stream_.deviceFormat[OUTPUT] );
}
- }
-
- if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
- //statistics.inputFrameSize = formatBytes( stream_.deviceFormat[1]) * stream_.nDeviceChannels[1];
-
- LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
- result = buffer->Start( DSCBSTART_LOOPING );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting input buffer!";
- errorText_ = errorStream_.str();
- goto unlock;
+ else {
+ // if there is no render stream, set callbackPushed flag
+ callbackPushed = true;
}
- }
-
- handle->drainCounter = 0;
- handle->internalDrain = false;
- stream_.state = STREAM_RUNNING;
-
- unlock:
- MUTEX_UNLOCK( &stream_.mutex );
- if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR );
-}
+ // Stream Capture
+ // ==============
+ // 1. Get capture buffer from stream
+ // 2. Push capture buffer into inputBuffer
+ // 3. If 2. was successful: Release capture buffer
-void RtApiDs :: stopStream()
-{
- verifyStream();
- if ( stream_.state == STREAM_STOPPED ) {
- errorText_ = "RtApiDs::stopStream(): the stream is already stopped!";
- error( RtError::WARNING );
- return;
- }
+ if ( captureAudioClient ) {
+ // if the callback input buffer was not pulled from captureBuffer, wait for next capture event
+ if ( !callbackPulled ) {
+ WaitForSingleObject( captureEvent, INFINITE );
+ }
- MUTEX_LOCK( &stream_.mutex );
+ // 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;
+ }
- HRESULT result;
- LPVOID audioPtr;
- DWORD dataLen;
- DsHandle *handle = (DsHandle *) stream_.apiHandle;
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
- if ( handle->drainCounter == 0 ) {
- handle->drainCounter = 1;
- MUTEX_UNLOCK( &stream_.mutex );
- WaitForMultipleObjects( 1, &handle->condition, FALSE, INFINITE ); // block until signaled
- ResetEvent( handle->condition );
- MUTEX_LOCK( &stream_.mutex );
+ 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;
+ }
+ }
}
- // Stop the buffer and clear memory
- LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
- result = buffer->Stop();
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") stopping output buffer!";
- errorText_ = errorStream_.str();
- goto unlock;
- }
+ // 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
- // Lock the buffer and clear it so that if we start to play again,
- // we won't have old data playing.
- result = buffer->Lock( 0, handle->dsBufferSize[0], &audioPtr, &dataLen, NULL, NULL, 0 );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") locking output buffer!";
- errorText_ = errorStream_.str();
- goto unlock;
- }
+ if ( renderAudioClient ) {
+ // if the callback output buffer was not pushed to renderBuffer, wait for next render event
+ if ( callbackPulled && !callbackPushed ) {
+ WaitForSingleObject( renderEvent, INFINITE );
+ }
- // Zero the DS buffer
- ZeroMemory( audioPtr, dataLen );
+ // Get render buffer from stream
+ hr = renderAudioClient->GetBufferSize( &bufferFrameCount );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer size.";
+ goto Exit;
+ }
- // Unlock the DS buffer
- result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") unlocking output buffer!";
- errorText_ = errorStream_.str();
- goto unlock;
- }
+ hr = renderAudioClient->GetCurrentPadding( &numFramesPadding );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer padding.";
+ goto Exit;
+ }
- // If we start playing again, we must begin at beginning of buffer.
- handle->bufferPointer[0] = 0;
- }
+ bufferFrameCount -= numFramesPadding;
- if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
- LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
- audioPtr = NULL;
- dataLen = 0;
+ if ( bufferFrameCount != 0 ) {
+ hr = renderClient->GetBuffer( bufferFrameCount, &streamBuffer );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer.";
+ goto Exit;
+ }
- result = buffer->Stop();
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") stopping input buffer!";
- errorText_ = errorStream_.str();
- goto unlock;
+ // 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;
+ }
+ }
}
- // Lock the buffer and clear it so that if we start to play again,
- // we won't have old data playing.
- result = buffer->Lock( 0, handle->dsBufferSize[1], &audioPtr, &dataLen, NULL, NULL, 0 );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") locking input buffer!";
- errorText_ = errorStream_.str();
- goto unlock;
+ // if the callback buffer was pushed renderBuffer reset callbackPulled flag
+ if ( callbackPushed ) {
+ callbackPulled = false;
+ // tick stream time
+ RtApi::tickStreamTime();
}
- // Zero the DS buffer
- ZeroMemory( audioPtr, dataLen );
+ }
- // Unlock the DS buffer
- result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") unlocking input buffer!";
- errorText_ = errorStream_.str();
- goto unlock;
- }
+Exit:
+ // clean up
+ CoTaskMemFree( captureFormat );
+ CoTaskMemFree( renderFormat );
- // If we start recording again, we must begin at beginning of buffer.
- handle->bufferPointer[1] = 0;
- }
+ CoUninitialize();
- unlock:
- timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows.
+ // update stream state
stream_.state = STREAM_STOPPED;
- MUTEX_UNLOCK( &stream_.mutex );
- if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR );
-}
-void RtApiDs :: abortStream()
-{
- verifyStream();
- if ( stream_.state == STREAM_STOPPED ) {
- errorText_ = "RtApiDs::abortStream(): the stream is already stopped!";
- error( RtError::WARNING );
+ if ( errorText_.empty() )
return;
- }
+ else
+ error( errorType );
+}
- DsHandle *handle = (DsHandle *) stream_.apiHandle;
- handle->drainCounter = 1;
+//******************** End of __WINDOWS_WASAPI__ *********************//
+#endif
- stopStream();
-}
-void RtApiDs :: callbackEvent()
-{
- if ( stream_.state == STREAM_STOPPED ) {
- Sleep(50); // sleep 50 milliseconds
- return;
- }
+#if defined(__WINDOWS_DS__) // Windows DirectSound API
- if ( stream_.state == STREAM_CLOSED ) {
- errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!";
- error( RtError::WARNING );
- return;
- }
+// 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
- CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
- DsHandle *handle = (DsHandle *) stream_.apiHandle;
+#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
- // Check if we were draining the stream and signal is finished.
- if ( handle->drainCounter > stream_.nBuffers + 2 ) {
- if ( handle->internalDrain == false )
- SetEvent( handle->condition );
- else
- stopStream();
- return;
- }
+#define MINIMUM_DEVICE_BUFFER_SIZE 32768
- MUTEX_LOCK( &stream_.mutex );
+#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
- // Invoke user callback to get fresh output data UNLESS we are
- // draining stream.
- if ( handle->drainCounter == 0 ) {
- RtAudioCallback callback = (RtAudioCallback) info->callback;
- double streamTime = getStreamTime();
- RtAudioStreamStatus status = 0;
- if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
- status |= RTAUDIO_OUTPUT_UNDERFLOW;
- handle->xrun[0] = false;
- }
- if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
- status |= RTAUDIO_INPUT_OVERFLOW;
- handle->xrun[1] = false;
- }
- handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1],
- stream_.bufferSize, streamTime, status, info->userData );
- if ( handle->drainCounter == 2 ) {
- MUTEX_UNLOCK( &stream_.mutex );
- abortStream();
- return;
- }
- else if ( handle->drainCounter == 1 )
- handle->internalDrain = true;
- }
+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;
+}
- HRESULT result;
- DWORD currentWritePos, safeWritePos;
- DWORD currentReadPos, safeReadPos;
- DWORD leadPos;
- UINT nextWritePos;
+// 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;
-#ifdef GENERATE_DEBUG_LOG
- DWORD writeTime, readTime;
-#endif
+ 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; }
+};
- LPVOID buffer1 = NULL;
- LPVOID buffer2 = NULL;
- DWORD bufferSize1 = 0;
- DWORD bufferSize2 = 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 );
- char *buffer;
- long bufferBytes;
+static const char* getErrorString( int code );
- if ( stream_.mode == DUPLEX && !buffersRolling ) {
- assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
+static unsigned __stdcall callbackHandler( void *ptr );
- // 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.
+struct DsDevice {
+ LPGUID id[2];
+ bool validId[2];
+ bool found;
+ std::string name;
- // 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.
+ DsDevice()
+ : found(false) { validId[0] = false; validId[1] = false; }
+};
- LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
- LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+struct DsProbeData {
+ bool isInput;
+ std::vector<struct DsDevice>* dsDevices;
+};
- DWORD initialWritePos, initialSafeWritePos;
- DWORD initialReadPos, initialSafeReadPos;
+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;
+}
- result = dsWriteBuffer->GetCurrentPosition( &initialWritePos, &initialSafeWritePos );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
- errorText_ = errorStream_.str();
- error( RtError::SYSTEM_ERROR );
- }
- result = dsCaptureBuffer->GetCurrentPosition( &initialReadPos, &initialSafeReadPos );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
- errorText_ = errorStream_.str();
- error( RtError::SYSTEM_ERROR );
- }
- while ( true ) {
- result = dsWriteBuffer->GetCurrentPosition( ¤tWritePos, &safeWritePos );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
- errorText_ = errorStream_.str();
- error( RtError::SYSTEM_ERROR );
- }
- result = dsCaptureBuffer->GetCurrentPosition( ¤tReadPos, &safeReadPos );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
- errorText_ = errorStream_.str();
- error( RtError::SYSTEM_ERROR );
- }
- if ( safeWritePos != initialSafeWritePos && safeReadPos != initialSafeReadPos ) break;
- Sleep( 1 );
- }
+RtApiDs :: ~RtApiDs()
+{
+ if ( stream_.state != STREAM_CLOSED ) closeStream();
+ if ( coInitialized_ ) CoUninitialize(); // balanced call.
+}
- assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
+// The DirectSound default output is always the first device.
+unsigned int RtApiDs :: getDefaultOutputDevice( void )
+{
+ return 0;
+}
- buffersRolling = true;
- handle->bufferPointer[0] = ( safeWritePos + handle->dsPointerLeadTime[0] );
- handle->bufferPointer[1] = safeReadPos;
+// 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 );
}
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-
- LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+ // 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 );
+ }
- 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 );
- }
+ // 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++;
+ }
- // 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 );
+ 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;
}
+ }
- // No byte swapping necessary in DirectSound implementation.
+ if ( device >= dsDevices.size() ) {
+ errorText_ = "RtApiDs::getDeviceInfo: device ID is invalid!";
+ error( RtAudioError::INVALID_USE );
+ return info;
+ }
- // 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 );
+ HRESULT result;
+ if ( dsDevices[ device ].validId[0] == false ) goto probeInput;
- DWORD dsBufferSize = handle->dsBufferSize[0];
- nextWritePos = handle->bufferPointer[0];
+ 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;
+ }
- DWORD endWrite;
- while ( true ) {
- // Find out where the read and "safe write" pointers are.
- result = dsBuffer->GetCurrentPosition( ¤tWritePos, &safeWritePos );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
- errorText_ = errorStream_.str();
- error( RtError::SYSTEM_ERROR );
- }
+ 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;
+ }
- leadPos = safeWritePos + handle->dsPointerLeadTime[0];
- if ( leadPos > dsBufferSize ) leadPos -= dsBufferSize;
- if ( leadPos < nextWritePos ) leadPos += dsBufferSize; // unwrap offset
- endWrite = nextWritePos + bufferBytes;
+ // Get output channel information.
+ info.outputChannels = ( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1;
- // Check whether the entire write region is behind the play pointer.
- if ( leadPos >= endWrite ) break;
-
- // If we are here, then we must wait until the play pointer gets
- // beyond the write region. The approach here is to use the
- // Sleep() function to suspend operation until safePos catches
- // up. Calculate number of milliseconds to wait as:
- // time = distance * (milliseconds/second) * fudgefactor /
- // ((bytes/sample) * (samples/second))
- // A "fudgefactor" less than 1 is used because it was found
- // that sleeping too long was MUCH worse than sleeping for
- // several shorter periods.
- double millis = ( endWrite - leadPos ) * 900.0;
- millis /= ( formatBytes( stream_.deviceFormat[0]) * stream_.nDeviceChannels[0] * stream_.sampleRate);
- if ( millis < 1.0 ) millis = 1.0;
- if ( millis > 50.0 ) {
- static int nOverruns = 0;
- ++nOverruns;
- }
- Sleep( (DWORD) millis );
+ // 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];
}
+ }
- //if ( statistics.writeDeviceSafeLeadBytes < dsPointerDifference( safeWritePos, currentWritePos, handle->dsBufferSize[0] ) ) {
- // statistics.writeDeviceSafeLeadBytes = dsPointerDifference( safeWritePos, currentWritePos, handle->dsBufferSize[0] );
- //}
+ // Get format information.
+ if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT ) info.nativeFormats |= RTAUDIO_SINT16;
+ if ( outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) info.nativeFormats |= RTAUDIO_SINT8;
- if ( dsPointerBetween( nextWritePos, safeWritePos, currentWritePos, dsBufferSize )
- || dsPointerBetween( endWrite, safeWritePos, currentWritePos, dsBufferSize ) ) {
- // We've strayed into the forbidden zone ... resync the read pointer.
- //++statistics.numberOfWriteUnderruns;
- handle->xrun[0] = true;
- nextWritePos = safeWritePos + handle->dsPointerLeadTime[0] - bufferBytes + dsBufferSize;
- while ( nextWritePos >= dsBufferSize ) nextWritePos -= dsBufferSize;
- handle->bufferPointer[0] = nextWritePos;
- endWrite = nextWritePos + bufferBytes;
- }
+ output->Release();
- // Lock free space in the buffer
- result = dsBuffer->Lock( nextWritePos, bufferBytes, &buffer1,
- &bufferSize1, &buffer2, &bufferSize2, 0 );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking buffer during playback!";
- errorText_ = errorStream_.str();
- error( RtError::SYSTEM_ERROR );
- }
+ if ( getDefaultOutputDevice() == device )
+ info.isDefaultOutput = true;
- // Copy our buffer into the DS buffer
- CopyMemory( buffer1, buffer, bufferSize1 );
- if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 );
+ if ( dsDevices[ device ].validId[1] == false ) {
+ info.name = dsDevices[ device ].name;
+ info.probed = true;
+ return info;
+ }
- // Update our buffer offset and unlock sound buffer
- dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!";
- errorText_ = errorStream_.str();
- error( RtError::SYSTEM_ERROR );
- }
- nextWritePos = ( nextWritePos + bufferSize1 + bufferSize2 ) % dsBufferSize;
- handle->bufferPointer[0] = nextWritePos;
+ probeInput:
- if ( handle->drainCounter ) {
- handle->drainCounter++;
- goto unlock;
- }
+ 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;
}
- if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+ 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;
+ }
- // 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 );
- }
+ // Get input channel information.
+ info.inputChannels = inCaps.dwChannels;
- LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
- long nextReadPos = handle->bufferPointer[1];
- DWORD dsBufferSize = handle->dsBufferSize[1];
+ // 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;
- // Find out where the write and "safe read" pointers are.
- result = dsBuffer->GetCurrentPosition( ¤tReadPos, &safeReadPos );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
- errorText_ = errorStream_.str();
- error( RtError::SYSTEM_ERROR );
+ if ( 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 ( safeReadPos < (DWORD)nextReadPos ) safeReadPos += dsBufferSize; // unwrap offset
- DWORD endRead = nextReadPos + bufferBytes;
-
- // Handling depends on whether we are INPUT or DUPLEX.
- // If we're in INPUT mode then waiting is a good thing. If we're in DUPLEX mode,
- // then a wait here will drag the write pointers into the forbidden zone.
- //
- // In DUPLEX mode, rather than wait, we will back off the read pointer until
- // it's in a safe position. This causes dropouts, but it seems to be the only
- // practical way to sync up the read and write pointers reliably, given the
- // the very complex relationship between phase and increment of the read and write
- // pointers.
- //
- // In order to minimize audible dropouts in DUPLEX mode, we will
- // provide a pre-roll period of 0.5 seconds in which we return
- // zeros from the read buffer while the pointers sync up.
-
- if ( stream_.mode == DUPLEX ) {
- if ( safeReadPos < endRead ) {
- if ( duplexPrerollBytes <= 0 ) {
- // Pre-roll time over. Be more agressive.
- int adjustment = endRead-safeReadPos;
-
- handle->xrun[1] = true;
- //++statistics.numberOfReadOverruns;
- // Two cases:
- // - large adjustments: we've probably run out of CPU cycles, so just resync exactly,
- // and perform fine adjustments later.
- // - small adjustments: back off by twice as much.
- if ( adjustment >= 2*bufferBytes )
- nextReadPos = safeReadPos-2*bufferBytes;
- else
- nextReadPos = safeReadPos-bufferBytes-adjustment;
-
- //statistics.readDeviceSafeLeadBytes = currentReadPos-nextReadPos;
- //if ( statistics.readDeviceSafeLeadBytes < 0) statistics.readDeviceSafeLeadBytes += dsBufferSize;
- if ( nextReadPos < 0 ) nextReadPos += dsBufferSize;
-
- }
- else {
- // In pre=roll time. Just do it.
- nextReadPos = safeReadPos-bufferBytes;
- while ( nextReadPos < 0 ) nextReadPos += dsBufferSize;
- }
- endRead = nextReadPos + bufferBytes;
- }
- }
- else { // mode == INPUT
- while ( safeReadPos < endRead ) {
- // See comments for playback.
- double millis = (endRead - safeReadPos) * 900.0;
- millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate);
- if ( millis < 1.0 ) millis = 1.0;
- Sleep( (DWORD) millis );
-
- // Wake up, find out where we are now
- result = dsBuffer->GetCurrentPosition( ¤tReadPos, &safeReadPos );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
- errorText_ = errorStream_.str();
- error( RtError::SYSTEM_ERROR );
- }
-
- if ( safeReadPos < (DWORD)nextReadPos ) safeReadPos += dsBufferSize; // unwrap offset
- }
- }
-
- //if (statistics.readDeviceSafeLeadBytes < dsPointerDifference( currentReadPos, nextReadPos, dsBufferSize ) )
- // statistics.readDeviceSafeLeadBytes = dsPointerDifference( currentReadPos, nextReadPos, dsBufferSize );
-
- // Lock free space in the buffer
- result = dsBuffer->Lock( nextReadPos, bufferBytes, &buffer1,
- &bufferSize1, &buffer2, &bufferSize2, 0 );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking capture buffer!";
- errorText_ = errorStream_.str();
- error( RtError::SYSTEM_ERROR );
- }
-
- if ( 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;
+ 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 );
}
-
- // Update our buffer offset and unlock sound buffer
- nextReadPos = ( nextReadPos + bufferSize1 + bufferSize2 ) % dsBufferSize;
- dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
- if ( FAILED( result ) ) {
- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking capture buffer!";
- errorText_ = errorStream_.str();
- error( RtError::SYSTEM_ERROR );
+ 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 );
}
- handle->bufferPointer[1] = nextReadPos;
-
- // No byte swapping necessary in DirectSound implementation.
-
- // If necessary, convert 8-bit data from unsigned to signed.
- if ( stream_.deviceFormat[1] == RTAUDIO_SINT8 )
- for ( int j=0; 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] );
- }
-#ifdef GENERATE_DEBUG_LOG
- if ( currentDebugLogEntry < debugLog.size() )
- {
- TTickRecord &r = debugLog[currentDebugLogEntry++];
- r.currentReadPointer = currentReadPos;
- r.safeReadPointer = safeReadPos;
- r.currentWritePointer = currentWritePos;
- r.safeWritePointer = safeWritePos;
- r.readTime = readTime;
- r.writeTime = writeTime;
- r.nextReadPointer = handles[1].bufferPointer;
- r.nextWritePointer = handles[0].bufferPointer;
}
-#endif
-
- unlock:
- MUTEX_UNLOCK( &stream_.mutex );
-
- RtApi::tickStreamTime();
-}
+ else info.inputChannels = 0; // technically, this would be an error
-// Definitions for utility functions and callbacks
-// specific to the DirectSound implementation.
+ input->Release();
-extern "C" unsigned __stdcall callbackHandler( void *ptr )
-{
- CallbackInfo *info = (CallbackInfo *) ptr;
- RtApiDs *object = (RtApiDs *) info->object;
- bool* isRunning = &info->isRunning;
+ if ( info.inputChannels == 0 ) return info;
- while ( *isRunning == true ) {
- object->callbackEvent();
+ // 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() );
- _endthreadex( 0 );
- return 0;
-}
-
-#include "tchar.h"
-
-std::string convertTChar( LPCTSTR name )
-{
- std::string s;
+ // 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 defined( UNICODE ) || defined( _UNICODE )
- // Yes, this conversion doesn't make sense for two-byte characters
- // but RtAudio is currently written to return an std::string of
- // one-byte chars for the device name.
- for ( unsigned int i=0; i<wcslen( name ); i++ )
- s.push_back( name[i] );
-#else
- s.append( std::string( name ) );
-#endif
+ if ( device == 0 ) info.isDefaultInput = true;
- return s;
+ // Copy name and return.
+ info.name = dsDevices[ device ].name;
+ info.probed = true;
+ return info;
}
-static bool CALLBACK deviceCountCallback( LPGUID lpguid,
- LPCTSTR description,
- LPCTSTR module,
- LPVOID lpContext )
+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 )
{
- EnumInfo *info = (EnumInfo *) lpContext;
-
- HRESULT hr;
- if ( info->isInput == true ) {
- DSCCAPS caps;
- LPDIRECTSOUNDCAPTURE object;
+ if ( channels + firstChannel > 2 ) {
+ errorText_ = "RtApiDs::probeDeviceOpen: DirectSound does not support more than 2 channels per device.";
+ return FAILURE;
+ }
- hr = DirectSoundCaptureCreate( lpguid, &object, NULL );
- if ( hr != DS_OK ) return true;
+ 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;
+ }
- caps.dwSize = sizeof(caps);
- hr = object->GetCaps( &caps );
- if ( hr == DS_OK ) {
- if ( caps.dwChannels > 0 && caps.dwFormats > 0 )
- info->counter++;
- }
- object->Release();
+ 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;
}
- else {
- DSCAPS caps;
- LPDIRECTSOUND object;
- hr = DirectSoundCreate( lpguid, &object, NULL );
- if ( hr != DS_OK ) return true;
- caps.dwSize = sizeof(caps);
- hr = object->GetCaps( &caps );
- if ( hr == DS_OK ) {
- if ( caps.dwFlags & DSCAPS_PRIMARYMONO || caps.dwFlags & DSCAPS_PRIMARYSTEREO )
- info->counter++;
+ if ( mode == OUTPUT ) {
+ if ( dsDevices[ device ].validId[0] == false ) {
+ errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support output!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
}
- object->Release();
}
-
- if ( info->getDefault && lpguid == NULL ) return false;
-
- if ( info->findIndex && info->counter > info->index ) {
- info->id = lpguid;
- info->name = convertTChar( description );
- return false;
+ else { // mode == INPUT
+ if ( dsDevices[ device ].validId[1] == false ) {
+ errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support input!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
}
- return true;
-}
+ // 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();
-static char* getErrorString( int code )
-{
- switch (code) {
+ // 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;
- case DSERR_ALLOCATED:
- return "Already allocated";
+ // Check the lower range of the user-specified buffer size and set
+ // (arbitrarily) to a lower bound of 32.
+ if ( *bufferSize < 32 ) *bufferSize = 32;
- case DSERR_CONTROLUNAVAIL:
- return "Control unavailable";
+ // 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;
- case DSERR_INVALIDPARAM:
- return "Invalid parameter";
+ // 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;
- case DSERR_INVALIDCALL:
- return "Invalid call";
+ void *ohandle = 0, *bhandle = 0;
+ HRESULT result;
+ if ( mode == OUTPUT ) {
- case DSERR_GENERIC:
- return "Generic error";
+ 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;
+ }
- case DSERR_PRIOLEVELNEEDED:
- return "Priority level needed";
+ 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;
+ }
- case DSERR_OUTOFMEMORY:
- return "Out of memory";
+ // 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;
+ }
- case DSERR_BADFORMAT:
- return "The sample rate or the channel format is not supported";
+ // 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;
- case DSERR_UNSUPPORTED:
- return "Not supported";
+ // 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;
- case DSERR_NODRIVER:
- return "No driver";
+ // If the user wants an even bigger buffer, increase the device buffer size accordingly.
+ while ( dsPointerLeadTime * 2U > dsBufferSize )
+ dsBufferSize *= 2;
- case DSERR_ALREADYINITIALIZED:
- return "Already initialized";
+ // Set cooperative level to DSSCL_EXCLUSIVE ... sound stops when window focus changes.
+ // result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE );
+ // Set cooperative level to DSSCL_PRIORITY ... sound remains when window focus changes.
+ result = output->SetCooperativeLevel( hWnd, DSSCL_PRIORITY );
+ if ( FAILED( result ) ) {
+ output->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting cooperative level (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Even though we will write to the secondary buffer, we need to
+ // access the primary buffer to set the correct output format
+ // (since the default is 8-bit, 22 kHz!). Setup the DS primary
+ // buffer description.
+ DSBUFFERDESC bufferDescription;
+ ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
+ bufferDescription.dwSize = sizeof( DSBUFFERDESC );
+ bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER;
+
+ // Obtain the primary buffer
+ LPDIRECTSOUNDBUFFER buffer;
+ result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
+ if ( FAILED( result ) ) {
+ output->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") accessing primary buffer (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Set the primary DS buffer sound format.
+ result = buffer->SetFormat( &waveFormat );
+ if ( FAILED( result ) ) {
+ output->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting primary buffer format (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Setup the secondary DS buffer description.
+ ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
+ bufferDescription.dwSize = sizeof( DSBUFFERDESC );
+ bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
+ DSBCAPS_GLOBALFOCUS |
+ DSBCAPS_GETCURRENTPOSITION2 |
+ DSBCAPS_LOCHARDWARE ); // Force hardware mixing
+ bufferDescription.dwBufferBytes = dsBufferSize;
+ bufferDescription.lpwfxFormat = &waveFormat;
+
+ // Try to create the secondary DS buffer. If that doesn't work,
+ // try to use software mixing. Otherwise, there's a problem.
+ result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
+ if ( FAILED( result ) ) {
+ bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
+ DSBCAPS_GLOBALFOCUS |
+ DSBCAPS_GETCURRENTPOSITION2 |
+ DSBCAPS_LOCSOFTWARE ); // Force software mixing
+ result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
+ if ( FAILED( result ) ) {
+ output->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating secondary buffer (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ }
+
+ // Get the buffer size ... might be different from what we specified.
+ DSBCAPS dsbcaps;
+ dsbcaps.dwSize = sizeof( DSBCAPS );
+ result = buffer->GetCaps( &dsbcaps );
+ if ( FAILED( result ) ) {
+ output->Release();
+ buffer->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ dsBufferSize = dsbcaps.dwBufferBytes;
+
+ // Lock the DS buffer
+ LPVOID audioPtr;
+ DWORD dataLen;
+ result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );
+ if ( FAILED( result ) ) {
+ output->Release();
+ buffer->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking buffer (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Zero the DS buffer
+ ZeroMemory( audioPtr, dataLen );
+
+ // Unlock the DS buffer
+ result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
+ if ( FAILED( result ) ) {
+ output->Release();
+ buffer->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking buffer (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ ohandle = (void *) output;
+ bhandle = (void *) buffer;
+ }
+
+ if ( mode == INPUT ) {
+
+ LPDIRECTSOUNDCAPTURE input;
+ result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ DSCCAPS inCaps;
+ inCaps.dwSize = sizeof( inCaps );
+ result = input->GetCaps( &inCaps );
+ if ( FAILED( result ) ) {
+ input->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Check channel information.
+ if ( inCaps.dwChannels < channels + firstChannel ) {
+ errorText_ = "RtApiDs::getDeviceInfo: the input device does not support requested input channels.";
+ return FAILURE;
+ }
+
+ // Check format information. Use 16-bit format unless user
+ // requests 8-bit.
+ DWORD deviceFormats;
+ if ( channels + firstChannel == 2 ) {
+ deviceFormats = WAVE_FORMAT_1S08 | WAVE_FORMAT_2S08 | WAVE_FORMAT_4S08 | WAVE_FORMAT_96S08;
+ if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
+ waveFormat.wBitsPerSample = 8;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT8;
+ }
+ else { // assume 16-bit is supported
+ waveFormat.wBitsPerSample = 16;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ }
+ }
+ else { // channel == 1
+ deviceFormats = WAVE_FORMAT_1M08 | WAVE_FORMAT_2M08 | WAVE_FORMAT_4M08 | WAVE_FORMAT_96M08;
+ if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
+ waveFormat.wBitsPerSample = 8;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT8;
+ }
+ else { // assume 16-bit is supported
+ waveFormat.wBitsPerSample = 16;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ }
+ }
+ stream_.userFormat = format;
+
+ // Update wave format structure and buffer information.
+ waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
+ waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
+ dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
+
+ // If the user wants an even bigger buffer, increase the device buffer size accordingly.
+ while ( dsPointerLeadTime * 2U > dsBufferSize )
+ dsBufferSize *= 2;
+
+ // Setup the secondary DS buffer description.
+ DSCBUFFERDESC bufferDescription;
+ ZeroMemory( &bufferDescription, sizeof( DSCBUFFERDESC ) );
+ bufferDescription.dwSize = sizeof( DSCBUFFERDESC );
+ bufferDescription.dwFlags = 0;
+ bufferDescription.dwReserved = 0;
+ bufferDescription.dwBufferBytes = dsBufferSize;
+ bufferDescription.lpwfxFormat = &waveFormat;
+
+ // Create the capture buffer.
+ LPDIRECTSOUNDCAPTUREBUFFER buffer;
+ result = input->CreateCaptureBuffer( &bufferDescription, &buffer, NULL );
+ if ( FAILED( result ) ) {
+ input->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating input buffer (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Get the buffer size ... might be different from what we specified.
+ DSCBCAPS dscbcaps;
+ dscbcaps.dwSize = sizeof( DSCBCAPS );
+ result = buffer->GetCaps( &dscbcaps );
+ if ( FAILED( result ) ) {
+ input->Release();
+ buffer->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ dsBufferSize = dscbcaps.dwBufferBytes;
+
+ // NOTE: We could have a problem here if this is a duplex stream
+ // and the play and capture hardware buffer sizes are different
+ // (I'm actually not sure if that is a problem or not).
+ // Currently, we are not verifying that.
+
+ // Lock the capture buffer
+ LPVOID audioPtr;
+ DWORD dataLen;
+ result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );
+ if ( FAILED( result ) ) {
+ input->Release();
+ buffer->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking input buffer (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Zero the buffer
+ ZeroMemory( audioPtr, dataLen );
+
+ // Unlock the buffer
+ result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
+ if ( FAILED( result ) ) {
+ input->Release();
+ buffer->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking input buffer (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ ohandle = (void *) input;
+ bhandle = (void *) buffer;
+ }
+
+ // Set various stream parameters
+ DsHandle *handle = 0;
+ stream_.nDeviceChannels[mode] = channels + firstChannel;
+ stream_.nUserChannels[mode] = channels;
+ stream_.bufferSize = *bufferSize;
+ stream_.channelOffset[mode] = firstChannel;
+ stream_.deviceInterleaved[mode] = true;
+ if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
+ else stream_.userInterleaved = true;
+
+ // Set flag for buffer conversion
+ stream_.doConvertBuffer[mode] = false;
+ if (stream_.nUserChannels[mode] != stream_.nDeviceChannels[mode])
+ stream_.doConvertBuffer[mode] = true;
+ if (stream_.userFormat != stream_.deviceFormat[mode])
+ stream_.doConvertBuffer[mode] = true;
+ if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+ stream_.nUserChannels[mode] > 1 )
+ stream_.doConvertBuffer[mode] = true;
+
+ // Allocate necessary internal buffers
+ long bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+ stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.userBuffer[mode] == NULL ) {
+ errorText_ = "RtApiDs::probeDeviceOpen: error allocating user buffer memory.";
+ goto error;
+ }
+
+ if ( stream_.doConvertBuffer[mode] ) {
+
+ bool makeBuffer = true;
+ bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
+ if ( mode == INPUT ) {
+ if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
+ unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
+ if ( bufferBytes <= (long) bytesOut ) makeBuffer = false;
+ }
+ }
+
+ if ( makeBuffer ) {
+ bufferBytes *= *bufferSize;
+ if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
+ stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.deviceBuffer == NULL ) {
+ errorText_ = "RtApiDs::probeDeviceOpen: error allocating device buffer memory.";
+ goto error;
+ }
+ }
+ }
+
+ // Allocate our DsHandle structures for the stream.
+ if ( stream_.apiHandle == 0 ) {
+ try {
+ handle = new DsHandle;
+ }
+ catch ( std::bad_alloc& ) {
+ errorText_ = "RtApiDs::probeDeviceOpen: error allocating AsioHandle memory.";
+ goto error;
+ }
+
+ // Create a manual-reset event.
+ handle->condition = CreateEvent( NULL, // no security
+ TRUE, // manual-reset
+ FALSE, // non-signaled initially
+ NULL ); // unnamed
+ stream_.apiHandle = (void *) handle;
+ }
+ else
+ handle = (DsHandle *) stream_.apiHandle;
+ handle->id[mode] = ohandle;
+ handle->buffer[mode] = bhandle;
+ handle->dsBufferSize[mode] = dsBufferSize;
+ handle->dsPointerLeadTime[mode] = dsPointerLeadTime;
+
+ stream_.device[mode] = device;
+ stream_.state = STREAM_STOPPED;
+ if ( stream_.mode == OUTPUT && mode == INPUT )
+ // We had already set up an output stream.
+ stream_.mode = DUPLEX;
+ else
+ stream_.mode = mode;
+ stream_.nBuffers = nBuffers;
+ stream_.sampleRate = sampleRate;
+
+ // Setup the buffer conversion information structure.
+ if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
+
+ // Setup the callback thread.
+ if ( stream_.callbackInfo.isRunning == false ) {
+ unsigned threadId;
+ stream_.callbackInfo.isRunning = true;
+ stream_.callbackInfo.object = (void *) this;
+ stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &callbackHandler,
+ &stream_.callbackInfo, 0, &threadId );
+ if ( stream_.callbackInfo.thread == 0 ) {
+ errorText_ = "RtApiDs::probeDeviceOpen: error creating callback thread!";
+ goto error;
+ }
+
+ // Boost DS thread priority
+ SetThreadPriority( (HANDLE) stream_.callbackInfo.thread, THREAD_PRIORITY_HIGHEST );
+ }
+ return SUCCESS;
+
+ error:
+ if ( handle ) {
+ if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
+ LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
+ LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+ if ( buffer ) buffer->Release();
+ object->Release();
+ }
+ if ( handle->buffer[1] ) {
+ LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
+ LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+ if ( buffer ) buffer->Release();
+ object->Release();
+ }
+ CloseHandle( handle->condition );
+ delete handle;
+ stream_.apiHandle = 0;
+ }
+
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
+ }
+
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
+
+ stream_.state = STREAM_CLOSED;
+ return FAILURE;
+}
+
+void RtApiDs :: closeStream()
+{
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiDs::closeStream(): no open stream to close!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ // Stop the callback thread.
+ stream_.callbackInfo.isRunning = false;
+ WaitForSingleObject( (HANDLE) stream_.callbackInfo.thread, INFINITE );
+ CloseHandle( (HANDLE) stream_.callbackInfo.thread );
+
+ DsHandle *handle = (DsHandle *) stream_.apiHandle;
+ if ( handle ) {
+ if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
+ LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
+ LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+ if ( buffer ) {
+ buffer->Stop();
+ buffer->Release();
+ }
+ object->Release();
+ }
+ if ( handle->buffer[1] ) {
+ LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
+ LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+ if ( buffer ) {
+ buffer->Stop();
+ buffer->Release();
+ }
+ object->Release();
+ }
+ CloseHandle( handle->condition );
+ delete handle;
+ stream_.apiHandle = 0;
+ }
+
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
+ }
+
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
+
+ stream_.mode = UNINITIALIZED;
+ stream_.state = STREAM_CLOSED;
+}
+
+void RtApiDs :: startStream()
+{
+ verifyStream();
+ if ( stream_.state == STREAM_RUNNING ) {
+ errorText_ = "RtApiDs::startStream(): the stream is already running!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ DsHandle *handle = (DsHandle *) stream_.apiHandle;
+
+ // Increase scheduler frequency on lesser windows (a side-effect of
+ // increasing timer accuracy). On greater windows (Win2K or later),
+ // this is already in effect.
+ timeBeginPeriod( 1 );
+
+ buffersRolling = false;
+ duplexPrerollBytes = 0;
+
+ if ( stream_.mode == DUPLEX ) {
+ // 0.5 seconds of silence in DUPLEX mode while the devices spin up and synchronize.
+ duplexPrerollBytes = (int) ( 0.5 * stream_.sampleRate * formatBytes( stream_.deviceFormat[1] ) * stream_.nDeviceChannels[1] );
+ }
+
+ HRESULT result = 0;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+
+ LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+ result = buffer->Play( 0, 0, DSBPLAY_LOOPING );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting output buffer!";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+ }
+
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+
+ LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+ result = buffer->Start( DSCBSTART_LOOPING );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting input buffer!";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+ }
+
+ handle->drainCounter = 0;
+ handle->internalDrain = false;
+ ResetEvent( handle->condition );
+ stream_.state = STREAM_RUNNING;
+
+ unlock:
+ if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );
+}
+
+void RtApiDs :: stopStream()
+{
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiDs::stopStream(): the stream is already stopped!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ HRESULT result = 0;
+ LPVOID audioPtr;
+ DWORD dataLen;
+ DsHandle *handle = (DsHandle *) stream_.apiHandle;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ if ( handle->drainCounter == 0 ) {
+ handle->drainCounter = 2;
+ WaitForSingleObject( handle->condition, INFINITE ); // block until signaled
+ }
+
+ stream_.state = STREAM_STOPPED;
+
+ MUTEX_LOCK( &stream_.mutex );
+
+ // Stop the buffer and clear memory
+ LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+ result = buffer->Stop();
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping output buffer!";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+
+ // Lock the buffer and clear it so that if we start to play again,
+ // we won't have old data playing.
+ result = buffer->Lock( 0, handle->dsBufferSize[0], &audioPtr, &dataLen, NULL, NULL, 0 );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking output buffer!";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+
+ // Zero the DS buffer
+ ZeroMemory( audioPtr, dataLen );
+
+ // Unlock the DS buffer
+ result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking output buffer!";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+
+ // If we start playing again, we must begin at beginning of buffer.
+ handle->bufferPointer[0] = 0;
+ }
+
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+ LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+ audioPtr = NULL;
+ dataLen = 0;
+
+ stream_.state = STREAM_STOPPED;
+
+ if ( stream_.mode != DUPLEX )
+ MUTEX_LOCK( &stream_.mutex );
+
+ result = buffer->Stop();
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping input buffer!";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+
+ // Lock the buffer and clear it so that if we start to play again,
+ // we won't have old data playing.
+ result = buffer->Lock( 0, handle->dsBufferSize[1], &audioPtr, &dataLen, NULL, NULL, 0 );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking input buffer!";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+
+ // Zero the DS buffer
+ ZeroMemory( audioPtr, dataLen );
+
+ // Unlock the DS buffer
+ result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking input buffer!";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+
+ // If we start recording again, we must begin at beginning of buffer.
+ handle->bufferPointer[1] = 0;
+ }
+
+ unlock:
+ timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows.
+ MUTEX_UNLOCK( &stream_.mutex );
+
+ if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );
+}
+
+void RtApiDs :: abortStream()
+{
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiDs::abortStream(): the stream is already stopped!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ DsHandle *handle = (DsHandle *) stream_.apiHandle;
+ handle->drainCounter = 2;
+
+ stopStream();
+}
+
+void RtApiDs :: callbackEvent()
+{
+ if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) {
+ Sleep( 50 ); // sleep 50 milliseconds
+ return;
+ }
+
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
+ DsHandle *handle = (DsHandle *) stream_.apiHandle;
+
+ // Check if we were draining the stream and signal is finished.
+ if ( handle->drainCounter > stream_.nBuffers + 2 ) {
+
+ stream_.state = STREAM_STOPPING;
+ if ( handle->internalDrain == false )
+ SetEvent( handle->condition );
+ else
+ stopStream();
+ return;
+ }
+
+ // Invoke user callback to get fresh output data UNLESS we are
+ // draining stream.
+ if ( handle->drainCounter == 0 ) {
+ RtAudioCallback callback = (RtAudioCallback) info->callback;
+ double streamTime = getStreamTime();
+ RtAudioStreamStatus status = 0;
+ if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
+ status |= RTAUDIO_OUTPUT_UNDERFLOW;
+ handle->xrun[0] = false;
+ }
+ if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
+ status |= RTAUDIO_INPUT_OVERFLOW;
+ handle->xrun[1] = false;
+ }
+ int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+ stream_.bufferSize, streamTime, status, info->userData );
+ if ( cbReturnValue == 2 ) {
+ stream_.state = STREAM_STOPPING;
+ handle->drainCounter = 2;
+ abortStream();
+ return;
+ }
+ else if ( cbReturnValue == 1 ) {
+ handle->drainCounter = 1;
+ handle->internalDrain = true;
+ }
+ }
+
+ HRESULT result;
+ DWORD currentWritePointer, safeWritePointer;
+ DWORD currentReadPointer, safeReadPointer;
+ UINT nextWritePointer;
+
+ LPVOID buffer1 = NULL;
+ LPVOID buffer2 = NULL;
+ DWORD bufferSize1 = 0;
+ DWORD bufferSize2 = 0;
+
+ char *buffer;
+ long bufferBytes;
+
+ MUTEX_LOCK( &stream_.mutex );
+ if ( stream_.state == STREAM_STOPPED ) {
+ MUTEX_UNLOCK( &stream_.mutex );
+ return;
+ }
+
+ if ( buffersRolling == false ) {
+ if ( stream_.mode == DUPLEX ) {
+ //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
+
+ // It takes a while for the devices to get rolling. As a result,
+ // there's no guarantee that the capture and write device pointers
+ // will move in lockstep. Wait here for both devices to start
+ // rolling, and then set our buffer pointers accordingly.
+ // e.g. Crystal Drivers: the capture buffer starts up 5700 to 9600
+ // bytes later than the write buffer.
+
+ // Stub: a serious risk of having a pre-emptive scheduling round
+ // take place between the two GetCurrentPosition calls... but I'm
+ // really not sure how to solve the problem. Temporarily boost to
+ // Realtime priority, maybe; but I'm not sure what priority the
+ // DirectSound service threads run at. We *should* be roughly
+ // within a ms or so of correct.
+
+ LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+ LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+
+ DWORD startSafeWritePointer, startSafeReadPointer;
+
+ result = dsWriteBuffer->GetCurrentPosition( NULL, &startSafeWritePointer );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+ result = dsCaptureBuffer->GetCurrentPosition( NULL, &startSafeReadPointer );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+ while ( true ) {
+ result = dsWriteBuffer->GetCurrentPosition( NULL, &safeWritePointer );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+ result = dsCaptureBuffer->GetCurrentPosition( NULL, &safeReadPointer );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+ if ( safeWritePointer != startSafeWritePointer && safeReadPointer != startSafeReadPointer ) break;
+ Sleep( 1 );
+ }
+
+ //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
+
+ handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];
+ if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];
+ handle->bufferPointer[1] = safeReadPointer;
+ }
+ else if ( stream_.mode == OUTPUT ) {
+
+ // Set the proper nextWritePosition after initial startup.
+ LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+ result = dsWriteBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+ handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];
+ if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];
+ }
+
+ buffersRolling = true;
+ }
+
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+
+ LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+
+ if ( handle->drainCounter > 1 ) { // write zeros to the output stream
+ bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
+ bufferBytes *= formatBytes( stream_.userFormat );
+ memset( stream_.userBuffer[0], 0, bufferBytes );
+ }
+
+ // Setup parameters and do buffer conversion if necessary.
+ if ( stream_.doConvertBuffer[0] ) {
+ buffer = stream_.deviceBuffer;
+ convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
+ bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[0];
+ bufferBytes *= formatBytes( stream_.deviceFormat[0] );
+ }
+ else {
+ buffer = stream_.userBuffer[0];
+ bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
+ bufferBytes *= formatBytes( stream_.userFormat );
+ }
+
+ // No byte swapping necessary in DirectSound implementation.
+
+ // Ahhh ... windoze. 16-bit data is signed but 8-bit data is
+ // unsigned. So, we need to convert our signed 8-bit data here to
+ // unsigned.
+ if ( stream_.deviceFormat[0] == RTAUDIO_SINT8 )
+ for ( int i=0; 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( ¤tWritePointer, &safeWritePointer );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+
+ // We will copy our output buffer into the region between
+ // safeWritePointer and leadPointer. If leadPointer is not
+ // beyond the next endWrite position, wait until it is.
+ leadPointer = safeWritePointer + handle->dsPointerLeadTime[0];
+ //std::cout << "safeWritePointer = " << safeWritePointer << ", leadPointer = " << leadPointer << ", nextWritePointer = " << nextWritePointer << std::endl;
+ if ( leadPointer > dsBufferSize ) leadPointer -= dsBufferSize;
+ if ( leadPointer < nextWritePointer ) leadPointer += dsBufferSize; // unwrap offset
+ endWrite = nextWritePointer + bufferBytes;
+
+ // Check whether the entire write region is behind the play pointer.
+ if ( leadPointer >= endWrite ) break;
+
+ // If we are here, then we must wait until the leadPointer advances
+ // beyond the end of our next write region. We use the
+ // Sleep() function to suspend operation until that happens.
+ double millis = ( endWrite - leadPointer ) * 1000.0;
+ millis /= ( formatBytes( stream_.deviceFormat[0]) * stream_.nDeviceChannels[0] * stream_.sampleRate);
+ if ( millis < 1.0 ) millis = 1.0;
+ Sleep( (DWORD) millis );
+ }
+
+ if ( dsPointerBetween( nextWritePointer, safeWritePointer, currentWritePointer, dsBufferSize )
+ || dsPointerBetween( endWrite, safeWritePointer, currentWritePointer, dsBufferSize ) ) {
+ // We've strayed into the forbidden zone ... resync the read pointer.
+ handle->xrun[0] = true;
+ nextWritePointer = safeWritePointer + handle->dsPointerLeadTime[0] - bufferBytes;
+ if ( nextWritePointer >= dsBufferSize ) nextWritePointer -= dsBufferSize;
+ handle->bufferPointer[0] = nextWritePointer;
+ endWrite = nextWritePointer + bufferBytes;
+ }
+
+ // Lock free space in the buffer
+ result = dsBuffer->Lock( nextWritePointer, bufferBytes, &buffer1,
+ &bufferSize1, &buffer2, &bufferSize2, 0 );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking buffer during playback!";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+
+ // Copy our buffer into the DS buffer
+ CopyMemory( buffer1, buffer, bufferSize1 );
+ if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 );
+
+ // Update our buffer offset and unlock sound buffer
+ dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+ nextWritePointer = ( nextWritePointer + bufferSize1 + bufferSize2 ) % dsBufferSize;
+ handle->bufferPointer[0] = nextWritePointer;
+ }
+
+ // Don't bother draining input
+ if ( handle->drainCounter ) {
+ handle->drainCounter++;
+ goto unlock;
+ }
+
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+
+ // Setup parameters.
+ if ( stream_.doConvertBuffer[1] ) {
+ buffer = stream_.deviceBuffer;
+ bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[1];
+ bufferBytes *= formatBytes( stream_.deviceFormat[1] );
+ }
+ else {
+ buffer = stream_.userBuffer[1];
+ bufferBytes = stream_.bufferSize * stream_.nUserChannels[1];
+ bufferBytes *= formatBytes( stream_.userFormat );
+ }
+
+ LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+ long nextReadPointer = handle->bufferPointer[1];
+ DWORD dsBufferSize = handle->dsBufferSize[1];
+
+ // Find out where the write and "safe read" pointers are.
+ result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+
+ if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset
+ DWORD endRead = nextReadPointer + bufferBytes;
+
+ // Handling depends on whether we are INPUT or DUPLEX.
+ // If we're in INPUT mode then waiting is a good thing. If we're in DUPLEX mode,
+ // then a wait here will drag the write pointers into the forbidden zone.
+ //
+ // In DUPLEX mode, rather than wait, we will back off the read pointer until
+ // it's in a safe position. This causes dropouts, but it seems to be the only
+ // practical way to sync up the read and write pointers reliably, given the
+ // the very complex relationship between phase and increment of the read and write
+ // pointers.
+ //
+ // In order to minimize audible dropouts in DUPLEX mode, we will
+ // provide a pre-roll period of 0.5 seconds in which we return
+ // zeros from the read buffer while the pointers sync up.
+
+ if ( stream_.mode == DUPLEX ) {
+ if ( safeReadPointer < endRead ) {
+ if ( duplexPrerollBytes <= 0 ) {
+ // Pre-roll time over. Be more agressive.
+ int adjustment = endRead-safeReadPointer;
+
+ handle->xrun[1] = true;
+ // Two cases:
+ // - large adjustments: we've probably run out of CPU cycles, so just resync exactly,
+ // and perform fine adjustments later.
+ // - small adjustments: back off by twice as much.
+ if ( adjustment >= 2*bufferBytes )
+ nextReadPointer = safeReadPointer-2*bufferBytes;
+ else
+ nextReadPointer = safeReadPointer-bufferBytes-adjustment;
+
+ if ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;
+
+ }
+ else {
+ // In pre=roll time. Just do it.
+ nextReadPointer = safeReadPointer - bufferBytes;
+ while ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;
+ }
+ endRead = nextReadPointer + bufferBytes;
+ }
+ }
+ else { // mode == INPUT
+ while ( safeReadPointer < endRead && stream_.callbackInfo.isRunning ) {
+ // See comments for playback.
+ double millis = (endRead - safeReadPointer) * 1000.0;
+ millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate);
+ if ( millis < 1.0 ) millis = 1.0;
+ Sleep( (DWORD) millis );
+
+ // Wake up and find out where we are now.
+ result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+
+ if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset
+ }
+ }
+
+ // Lock free space in the buffer
+ result = dsBuffer->Lock( nextReadPointer, bufferBytes, &buffer1,
+ &bufferSize1, &buffer2, &bufferSize2, 0 );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking capture buffer!";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+
+ if ( duplexPrerollBytes <= 0 ) {
+ // Copy our buffer into the DS buffer
+ CopyMemory( buffer, buffer1, bufferSize1 );
+ if ( buffer2 != NULL ) CopyMemory( buffer+bufferSize1, buffer2, bufferSize2 );
+ }
+ else {
+ memset( buffer, 0, bufferSize1 );
+ if ( buffer2 != NULL ) memset( buffer + bufferSize1, 0, bufferSize2 );
+ duplexPrerollBytes -= bufferSize1 + bufferSize2;
+ }
+
+ // Update our buffer offset and unlock sound buffer
+ nextReadPointer = ( nextReadPointer + bufferSize1 + bufferSize2 ) % dsBufferSize;
+ dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking capture buffer!";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+ handle->bufferPointer[1] = nextReadPointer;
+
+ // No byte swapping necessary in DirectSound implementation.
+
+ // If necessary, convert 8-bit data from unsigned to signed.
+ if ( stream_.deviceFormat[1] == RTAUDIO_SINT8 )
+ for ( int j=0; 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_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()
+ :synchronized(false), runnable(false) { xrun[0] = false; xrun[1] = false; }
+};
+
+static void *alsaCallbackHandler( void * ptr );
+
+RtApiAlsa :: RtApiAlsa()
+{
+ // Nothing to do here.
+}
+
+RtApiAlsa :: ~RtApiAlsa()
+{
+ if ( stream_.state != STREAM_CLOSED ) closeStream();
+}
+
+unsigned int RtApiAlsa :: getDeviceCount( void )
+{
+ unsigned nDevices = 0;
+ int result, subdevice, card;
+ char name[64];
+ snd_ctl_t *handle;
+
+ // Count cards and devices
+ card = -1;
+ snd_card_next( &card );
+ while ( card >= 0 ) {
+ sprintf( name, "hw:%d", card );
+ result = snd_ctl_open( &handle, name, 0 );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ goto nextcard;
+ }
+ subdevice = -1;
+ while( 1 ) {
+ result = snd_ctl_pcm_next_device( handle, &subdevice );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ break;
+ }
+ if ( subdevice < 0 )
+ break;
+ nDevices++;
+ }
+ nextcard:
+ snd_ctl_close( handle );
+ snd_card_next( &card );
+ }
+
+ result = snd_ctl_open( &handle, "default", 0 );
+ if (result == 0) {
+ nDevices++;
+ snd_ctl_close( handle );
+ }
+
+ return nDevices;
+}
+
+RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device )
+{
+ RtAudio::DeviceInfo info;
+ info.probed = false;
+
+ unsigned nDevices = 0;
+ int result, subdevice, card;
+ char name[64];
+ snd_ctl_t *chandle;
+
+ // Count cards and devices
+ card = -1;
+ subdevice = -1;
+ snd_card_next( &card );
+ while ( card >= 0 ) {
+ sprintf( name, "hw:%d", card );
+ result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ goto nextcard;
+ }
+ subdevice = -1;
+ while( 1 ) {
+ result = snd_ctl_pcm_next_device( chandle, &subdevice );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ break;
+ }
+ if ( subdevice < 0 ) break;
+ if ( nDevices == device ) {
+ sprintf( name, "hw:%d,%d", card, subdevice );
+ goto foundDevice;
+ }
+ nDevices++;
+ }
+ nextcard:
+ snd_ctl_close( chandle );
+ snd_card_next( &card );
+ }
+
+ result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
+ if ( result == 0 ) {
+ if ( nDevices == device ) {
+ strcpy( name, "default" );
+ goto foundDevice;
+ }
+ nDevices++;
+ }
+
+ if ( nDevices == 0 ) {
+ errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!";
+ error( RtAudioError::INVALID_USE );
+ return info;
+ }
+
+ if ( device >= nDevices ) {
+ errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!";
+ error( RtAudioError::INVALID_USE );
+ return info;
+ }
+
+ foundDevice:
+
+ // If a stream is already open, we cannot probe the stream devices.
+ // Thus, use the saved results.
+ if ( stream_.state != STREAM_CLOSED &&
+ ( stream_.device[0] == device || stream_.device[1] == device ) ) {
+ snd_ctl_close( chandle );
+ if ( device >= devices_.size() ) {
+ errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened.";
+ error( RtAudioError::WARNING );
+ return info;
+ }
+ return devices_[ device ];
+ }
+
+ int openMode = SND_PCM_ASYNC;
+ snd_pcm_stream_t stream;
+ snd_pcm_info_t *pcminfo;
+ snd_pcm_info_alloca( &pcminfo );
+ snd_pcm_t *phandle;
+ snd_pcm_hw_params_t *params;
+ snd_pcm_hw_params_alloca( ¶ms );
+
+ // First try for playback unless default device (which has subdev -1)
+ stream = SND_PCM_STREAM_PLAYBACK;
+ snd_pcm_info_set_stream( pcminfo, stream );
+ if ( subdevice != -1 ) {
+ snd_pcm_info_set_device( pcminfo, subdevice );
+ snd_pcm_info_set_subdevice( pcminfo, 0 );
+
+ result = snd_ctl_pcm_info( chandle, pcminfo );
+ if ( result < 0 ) {
+ // Device probably doesn't support playback.
+ goto captureProbe;
+ }
+ }
+
+ result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ goto captureProbe;
+ }
+
+ // The device is open ... fill the parameter structure.
+ result = snd_pcm_hw_params_any( phandle, params );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ goto captureProbe;
+ }
+
+ // Get output channel information.
+ unsigned int value;
+ result = snd_pcm_hw_params_get_channels_max( params, &value );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ goto captureProbe;
+ }
+ info.outputChannels = value;
+ snd_pcm_close( phandle );
+
+ captureProbe:
+ stream = SND_PCM_STREAM_CAPTURE;
+ snd_pcm_info_set_stream( pcminfo, stream );
+
+ // Now try for capture unless default device (with subdev = -1)
+ if ( subdevice != -1 ) {
+ result = snd_ctl_pcm_info( chandle, pcminfo );
+ snd_ctl_close( chandle );
+ if ( result < 0 ) {
+ // Device probably doesn't support capture.
+ if ( info.outputChannels == 0 ) return info;
+ goto probeParameters;
+ }
+ }
+ else
+ snd_ctl_close( chandle );
+
+ result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ if ( info.outputChannels == 0 ) return info;
+ goto probeParameters;
+ }
+
+ // The device is open ... fill the parameter structure.
+ result = snd_pcm_hw_params_any( phandle, params );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ if ( info.outputChannels == 0 ) return info;
+ goto probeParameters;
+ }
+
+ result = snd_pcm_hw_params_get_channels_max( params, &value );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ if ( info.outputChannels == 0 ) return info;
+ goto probeParameters;
+ }
+ info.inputChannels = value;
+ snd_pcm_close( phandle );
+
+ // If device opens for both playback and capture, we determine the channels.
+ if ( info.outputChannels > 0 && info.inputChannels > 0 )
+ info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
+
+ // ALSA doesn't provide default devices so we'll use the first available one.
+ if ( device == 0 && info.outputChannels > 0 )
+ info.isDefaultOutput = true;
+ if ( device == 0 && info.inputChannels > 0 )
+ info.isDefaultInput = true;
+
+ probeParameters:
+ // At this point, we just need to figure out the supported data
+ // formats and sample rates. We'll proceed by opening the device in
+ // the direction with the maximum number of channels, or playback if
+ // they are equal. This might limit our sample rate options, but so
+ // be it.
+
+ if ( info.outputChannels >= info.inputChannels )
+ stream = SND_PCM_STREAM_PLAYBACK;
+ else
+ stream = SND_PCM_STREAM_CAPTURE;
+ snd_pcm_info_set_stream( pcminfo, stream );
+
+ result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ // The device is open ... fill the parameter structure.
+ result = snd_pcm_hw_params_any( phandle, params );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ // Test our discrete set of sample rate values.
+ info.sampleRates.clear();
+ for ( unsigned int i=0; i<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;
- case DSERR_OTHERAPPHASPRIO:
- return "Another application already has priority";
+ // 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;
+ }
- case DSERR_UNINITIALIZED:
- return "Uninitialized";
+ // Get the device name
+ char *cardname;
+ result = snd_card_get_name( card, &cardname );
+ if ( result >= 0 ) {
+ sprintf( name, "hw:%s,%d", cardname, subdevice );
+ free( cardname );
+ }
+ info.name = name;
- default:
- return "DirectSound unknown error";
- }
+ // That's all ... close the device and return
+ snd_pcm_close( phandle );
+ info.probed = true;
+ return info;
}
-//******************** 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];
+void RtApiAlsa :: saveDeviceInfo( void )
+{
+ devices_.clear();
- AlsaHandle()
- :synchronized(false) { xrun[0] = false; xrun[1] = false; }
-};
+ unsigned int nDevices = getDeviceCount();
+ devices_.resize( nDevices );
+ for ( unsigned int i=0; i<nDevices; i++ )
+ devices_[i] = getDeviceInfo( i );
+}
-extern "C" void *alsaCallbackHandler( void * ptr );
+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 )
-RtApiAlsa :: RtApiAlsa()
{
- // Nothing to do here.
-}
+#if defined(__RTAUDIO_DEBUG__)
+ snd_output_t *out;
+ snd_output_stdio_attach(&out, stderr, 0);
+#endif
-RtApiAlsa :: ~RtApiAlsa()
-{
- if ( stream_.state != STREAM_CLOSED ) closeStream();
-}
+ // I'm not using the "plug" interface ... too much inconsistent behavior.
-unsigned int RtApiAlsa :: getDeviceCount( void )
-{
unsigned nDevices = 0;
int result, subdevice, card;
char name[64];
- snd_ctl_t *handle;
+ snd_ctl_t *chandle;
- // Count cards and devices
- card = -1;
- snd_card_next( &card );
- while ( card >= 0 ) {
- sprintf( name, "hw:%d", card );
- result = snd_ctl_open( &handle, name, 0 );
- if ( result < 0 ) {
- errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- error( RtError::WARNING );
- goto nextcard;
- }
- subdevice = -1;
- while( 1 ) {
- result = snd_ctl_pcm_next_device( handle, &subdevice );
- if ( result < 0 ) {
- errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
+ if ( options && options->flags & RTAUDIO_ALSA_USE_DEFAULT )
+ snprintf(name, sizeof(name), "%s", "default");
+ else {
+ // Count cards and devices
+ card = -1;
+ snd_card_next( &card );
+ while ( card >= 0 ) {
+ sprintf( name, "hw:%d", card );
+ result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: control open, card = " << card << ", " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
- error( RtError::WARNING );
- break;
+ return FAILURE;
+ }
+ subdevice = -1;
+ while( 1 ) {
+ result = snd_ctl_pcm_next_device( chandle, &subdevice );
+ if ( result < 0 ) break;
+ if ( subdevice < 0 ) break;
+ if ( nDevices == device ) {
+ sprintf( name, "hw:%d,%d", card, subdevice );
+ snd_ctl_close( chandle );
+ goto foundDevice;
+ }
+ nDevices++;
+ }
+ snd_ctl_close( chandle );
+ snd_card_next( &card );
+ }
+
+ result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
+ if ( result == 0 ) {
+ if ( nDevices == device ) {
+ strcpy( name, "default" );
+ goto foundDevice;
}
- if ( subdevice < 0 )
- break;
nDevices++;
}
- nextcard:
- snd_ctl_close( handle );
- 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;
+ }
}
- return nDevices;
-}
+ foundDevice:
-RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device )
-{
- RtAudio::DeviceInfo info;
- info.probed = false;
+ // 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();
- unsigned nDevices = 0;
- int result, subdevice, card;
- char name[64];
- snd_ctl_t *chandle;
+ snd_pcm_stream_t stream;
+ if ( mode == OUTPUT )
+ stream = SND_PCM_STREAM_PLAYBACK;
+ else
+ stream = SND_PCM_STREAM_CAPTURE;
- // 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 );
+ 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 ) {
- errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- error( RtError::WARNING );
- goto nextcard;
- }
- subdevice = -1;
- while( 1 ) {
- result = snd_ctl_pcm_next_device( chandle, &subdevice );
- if ( result < 0 ) {
- errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- error( RtError::WARNING );
- break;
- }
- if ( subdevice < 0 ) break;
- if ( nDevices == device ) {
- sprintf( name, "hw:%d,%d", card, subdevice );
- goto foundDevice;
- }
- nDevices++;
+ result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
+ stream_.deviceInterleaved[mode] = true;
}
- nextcard:
- snd_ctl_close( chandle );
- snd_card_next( &card );
+ 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 ( nDevices == 0 ) {
- errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!";
- error( RtError::INVALID_USE );
- }
+ // 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;
- if ( device >= nDevices ) {
- errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!";
- error( RtError::INVALID_USE );
+ 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;
}
- foundDevice:
+ // 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;
+ }
+ }
- int openMode = SND_PCM_ASYNC;
- snd_pcm_stream_t stream;
- snd_pcm_info_t *pcminfo;
- snd_pcm_info_alloca( &pcminfo );
- snd_pcm_t *phandle;
- snd_pcm_hw_params_t *params;
- snd_pcm_hw_params_alloca( ¶ms );
+ // 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;
+ }
- // First try for playback
- stream = SND_PCM_STREAM_PLAYBACK;
- snd_pcm_info_set_device( pcminfo, subdevice );
- snd_pcm_info_set_subdevice( pcminfo, 0 );
- snd_pcm_info_set_stream( pcminfo, stream );
+ // 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_ctl_pcm_info( chandle, pcminfo );
+ result = snd_pcm_hw_params_get_channels_min( hw_params, &value );
if ( result < 0 ) {
- // Device probably doesn't support playback.
- goto captureProbe;
+ 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;
- result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK );
+ // Set the device channels.
+ result = snd_pcm_hw_params_set_channels( phandle, hw_params, deviceChannels );
if ( result < 0 ) {
- errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting channels for device (" << name << "), " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
- error( RtError::WARNING );
- goto captureProbe;
+ return FAILURE;
}
- // The device is open ... fill the parameter structure.
- result = snd_pcm_hw_params_any( phandle, params );
+ // 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::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting period size for device (" << name << "), " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
- error( RtError::WARNING );
- goto captureProbe;
+ return FAILURE;
}
+ *bufferSize = periodSize;
- // Get output channel information.
- unsigned int value;
- result = snd_pcm_hw_params_get_channels_max( params, &value );
+ // 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::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << ".";
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
- error( RtError::WARNING );
- goto captureProbe;
+ return FAILURE;
}
- info.outputChannels = value;
- snd_pcm_close( phandle );
- captureProbe:
- // Now try for capture
- stream = SND_PCM_STREAM_CAPTURE;
- snd_pcm_info_set_stream( pcminfo, stream );
+ // 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;
- result = snd_ctl_pcm_info( chandle, pcminfo );
- snd_ctl_close( chandle );
+ // Install the hardware configuration
+ result = snd_pcm_hw_params( phandle, hw_params );
if ( result < 0 ) {
- // Device probably doesn't support capture.
- if ( info.outputChannels == 0 ) return info;
- goto probeParameters;
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing hardware configuration on device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
}
- result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
+#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 ) {
- errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing software configuration on device (" << name << "), " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
- error( RtError::WARNING );
- if ( info.outputChannels == 0 ) return info;
- goto probeParameters;
+ 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, ¶m);
+ }
+ 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;
+ }
+ }
}
- // The device is open ... fill the parameter structure.
- result = snd_pcm_hw_params_any( phandle, params );
- if ( result < 0 ) {
- snd_pcm_close( phandle );
- errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- error( RtError::WARNING );
- if ( info.outputChannels == 0 ) return info;
- goto probeParameters;
- }
+ return SUCCESS;
- result = snd_pcm_hw_params_get_channels_max( params, &value );
- if ( result < 0 ) {
- snd_pcm_close( phandle );
- errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- error( RtError::WARNING );
- if ( info.outputChannels == 0 ) return info;
- goto probeParameters;
+ 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;
}
- 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;
+ if ( phandle) snd_pcm_close( phandle );
- // 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;
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
+ }
- 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 ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
- if ( info.outputChannels >= info.inputChannels )
- stream = SND_PCM_STREAM_PLAYBACK;
- else
- stream = SND_PCM_STREAM_CAPTURE;
- snd_pcm_info_set_stream( pcminfo, stream );
+ stream_.state = STREAM_CLOSED;
+ return FAILURE;
+}
- result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
- if ( result < 0 ) {
- errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- error( RtError::WARNING );
- return info;
+void RtApiAlsa :: closeStream()
+{
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiAlsa::closeStream(): no open stream to close!";
+ error( RtAudioError::WARNING );
+ return;
}
- // The device is open ... fill the parameter structure.
- result = snd_pcm_hw_params_any( phandle, params );
- if ( result < 0 ) {
- snd_pcm_close( phandle );
- errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- error( RtError::WARNING );
- return info;
+ 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 );
- // 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.sampleRates.size() == 0 ) {
- snd_pcm_close( phandle );
- errorStream_ << "RtApiAlsa::getDeviceInfo: no supported sample rates found for device (" << name << ").";
- errorText_ = errorStream_.str();
- error( RtError::WARNING );
- return info;
+ 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] );
}
- // 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;
+ 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;
+ }
- // Check that we have at least one supported format
- if ( info.nativeFormats == 0 ) {
- errorStream_ << "RtApiAlsa::getDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio.";
- errorText_ = errorStream_.str();
- error( RtError::WARNING );
- return info;
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
}
- // Get the device name
- char *cardname;
- result = snd_card_get_name( card, &cardname );
- if ( result >= 0 )
- sprintf( name, "hw:%s,%d", cardname, subdevice );
- info.name = name;
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
- // That's all ... close the device and return
- snd_pcm_close( phandle );
- info.probed = true;
- return info;
+ stream_.mode = UNINITIALIZED;
+ stream_.state = STREAM_CLOSED;
}
-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 )
-
+void RtApiAlsa :: startStream()
{
-#if defined(__RTAUDIO_DEBUG__)
- snd_output_t *out;
- snd_output_stdio_attach(&out, stderr, 0);
-#endif
+ // This method calls snd_pcm_prepare if the device isn't already in that state.
- // I'm not using the "plug" interface ... too much inconsistent behavior.
+ verifyStream();
+ if ( stream_.state == STREAM_RUNNING ) {
+ errorText_ = "RtApiAlsa::startStream(): the stream is already running!";
+ error( RtAudioError::WARNING );
+ return;
+ }
- unsigned nDevices = 0;
- int result, subdevice, card;
- char name[64];
- snd_ctl_t *chandle;
+ MUTEX_LOCK( &stream_.mutex );
- // Count cards and devices
- card = -1;
- snd_card_next( &card );
- while ( card >= 0 ) {
- sprintf( name, "hw:%d", card );
- result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
- if ( result < 0 ) {
- errorStream_ << "RtApiAlsa::probeDeviceOpen: control open, card = " << card << ", " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
- subdevice = -1;
- while( 1 ) {
- result = snd_ctl_pcm_next_device( chandle, &subdevice );
- if ( result < 0 ) break;
- if ( subdevice < 0 ) break;
- if ( nDevices == device ) {
- sprintf( name, "hw:%d,%d", card, subdevice );
- goto foundDevice;
+ 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;
}
- 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 ( ( 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;
+ }
+ }
}
- 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;
- }
+ stream_.state = STREAM_RUNNING;
- foundDevice:
+ unlock:
+ apiInfo->runnable = true;
+ pthread_cond_signal( &apiInfo->runnable_cv );
+ MUTEX_UNLOCK( &stream_.mutex );
- snd_pcm_stream_t stream;
- if ( mode == OUTPUT )
- stream = SND_PCM_STREAM_PLAYBACK;
- else
- stream = SND_PCM_STREAM_CAPTURE;
+ if ( result >= 0 ) return;
+ error( RtAudioError::SYSTEM_ERROR );
+}
- 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.";
+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
- errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for input.";
- errorText_ = errorStream_.str();
- return FAILURE;
+ 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;
+ }
}
- // 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 ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
+ result = snd_pcm_drop( handle[1] );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::stopStream: error stopping input pcm device, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
}
-#if defined(__RTAUDIO_DEBUG__)
- fprintf( stderr, "\nRtApiAlsa: dump hardware params just after device open:\n\n" );
- snd_pcm_hw_params_dump( hw_params, out );
-#endif
+ unlock:
+ apiInfo->runnable = false; // fixes high CPU usage when stopped
+ MUTEX_UNLOCK( &stream_.mutex );
- // 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 ) 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 ) {
- result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
- stream_.deviceInterleaved[mode] = true;
+ errorStream_ << "RtApiAlsa::abortStream: error aborting output pcm device, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ goto unlock;
}
- 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 ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
+ result = snd_pcm_drop( handle[1] );
if ( result < 0 ) {
- result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
- stream_.deviceInterleaved[mode] = false;
+ errorStream_ << "RtApiAlsa::abortStream: error aborting input pcm device, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ goto unlock;
}
- 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;
- }
+ unlock:
+ apiInfo->runnable = false; // fixes high CPU usage when stopped
+ MUTEX_UNLOCK( &stream_.mutex );
- // Determine how to set the device format.
- stream_.userFormat = format;
- snd_pcm_format_t deviceFormat = SND_PCM_FORMAT_UNKNOWN;
+ if ( result >= 0 ) return;
+ error( RtAudioError::SYSTEM_ERROR );
+}
- 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;
+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 ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat) == 0) {
- stream_.deviceFormat[mode] = format;
- goto setFormat;
+ if ( stream_.state != STREAM_RUNNING ) {
+ MUTEX_UNLOCK( &stream_.mutex );
+ return;
+ }
+ MUTEX_UNLOCK( &stream_.mutex );
}
- // 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;
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!";
+ error( RtAudioError::WARNING );
+ return;
}
- deviceFormat = SND_PCM_FORMAT_FLOAT;
- if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
- stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
- goto setFormat;
+ 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;
}
-
- deviceFormat = SND_PCM_FORMAT_S32;
- if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
- stream_.deviceFormat[mode] = RTAUDIO_SINT32;
- goto setFormat;
+ 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 );
- deviceFormat = SND_PCM_FORMAT_S24;
- if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
- stream_.deviceFormat[mode] = RTAUDIO_SINT24;
- goto setFormat;
+ if ( doStopStream == 2 ) {
+ abortStream();
+ return;
}
- deviceFormat = SND_PCM_FORMAT_S16;
- if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
- stream_.deviceFormat[mode] = RTAUDIO_SINT16;
- goto setFormat;
- }
+ MUTEX_LOCK( &stream_.mutex );
- deviceFormat = SND_PCM_FORMAT_S8;
- if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
- stream_.deviceFormat[mode] = RTAUDIO_SINT8;
- goto setFormat;
- }
+ // The state might change while waiting on a mutex.
+ if ( stream_.state == STREAM_STOPPED ) goto unlock;
- // If we get here, no supported format was found.
- errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device " << device << " data format not supported by RtAudio.";
- errorText_ = errorStream_.str();
- return FAILURE;
+ int result;
+ char *buffer;
+ int channels;
+ snd_pcm_t **handle;
+ snd_pcm_sframes_t frames;
+ RtAudioFormat format;
+ handle = (snd_pcm_t **) apiInfo->handles;
- 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;
+ 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;
}
- // 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;
+ 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 );
}
- }
- // 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;
- }
+ 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;
+ }
- // 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;
+ // Check stream latency
+ result = snd_pcm_delay( handle[0], &frames );
+ if ( result == 0 && frames > 0 ) stream_.latency[0] = frames;
}
- 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;
+ unlock:
+ MUTEX_UNLOCK( &stream_.mutex );
- // 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;
- }
+ RtApi::tickStreamTime();
+ if ( doStopStream == 1 ) this->stopStream();
+}
- // Set the buffer number, which in ALSA is referred to as the "period".
- int dir;
- unsigned int periods = 0;
- if ( options ) periods = options->numberOfBuffers;
- if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2;
- // Even though the hardware might allow 1 buffer, it won't work reliably.
- if ( periods < 2 ) periods = 2;
- result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir );
- if ( result < 0 ) {
- snd_pcm_close( phandle );
- errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
+static void *alsaCallbackHandler( void *ptr )
+{
+ CallbackInfo *info = (CallbackInfo *) ptr;
+ RtApiAlsa *object = (RtApiAlsa *) info->object;
+ bool *isRunning = &info->isRunning;
- // Set the buffer (or period) size.
- snd_pcm_uframes_t periodSize = *bufferSize;
- result = snd_pcm_hw_params_set_period_size_near( phandle, hw_params, &periodSize, &dir );
- if ( result < 0 ) {
- snd_pcm_close( phandle );
- errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting period size for device (" << name << "), " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- return FAILURE;
+#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;
}
- *bufferSize = periodSize;
+#endif
- // If attempting to setup a duplex stream, the bufferSize parameter
- // MUST be the same in both directions!
- if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {
- errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ").";
- errorText_ = errorStream_.str();
- return FAILURE;
+ while ( *isRunning == true ) {
+ pthread_testcancel();
+ object->callbackEvent();
}
- 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;
- }
+ pthread_exit( NULL );
+}
-#if defined(__RTAUDIO_DEBUG__)
- fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n");
- snd_pcm_hw_params_dump( hw_params, out );
+//******************** End of __LINUX_ALSA__ *********************//
#endif
- // Set the software configuration to fill buffers with zeros and prevent device stopping on xruns.
- snd_pcm_sw_params_t *sw_params = NULL;
- snd_pcm_sw_params_alloca( &sw_params );
- snd_pcm_sw_params_current( phandle, sw_params );
- snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize );
- snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, 0x7fffffff );
- snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 );
- snd_pcm_sw_params_set_silence_size( phandle, sw_params, INT_MAX );
- result = snd_pcm_sw_params( phandle, sw_params );
- if ( result < 0 ) {
- snd_pcm_close( phandle );
- errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing software configuration on device (" << name << "), " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- return FAILURE;
- }
+#if defined(__LINUX_PULSE__)
-#if defined(__RTAUDIO_DEBUG__)
- fprintf(stderr, "\nRtApiAlsa: dump software params after installation:\n\n");
- snd_pcm_sw_params_dump( sw_params, out );
-#endif
+// Code written by Peter Meerwald, pmeerw@pmeerw.net
+// and Tristan Matthews.
- // 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;
+#include <pulse/error.h>
+#include <pulse/simple.h>
+#include <cstdio>
- // Allocate the ApiHandle if necessary and then save.
- AlsaHandle *apiInfo = 0;
- if ( stream_.apiHandle == 0 ) {
- try {
- apiInfo = (AlsaHandle *) new AlsaHandle;
- }
- catch ( std::bad_alloc& ) {
- errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating AlsaHandle memory.";
- goto error;
- }
- stream_.apiHandle = (void *) apiInfo;
- apiInfo->handles[0] = 0;
- apiInfo->handles[1] = 0;
- }
- else {
- apiInfo = (AlsaHandle *) stream_.apiHandle;
- }
- apiInfo->handles[mode] = phandle;
+static const unsigned int SUPPORTED_SAMPLERATES[] = { 8000, 16000, 22050, 32000,
+ 44100, 48000, 96000, 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;
- }
+struct rtaudio_pa_format_mapping_t {
+ RtAudioFormat rtaudio_format;
+ pa_sample_format_t pa_format;
+};
- if ( stream_.doConvertBuffer[mode] ) {
+static const rtaudio_pa_format_mapping_t supported_sampleformats[] = {
+ {RTAUDIO_SINT16, PA_SAMPLE_S16LE},
+ {RTAUDIO_SINT32, PA_SAMPLE_S32LE},
+ {RTAUDIO_FLOAT32, PA_SAMPLE_FLOAT32LE},
+ {0, PA_SAMPLE_INVALID}};
+
+struct PulseAudioHandle {
+ pa_simple *s_play;
+ pa_simple *s_rec;
+ pthread_t thread;
+ pthread_cond_t runnable_cv;
+ bool runnable;
+ PulseAudioHandle() : s_play(0), s_rec(0), runnable(false) { }
+};
- 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;
- }
- }
+RtApiPulse::~RtApiPulse()
+{
+ if ( stream_.state != STREAM_CLOSED )
+ closeStream();
+}
- 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;
- }
- }
- }
+unsigned int RtApiPulse::getDeviceCount( void )
+{
+ return 1;
+}
- stream_.sampleRate = sampleRate;
- stream_.nBuffers = periods;
- stream_.device[mode] = device;
- stream_.state = STREAM_STOPPED;
+RtAudio::DeviceInfo RtApiPulse::getDeviceInfo( unsigned int /*device*/ )
+{
+ RtAudio::DeviceInfo info;
+ info.probed = true;
+ info.name = "PulseAudio";
+ info.outputChannels = 2;
+ info.inputChannels = 2;
+ info.duplexChannels = 2;
+ info.isDefaultOutput = true;
+ info.isDefaultInput = true;
- // Setup the buffer conversion information structure.
- if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
+ for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr )
+ info.sampleRates.push_back( *sr );
- // Setup thread if necessary.
- if ( stream_.mode == OUTPUT && mode == INPUT ) {
- // We had already set up an output stream.
- stream_.mode = DUPLEX;
- // Link the streams if possible.
- apiInfo->synchronized = false;
- if ( snd_pcm_link( apiInfo->handles[0], apiInfo->handles[1] ) == 0 )
- apiInfo->synchronized = true;
- else {
- errorText_ = "RtApiAlsa::probeDeviceOpen: unable to synchronize input and output devices.";
- error( RtError::WARNING );
- }
- }
- else {
- stream_.mode = mode;
+ info.preferredSampleRate = 48000;
+ info.nativeFormats = RTAUDIO_SINT16 | RTAUDIO_SINT32 | RTAUDIO_FLOAT32;
- // Setup callback thread.
- stream_.callbackInfo.object = (void *) this;
+ return info;
+}
- // Set the thread attributes for joinable and realtime scheduling
- // priority. The higher priority will only take affect if the
- // program is run as root or suid.
- pthread_attr_t attr;
- pthread_attr_init( &attr );
- pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
-#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
- pthread_attr_setschedpolicy( &attr, SCHED_RR );
-#else
- pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
+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();
+ }
- stream_.callbackInfo.isRunning = true;
- result = pthread_create( &stream_.callbackInfo.thread, &attr, alsaCallbackHandler, &stream_.callbackInfo );
- pthread_attr_destroy( &attr );
- if ( result ) {
- stream_.callbackInfo.isRunning = false;
- errorText_ = "RtApiAlsa::error creating callback thread!";
- goto error;
+ 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 );
- return SUCCESS;
+ 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 );
- error:
- if ( apiInfo ) {
- if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
- if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
- delete apiInfo;
+ 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_.userBuffer[0] ) {
+ free( stream_.userBuffer[0] );
+ stream_.userBuffer[0] = 0;
}
-
- if ( stream_.deviceBuffer ) {
- free( stream_.deviceBuffer );
- stream_.deviceBuffer = 0;
+ if ( stream_.userBuffer[1] ) {
+ free( stream_.userBuffer[1] );
+ stream_.userBuffer[1] = 0;
}
- return FAILURE;
+ stream_.state = STREAM_CLOSED;
+ stream_.mode = UNINITIALIZED;
}
-void RtApiAlsa :: closeStream()
+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_ = "RtApiAlsa::closeStream(): no open stream to close!";
- error( RtError::WARNING );
+ errorText_ = "RtApiPulse::callbackEvent(): the stream is closed ... "
+ "this shouldn't happen!";
+ error( RtAudioError::WARNING );
return;
}
- stream_.callbackInfo.isRunning = false;
- pthread_join( stream_.callbackInfo.thread, NULL );
+ 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 );
- AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
- if ( stream_.state == STREAM_RUNNING ) {
- stream_.state = STREAM_STOPPED;
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
- snd_pcm_drop( apiInfo->handles[0] );
- if ( stream_.mode == INPUT || stream_.mode == DUPLEX )
- snd_pcm_drop( apiInfo->handles[1] );
+ if ( doStopStream == 2 ) {
+ abortStream();
+ return;
}
- if ( apiInfo ) {
- if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
- if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
- delete apiInfo;
- stream_.apiHandle = 0;
- }
+ 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];
- for ( int i=0; i<2; i++ ) {
- if ( stream_.userBuffer[i] ) {
- free( stream_.userBuffer[i] );
- stream_.userBuffer[i] = 0;
+ 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_.deviceBuffer ) {
- free( stream_.deviceBuffer );
- stream_.deviceBuffer = 0;
+ 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] );
+ }
}
- stream_.mode = UNINITIALIZED;
- stream_.state = STREAM_CLOSED;
+ unlock:
+ MUTEX_UNLOCK( &stream_.mutex );
+ RtApi::tickStreamTime();
+
+ if ( doStopStream == 1 )
+ stopStream();
}
-void RtApiAlsa :: startStream()
+void RtApiPulse::startStream( void )
{
- // This method calls snd_pcm_prepare if the device isn't already in that state.
+ PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
- verifyStream();
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiPulse::startStream(): the stream is not open!";
+ error( RtAudioError::INVALID_USE );
+ return;
+ }
if ( stream_.state == STREAM_RUNNING ) {
- errorText_ = "RtApiAlsa::startStream(): the stream is already running!";
- error( RtError::WARNING );
+ errorText_ = "RtApiPulse::startStream(): the stream is already running!";
+ error( RtAudioError::WARNING );
return;
}
MUTEX_LOCK( &stream_.mutex );
- int result = 0;
- snd_pcm_state_t state;
- AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
- snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
- state = snd_pcm_state( handle[0] );
- if ( state != SND_PCM_STATE_PREPARED ) {
- result = snd_pcm_prepare( handle[0] );
- if ( result < 0 ) {
- errorStream_ << "RtApiAlsa::startStream: error preparing output pcm device, " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- goto unlock;
- }
- }
- }
-
- if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
- 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:
+ pah->runnable = true;
+ pthread_cond_signal( &pah->runnable_cv );
MUTEX_UNLOCK( &stream_.mutex );
-
- if ( result >= 0 ) return;
- error( RtError::SYSTEM_ERROR );
}
-void RtApiAlsa :: stopStream()
+void RtApiPulse::stopStream( void )
{
- verifyStream();
+ 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_ = "RtApiAlsa::stopStream(): the stream is already stopped!";
- error( RtError::WARNING );
+ errorText_ = "RtApiPulse::stopStream(): the stream is already stopped!";
+ error( RtAudioError::WARNING );
return;
}
- // Change the state before the lock to improve shutdown response
- // when using a callback.
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 ) << ".";
+ if ( pah && pah->s_play ) {
+ int pa_error;
+ if ( pa_simple_drain( pah->s_play, &pa_error ) < 0 ) {
+ errorStream_ << "RtApiPulse::stopStream: error draining output device, " <<
+ pa_strerror( pa_error ) << ".";
errorText_ = errorStream_.str();
- goto unlock;
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
}
}
- unlock:
+ stream_.state = STREAM_STOPPED;
MUTEX_UNLOCK( &stream_.mutex );
-
- if ( result >= 0 ) return;
- error( RtError::SYSTEM_ERROR );
}
-void RtApiAlsa :: abortStream()
+void RtApiPulse::abortStream( void )
{
- verifyStream();
+ 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_ = "RtApiAlsa::abortStream(): the stream is already stopped!";
- error( RtError::WARNING );
+ errorText_ = "RtApiPulse::abortStream(): the stream is already stopped!";
+ error( RtAudioError::WARNING );
return;
}
- // Change the state before the lock to improve shutdown response
- // when using a callback.
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 ) << ".";
+ if ( pah && pah->s_play ) {
+ int pa_error;
+ if ( pa_simple_flush( pah->s_play, &pa_error ) < 0 ) {
+ errorStream_ << "RtApiPulse::abortStream: error flushing output device, " <<
+ pa_strerror( pa_error ) << ".";
errorText_ = errorStream_.str();
- goto unlock;
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
}
}
- unlock:
- MUTEX_UNLOCK( &stream_.mutex );
-
stream_.state = STREAM_STOPPED;
- if ( result >= 0 ) return;
- error( RtError::SYSTEM_ERROR );
+ MUTEX_UNLOCK( &stream_.mutex );
}
-void RtApiAlsa :: callbackEvent()
+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 )
{
- if ( stream_.state == STREAM_STOPPED ) {
- if ( stream_.callbackInfo.isRunning ) usleep( 50000 ); // sleep 50 milliseconds
- return;
+ PulseAudioHandle *pah = 0;
+ unsigned long bufferBytes = 0;
+ pa_sample_spec ss;
+
+ if ( device != 0 ) return false;
+ if ( mode != INPUT && mode != OUTPUT ) return false;
+ if ( channels != 1 && channels != 2 ) {
+ errorText_ = "RtApiPulse::probeDeviceOpen: unsupported number of channels.";
+ return false;
}
+ ss.channels = channels;
- if ( stream_.state == STREAM_CLOSED ) {
- errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!";
- error( RtError::WARNING );
- return;
- }
+ if ( firstChannel != 0 ) return false;
- int doStopStream = 0;
- AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
- RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
- double streamTime = getStreamTime();
- RtAudioStreamStatus status = 0;
- if ( stream_.mode != INPUT && apiInfo->xrun[0] == true ) {
- status |= RTAUDIO_OUTPUT_UNDERFLOW;
- apiInfo->xrun[0] = false;
+ 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 ( stream_.mode != OUTPUT && apiInfo->xrun[1] == true ) {
- status |= RTAUDIO_INPUT_OVERFLOW;
- apiInfo->xrun[1] = false;
+ if ( !sr_found ) {
+ errorText_ = "RtApiPulse::probeDeviceOpen: unsupported sample rate.";
+ return false;
}
- doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
- stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
-
- 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;
+ 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;
+ }
- if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+ // Set other stream parameters.
+ if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
+ else stream_.userInterleaved = true;
+ stream_.deviceInterleaved[mode] = true;
+ stream_.nBuffers = 1;
+ stream_.doByteSwap[mode] = false;
+ stream_.nUserChannels[mode] = channels;
+ stream_.nDeviceChannels[mode] = channels + firstChannel;
+ stream_.channelOffset[mode] = 0;
+ std::string streamName = "RtAudio";
- // 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;
- }
+ // 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;
- // 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 );
- }
+ // 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 ( result < (int) stream_.bufferSize ) {
- // Either an error or underrun occured.
- if ( result == -EPIPE ) {
- snd_pcm_state_t state = snd_pcm_state( handle[1] );
- if ( state == SND_PCM_STATE_XRUN ) {
- apiInfo->xrun[1] = true;
- result = snd_pcm_prepare( handle[1] );
- if ( result < 0 ) {
- errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after overrun, " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- }
- }
- else {
- errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- }
+ 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;
}
- else {
- errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
+ }
+
+ 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;
}
- error( RtError::WARNING );
- goto unlock;
}
+ }
- // Do byte swapping if necessary.
- if ( stream_.doByteSwap[1] )
- byteSwapBuffer( buffer, stream_.bufferSize * channels, format );
+ stream_.device[mode] = device;
- // Do buffer conversion if necessary.
- if ( stream_.doConvertBuffer[1] )
- convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
+ // Setup the buffer conversion information structure.
+ if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
- // Check stream latency
- result = snd_pcm_delay( handle[1], &frames );
- if ( result == 0 && frames > 0 ) stream_.latency[1] = frames;
+ 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 );
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ int error;
+ if ( options && !options->streamName.empty() ) streamName = options->streamName;
+ switch ( mode ) {
+ case INPUT:
+ pa_buffer_attr buffer_attr;
+ buffer_attr.fragsize = bufferBytes;
+ buffer_attr.maxlength = -1;
- // 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];
+ pah->s_rec = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_RECORD, NULL, "Record", &ss, NULL, &buffer_attr, &error );
+ if ( !pah->s_rec ) {
+ errorText_ = "RtApiPulse::probeDeviceOpen: error connecting input to PulseAudio server.";
+ goto error;
}
- else {
- buffer = stream_.userBuffer[0];
- channels = stream_.nUserChannels[0];
- format = stream_.userFormat;
+ break;
+ case OUTPUT:
+ pah->s_play = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_PLAYBACK, NULL, "Playback", &ss, NULL, NULL, &error );
+ if ( !pah->s_play ) {
+ errorText_ = "RtApiPulse::probeDeviceOpen: error connecting output to PulseAudio server.";
+ goto error;
}
+ break;
+ default:
+ goto error;
+ }
- // Do byte swapping if necessary.
- if ( stream_.doByteSwap[0] )
- byteSwapBuffer(buffer, stream_.bufferSize * channels, format);
+ if ( stream_.mode == UNINITIALIZED )
+ stream_.mode = mode;
+ else if ( stream_.mode == mode )
+ goto error;
+ else
+ stream_.mode = DUPLEX;
- // 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 ( !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, ¶m);
}
+ else
+ pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
+#else
+ pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
+#endif
- if ( result < (int) stream_.bufferSize ) {
- // Either an error or underrun occured.
- if ( result == -EPIPE ) {
- snd_pcm_state_t state = snd_pcm_state( handle[0] );
- if ( state == SND_PCM_STATE_XRUN ) {
- apiInfo->xrun[0] = true;
- result = snd_pcm_prepare( handle[0] );
- if ( result < 0 ) {
- errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after underrun, " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- }
- }
- else {
- errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
- }
- }
- else {
- errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << ".";
- errorText_ = errorStream_.str();
+ 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;
}
- error( RtError::WARNING );
- goto unlock;
}
-
- // Check stream latency
- result = snd_pcm_delay( handle[0], &frames );
- if ( result == 0 && frames > 0 ) stream_.latency[0] = frames;
}
- unlock:
- MUTEX_UNLOCK( &stream_.mutex );
-
- RtApi::tickStreamTime();
- if ( doStopStream == 1 ) this->stopStream();
- else if ( doStopStream == 2 ) this->abortStream();
-}
-
-extern "C" void *alsaCallbackHandler( void *ptr )
-{
- CallbackInfo *info = (CallbackInfo *) ptr;
- RtApiAlsa *object = (RtApiAlsa *) info->object;
- bool *isRunning = &info->isRunning;
+ return SUCCESS;
+
+ error:
+ if ( pah && stream_.callbackInfo.isRunning ) {
+ pthread_cond_destroy( &pah->runnable_cv );
+ delete pah;
+ stream_.apiHandle = 0;
+ }
-#ifdef SCHED_RR
- // Set a higher scheduler priority (P.J. Leonard)
- struct sched_param param;
- int min = sched_get_priority_min( SCHED_RR );
- int max = sched_get_priority_max( SCHED_RR );
- param.sched_priority = min + ( max - min ) / 2; // Is this the best number?
- sched_setscheduler( 0, SCHED_RR, ¶m );
-#endif
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
+ }
- while ( *isRunning == true ) {
- pthread_testcancel();
- object->callbackEvent();
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
}
- pthread_exit( NULL );
+ stream_.state = STREAM_CLOSED;
+ return FAILURE;
}
-//******************** End of __LINUX_ALSA__ *********************//
+//******************** End of __LINUX_PULSE__ *********************//
#endif
-
#if defined(__LINUX_OSS__)
#include <unistd.h>
#include <sys/ioctl.h>
#include <unistd.h>
#include <fcntl.h>
-#include "oss/soundcard.h"
+#include <sys/soundcard.h>
#include <errno.h>
#include <math.h>
-extern "C" void *ossCallbackHandler(void * ptr);
+static void *ossCallbackHandler(void * ptr);
// A structure to hold various information related to the OSS API
// implementation.
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; }
int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
if ( mixerfd == -1 ) {
errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'.";
- error( RtError::WARNING );
+ error( RtAudioError::WARNING );
return 0;
}
if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) {
close( mixerfd );
errorText_ = "RtApiOss::getDeviceCount: error getting sysinfo, OSS version >= 4.0 is required.";
- error( RtError::WARNING );
+ error( RtAudioError::WARNING );
return 0;
}
+ close( mixerfd );
return sysinfo.numaudios;
}
int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
if ( mixerfd == -1 ) {
errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'.";
- error( RtError::WARNING );
+ error( RtAudioError::WARNING );
return info;
}
if ( result == -1 ) {
close( mixerfd );
errorText_ = "RtApiOss::getDeviceInfo: error getting sysinfo, OSS version >= 4.0 is required.";
- error( RtError::WARNING );
+ error( RtAudioError::WARNING );
return info;
}
if ( nDevices == 0 ) {
close( mixerfd );
errorText_ = "RtApiOss::getDeviceInfo: no devices found!";
- error( RtError::INVALID_USE );
+ error( RtAudioError::INVALID_USE );
+ return info;
}
if ( device >= nDevices ) {
close( mixerfd );
errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!";
- error( RtError::INVALID_USE );
+ error( RtAudioError::INVALID_USE );
+ return info;
}
oss_audioinfo ainfo;
if ( result == -1 ) {
errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
errorText_ = errorStream_.str();
- error( RtError::WARNING );
+ error( RtAudioError::WARNING );
return info;
}
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;
if ( info.nativeFormats == 0 ) {
errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio.";
errorText_ = errorStream_.str();
- error( RtError::WARNING );
+ error( RtAudioError::WARNING );
return info;
}
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] )
+ 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( RtError::WARNING );
+ error( RtAudioError::WARNING );
}
else {
info.probed = true;
bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
- unsigned int firstChannel, unsigned int sampleRate,
+ unsigned int firstChannel, unsigned int sampleRate,
RtAudioFormat format, unsigned int *bufferSize,
RtAudio::StreamOptions *options )
{
// For duplex operation, specifically set this mode (this doesn't seem to work).
/*
- if ( flags | O_RDWR ) {
+ 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;
+ errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
}
- }
*/
// Check the device channel support.
}
// Verify the sample rate setup worked.
- if ( abs( srate - sampleRate ) > 100 ) {
+ if ( abs( srate - (int)sampleRate ) > 100 ) {
close( fd );
errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ").";
errorText_ = errorStream_.str();
goto error;
}
+ if ( pthread_cond_init( &handle->runnable, NULL ) ) {
+ errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable.";
+ goto error;
+ }
+
stream_.apiHandle = (void *) handle;
}
else {
pthread_attr_t attr;
pthread_attr_init( &attr );
pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
-#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
- pthread_attr_setschedpolicy( &attr, SCHED_RR );
+#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, ¶m);
+ }
+ else
+ pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
#else
pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
#endif
result = pthread_create( &stream_.callbackInfo.thread, &attr, ossCallbackHandler, &stream_.callbackInfo );
pthread_attr_destroy( &attr );
if ( result ) {
- stream_.callbackInfo.isRunning = false;
- errorText_ = "RtApiOss::error creating callback thread!";
- goto error;
+ // 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;
+ }
}
}
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_.deviceBuffer = 0;
}
+ stream_.state = STREAM_CLOSED;
return FAILURE;
}
{
if ( stream_.state == STREAM_CLOSED ) {
errorText_ = "RtApiOss::closeStream(): no open stream to close!";
- error( RtError::WARNING );
+ 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 );
- OssHandle *handle = (OssHandle *) stream_.apiHandle;
if ( stream_.state == STREAM_RUNNING ) {
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
}
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;
verifyStream();
if ( stream_.state == STREAM_RUNNING ) {
errorText_ = "RtApiOss::startStream(): the stream is already running!";
- error( RtError::WARNING );
+ error( RtAudioError::WARNING );
return;
}
// when fed samples.
MUTEX_UNLOCK( &stream_.mutex );
+
+ OssHandle *handle = (OssHandle *) stream_.apiHandle;
+ pthread_cond_signal( &handle->runnable );
}
void RtApiOss :: stopStream()
verifyStream();
if ( stream_.state == STREAM_STOPPED ) {
errorText_ = "RtApiOss::stopStream(): the stream is already stopped!";
- error( RtError::WARNING );
+ error( RtAudioError::WARNING );
return;
}
- // Change the state before the lock to improve shutdown response
- // when using a callback.
- stream_.state = STREAM_STOPPED;
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 = write( handle->id[0], buffer, samples * formatBytes(format) );
if ( result == -1 ) {
errorText_ = "RtApiOss::stopStream: audio write error.";
- error( RtError::WARNING );
+ error( RtAudioError::WARNING );
}
}
}
unlock:
+ stream_.state = STREAM_STOPPED;
MUTEX_UNLOCK( &stream_.mutex );
- stream_.state = STREAM_STOPPED;
if ( result != -1 ) return;
- error( RtError::SYSTEM_ERROR );
+ error( RtAudioError::SYSTEM_ERROR );
}
void RtApiOss :: abortStream()
verifyStream();
if ( stream_.state == STREAM_STOPPED ) {
errorText_ = "RtApiOss::abortStream(): the stream is already stopped!";
- error( RtError::WARNING );
+ error( RtAudioError::WARNING );
return;
}
- // Change the state before the lock to improve shutdown response
- // when using a callback.
- stream_.state = STREAM_STOPPED;
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 ) {
}
unlock:
+ stream_.state = STREAM_STOPPED;
MUTEX_UNLOCK( &stream_.mutex );
- stream_.state = STREAM_STOPPED;
if ( result != -1 ) return;
- error( RtError::SYSTEM_ERROR );
+ error( RtAudioError::SYSTEM_ERROR );
}
void RtApiOss :: callbackEvent()
{
+ OssHandle *handle = (OssHandle *) stream_.apiHandle;
if ( stream_.state == STREAM_STOPPED ) {
- if ( stream_.callbackInfo.isRunning ) usleep( 50000 ); // sleep 50 milliseconds
- return;
+ MUTEX_LOCK( &stream_.mutex );
+ pthread_cond_wait( &handle->runnable, &stream_.mutex );
+ if ( stream_.state != STREAM_RUNNING ) {
+ MUTEX_UNLOCK( &stream_.mutex );
+ return;
+ }
+ MUTEX_UNLOCK( &stream_.mutex );
}
if ( stream_.state == STREAM_CLOSED ) {
errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!";
- error( RtError::WARNING );
+ error( RtAudioError::WARNING );
return;
}
RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
double streamTime = getStreamTime();
RtAudioStreamStatus status = 0;
- OssHandle *handle = (OssHandle *) stream_.apiHandle;
if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
status |= RTAUDIO_OUTPUT_UNDERFLOW;
handle->xrun[0] = false;
}
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 );
// specific means for determining that.
handle->xrun[0] = true;
errorText_ = "RtApiOss::callbackEvent: audio write error.";
- error( RtError::WARNING );
- goto unlock;
+ error( RtAudioError::WARNING );
+ // Continue on to input section.
}
}
// specific means for determining that.
handle->xrun[1] = true;
errorText_ = "RtApiOss::callbackEvent: audio read error.";
- error( RtError::WARNING );
+ error( RtAudioError::WARNING );
goto unlock;
}
RtApi::tickStreamTime();
if ( doStopStream == 1 ) this->stopStream();
- else if ( doStopStream == 2 ) this->abortStream();
}
-extern "C" void *ossCallbackHandler( void *ptr )
+static void *ossCallbackHandler( void *ptr )
{
CallbackInfo *info = (CallbackInfo *) ptr;
RtApiOss *object = (RtApiOss *) info->object;
bool *isRunning = &info->isRunning;
-#ifdef SCHED_RR
- // Set a higher scheduler priority (P.J. Leonard)
- struct sched_param param;
- param.sched_priority = 39; // Is this the best number?
- sched_setscheduler( 0, SCHED_RR, ¶m );
+#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 ) {
// This method can be modified to control the behavior of error
// message printing.
-void RtApi :: error( RtError::Type type )
+void RtApi :: error( RtAudioError::Type type )
{
- if ( type == RtError::WARNING && showWarnings_ == true )
- std::cerr << '\n' << errorText_ << "\n\n";
- else
- throw( RtError( errorText_, 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( RtError::INVALID_USE );
+ error( RtAudioError::INVALID_USE );
}
}
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] = 0;
+ stream_.device[i] = 11111;
stream_.doConvertBuffer[i] = false;
stream_.deviceInterleaved[i] = true;
stream_.doByteSwap[i] = false;
{
if ( format == RTAUDIO_SINT16 )
return 2;
- else if ( format == RTAUDIO_SINT24 || format == RTAUDIO_SINT32 ||
- format == RTAUDIO_FLOAT32 )
+ 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( RtError::WARNING );
+ error( RtAudioError::WARNING );
return 0;
}
{
// This function does format conversion, input/output channel compensation, and
// data interleaving/deinterleaving. 24-bit integers are assumed to occupy
- // the upper three bytes of a 32-bit integer.
+ // the lower three bytes of a 32-bit integer.
// Clear our device buffer when in/out duplex device channels are different
if ( outBuffer == stream_.deviceBuffer && stream_.mode == DUPLEX &&
if (info.inFormat == RTAUDIO_SINT8) {
signed char *in = (signed char *)inBuffer;
- scale = 1.0 / 128.0;
+ scale = 1.0 / 127.5;
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]];
+ out[info.outOffset[j]] += 0.5;
out[info.outOffset[j]] *= scale;
}
in += info.inJump;
}
else if (info.inFormat == RTAUDIO_SINT16) {
Int16 *in = (Int16 *)inBuffer;
- scale = 1.0 / 32768.0;
+ scale = 1.0 / 32767.5;
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]];
+ out[info.outOffset[j]] += 0.5;
out[info.outOffset[j]] *= scale;
}
in += info.inJump;
}
}
else if (info.inFormat == RTAUDIO_SINT24) {
- Int32 *in = (Int32 *)inBuffer;
- scale = 1.0 / 8388608.0;
+ Int24 *in = (Int24 *)inBuffer;
+ scale = 1.0 / 8388607.5;
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]] & 0x00ffffff);
+ out[info.outOffset[j]] = (Float64) (in[info.inOffset[j]].asInt());
+ out[info.outOffset[j]] += 0.5;
out[info.outOffset[j]] *= scale;
}
in += info.inJump;
}
else if (info.inFormat == RTAUDIO_SINT32) {
Int32 *in = (Int32 *)inBuffer;
- scale = 1.0 / 2147483648.0;
+ scale = 1.0 / 2147483647.5;
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]];
+ out[info.outOffset[j]] += 0.5;
out[info.outOffset[j]] *= scale;
}
in += info.inJump;
if (info.inFormat == RTAUDIO_SINT8) {
signed char *in = (signed char *)inBuffer;
- scale = 1.0 / 128.0;
+ scale = (Float32) ( 1.0 / 127.5 );
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]];
+ out[info.outOffset[j]] += 0.5;
out[info.outOffset[j]] *= scale;
}
in += info.inJump;
}
else if (info.inFormat == RTAUDIO_SINT16) {
Int16 *in = (Int16 *)inBuffer;
- scale = 1.0 / 32768.0;
+ scale = (Float32) ( 1.0 / 32767.5 );
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]];
+ out[info.outOffset[j]] += 0.5;
out[info.outOffset[j]] *= scale;
}
in += info.inJump;
}
}
else if (info.inFormat == RTAUDIO_SINT24) {
- Int32 *in = (Int32 *)inBuffer;
- scale = 1.0 / 8388608.0;
+ Int24 *in = (Int24 *)inBuffer;
+ scale = (Float32) ( 1.0 / 8388607.5 );
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]] & 0x00ffffff);
+ out[info.outOffset[j]] = (Float32) (in[info.inOffset[j]].asInt());
+ out[info.outOffset[j]] += 0.5;
out[info.outOffset[j]] *= scale;
}
in += info.inJump;
}
else if (info.inFormat == RTAUDIO_SINT32) {
Int32 *in = (Int32 *)inBuffer;
- scale = 1.0 / 2147483648.0;
+ scale = (Float32) ( 1.0 / 2147483647.5 );
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]];
+ out[info.outOffset[j]] += 0.5;
out[info.outOffset[j]] *= scale;
}
in += info.inJump;
}
}
else if (info.inFormat == RTAUDIO_SINT24) {
- Int32 *in = (Int32 *)inBuffer;
+ 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]];
+ out[info.outOffset[j]] = (Int32) in[info.inOffset[j]].asInt();
out[info.outOffset[j]] <<= 8;
}
in += info.inJump;
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) (in[info.inOffset[j]] * 2147483647.0);
+ out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
}
in += info.inJump;
out += info.outJump;
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) (in[info.inOffset[j]] * 2147483647.0);
+ out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
}
in += info.inJump;
out += info.outJump;
}
}
else if (info.outFormat == RTAUDIO_SINT24) {
- Int32 *out = (Int32 *)outBuffer;
+ 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]];
- out[info.outOffset[j]] <<= 16;
+ out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] << 16);
+ //out[info.outOffset[j]] <<= 16;
}
in += info.inJump;
out += info.outJump;
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]] <<= 8;
+ 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.
- Int32 *in = (Int32 *)inBuffer;
+ 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]];
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]];
- out[info.outOffset[j]] >>= 8;
+ out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] >> 8);
+ //out[info.outOffset[j]] >>= 8;
}
in += info.inJump;
out += info.outJump;
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) (in[info.inOffset[j]] * 8388608.0);
+ out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
}
in += info.inJump;
out += info.outJump;
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) (in[info.inOffset[j]] * 2147483647.0);
+ out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
}
in += info.inJump;
out += info.outJump;
}
}
else if (info.inFormat == RTAUDIO_SINT24) {
- Int32 *in = (Int32 *)inBuffer;
+ 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]] >> 8) & 0x0000ffff);
+ out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]].asInt() >> 8);
}
in += info.inJump;
out += info.outJump;
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) (in[info.inOffset[j]] * 32767.0);
+ out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
}
in += info.inJump;
out += info.outJump;
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) (in[info.inOffset[j]] * 32767.0);
+ out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
}
in += info.inJump;
out += info.outJump;
}
}
else if (info.inFormat == RTAUDIO_SINT24) {
- Int32 *in = (Int32 *)inBuffer;
+ 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]] >> 16) & 0x000000ff);
+ out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]].asInt() >> 16);
}
in += info.inJump;
out += info.outJump;
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) (in[info.inOffset[j]] * 127.0);
+ out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
}
in += info.inJump;
out += info.outJump;
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) (in[info.inOffset[j]] * 127.0);
+ out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
}
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 )
{
- register char val;
- register char *ptr;
+ char val;
+ char *ptr;
ptr = buffer;
if ( format == RTAUDIO_SINT16 ) {
ptr += 2;
}
}
- else if ( format == RTAUDIO_SINT24 ||
- format == RTAUDIO_SINT32 ||
+ else if ( format == RTAUDIO_SINT32 ||
format == RTAUDIO_FLOAT32 ) {
for ( unsigned int i=0; i<samples; i++ ) {
// Swap 1st and 4th bytes.
*(ptr) = *(ptr+1);
*(ptr+1) = val;
- // Increment 4 bytes.
- ptr += 4;
+ // 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 ) {
*(ptr) = *(ptr+1);
*(ptr+1) = val;
- // Increment 8 bytes.
- ptr += 8;
+ // 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
+ // Indentation settings for Vim and Emacs
+ //
+ // Local Variables:
+ // c-basic-offset: 2
+ // indent-tabs-mode: nil
+ // End:
+ //
+ // vim: et sts=2 sw=2
projects / rtaudio-cdist.git / blobdiff