1 /************************************************************************/
3 \brief Realtime audio i/o C++ classes.
5 RtAudio provides a common API (Application Programming Interface)
6 for realtime audio input/output across Linux (native ALSA, Jack,
7 and OSS), Macintosh OS X (CoreAudio and Jack), and Windows
8 (DirectSound, ASIO and WASAPI) operating systems.
10 RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/
12 RtAudio: realtime audio i/o C++ classes
13 Copyright (c) 2001-2017 Gary P. Scavone
15 Permission is hereby granted, free of charge, to any person
16 obtaining a copy of this software and associated documentation files
17 (the "Software"), to deal in the Software without restriction,
18 including without limitation the rights to use, copy, modify, merge,
19 publish, distribute, sublicense, and/or sell copies of the Software,
20 and to permit persons to whom the Software is furnished to do so,
21 subject to the following conditions:
23 The above copyright notice and this permission notice shall be
24 included in all copies or substantial portions of the Software.
26 Any person wishing to distribute modifications to the Software is
27 asked to send the modifications to the original developer so that
28 they can be incorporated into the canonical version. This is,
29 however, not a binding provision of this license.
31 THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
32 EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
33 MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
34 IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
35 ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
36 CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
37 WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
39 /************************************************************************/
41 // RtAudio: Version 5.0.0
51 // Static variable definitions.
52 const unsigned int RtApi::MAX_SAMPLE_RATES = 14;
53 const unsigned int RtApi::SAMPLE_RATES[] = {
54 4000, 5512, 8000, 9600, 11025, 16000, 22050,
55 32000, 44100, 48000, 88200, 96000, 176400, 192000
58 #if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) || defined(__WINDOWS_WASAPI__)
59 #define MUTEX_INITIALIZE(A) InitializeCriticalSection(A)
60 #define MUTEX_DESTROY(A) DeleteCriticalSection(A)
61 #define MUTEX_LOCK(A) EnterCriticalSection(A)
62 #define MUTEX_UNLOCK(A) LeaveCriticalSection(A)
66 static std::string convertCharPointerToStdString(const char *text)
68 return std::string(text);
71 static std::string convertCharPointerToStdString(const wchar_t *text)
73 int length = WideCharToMultiByte(CP_UTF8, 0, text, -1, NULL, 0, NULL, NULL);
74 std::string s( length-1, '\0' );
75 WideCharToMultiByte(CP_UTF8, 0, text, -1, &s[0], length, NULL, NULL);
79 #elif defined(__LINUX_ALSA__) || defined(__LINUX_PULSE__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__)
81 #define MUTEX_INITIALIZE(A) pthread_mutex_init(A, NULL)
82 #define MUTEX_DESTROY(A) pthread_mutex_destroy(A)
83 #define MUTEX_LOCK(A) pthread_mutex_lock(A)
84 #define MUTEX_UNLOCK(A) pthread_mutex_unlock(A)
86 #define MUTEX_INITIALIZE(A) abs(*A) // dummy definitions
87 #define MUTEX_DESTROY(A) abs(*A) // dummy definitions
90 // *************************************************** //
92 // RtAudio definitions.
94 // *************************************************** //
96 std::string RtAudio :: getVersion( void )
98 return RTAUDIO_VERSION;
101 void RtAudio :: getCompiledApi( std::vector<RtAudio::Api> &apis )
105 // The order here will control the order of RtAudio's API search in
107 #if defined(__UNIX_JACK__)
108 apis.push_back( UNIX_JACK );
110 #if defined(__LINUX_ALSA__)
111 apis.push_back( LINUX_ALSA );
113 #if defined(__LINUX_PULSE__)
114 apis.push_back( LINUX_PULSE );
116 #if defined(__LINUX_OSS__)
117 apis.push_back( LINUX_OSS );
119 #if defined(__WINDOWS_ASIO__)
120 apis.push_back( WINDOWS_ASIO );
122 #if defined(__WINDOWS_WASAPI__)
123 apis.push_back( WINDOWS_WASAPI );
125 #if defined(__WINDOWS_DS__)
126 apis.push_back( WINDOWS_DS );
128 #if defined(__MACOSX_CORE__)
129 apis.push_back( MACOSX_CORE );
131 #if defined(__RTAUDIO_DUMMY__)
132 apis.push_back( RTAUDIO_DUMMY );
136 void RtAudio :: openRtApi( RtAudio::Api api )
142 #if defined(__UNIX_JACK__)
143 if ( api == UNIX_JACK )
144 rtapi_ = new RtApiJack();
146 #if defined(__LINUX_ALSA__)
147 if ( api == LINUX_ALSA )
148 rtapi_ = new RtApiAlsa();
150 #if defined(__LINUX_PULSE__)
151 if ( api == LINUX_PULSE )
152 rtapi_ = new RtApiPulse();
154 #if defined(__LINUX_OSS__)
155 if ( api == LINUX_OSS )
156 rtapi_ = new RtApiOss();
158 #if defined(__WINDOWS_ASIO__)
159 if ( api == WINDOWS_ASIO )
160 rtapi_ = new RtApiAsio();
162 #if defined(__WINDOWS_WASAPI__)
163 if ( api == WINDOWS_WASAPI )
164 rtapi_ = new RtApiWasapi();
166 #if defined(__WINDOWS_DS__)
167 if ( api == WINDOWS_DS )
168 rtapi_ = new RtApiDs();
170 #if defined(__MACOSX_CORE__)
171 if ( api == MACOSX_CORE )
172 rtapi_ = new RtApiCore();
174 #if defined(__RTAUDIO_DUMMY__)
175 if ( api == RTAUDIO_DUMMY )
176 rtapi_ = new RtApiDummy();
180 RtAudio :: RtAudio( RtAudio::Api api )
184 if ( api != UNSPECIFIED ) {
185 // Attempt to open the specified API.
187 if ( rtapi_ ) return;
189 // No compiled support for specified API value. Issue a debug
190 // warning and continue as if no API was specified.
191 std::cerr << "\nRtAudio: no compiled support for specified API argument!\n" << std::endl;
194 // Iterate through the compiled APIs and return as soon as we find
195 // one with at least one device or we reach the end of the list.
196 std::vector< RtAudio::Api > apis;
197 getCompiledApi( apis );
198 for ( unsigned int i=0; i<apis.size(); i++ ) {
199 openRtApi( apis[i] );
200 if ( rtapi_ && rtapi_->getDeviceCount() ) break;
203 if ( rtapi_ ) return;
205 // It should not be possible to get here because the preprocessor
206 // definition __RTAUDIO_DUMMY__ is automatically defined if no
207 // API-specific definitions are passed to the compiler. But just in
208 // case something weird happens, we'll thow an error.
209 std::string errorText = "\nRtAudio: no compiled API support found ... critical error!!\n\n";
210 throw( RtAudioError( errorText, RtAudioError::UNSPECIFIED ) );
213 RtAudio :: ~RtAudio()
219 void RtAudio :: openStream( RtAudio::StreamParameters *outputParameters,
220 RtAudio::StreamParameters *inputParameters,
221 RtAudioFormat format, unsigned int sampleRate,
222 unsigned int *bufferFrames,
223 RtAudioCallback callback, void *userData,
224 RtAudio::StreamOptions *options,
225 RtAudioErrorCallback errorCallback )
227 return rtapi_->openStream( outputParameters, inputParameters, format,
228 sampleRate, bufferFrames, callback,
229 userData, options, errorCallback );
232 // *************************************************** //
234 // Public RtApi definitions (see end of file for
235 // private or protected utility functions).
237 // *************************************************** //
241 stream_.state = STREAM_CLOSED;
242 stream_.mode = UNINITIALIZED;
243 stream_.apiHandle = 0;
244 stream_.userBuffer[0] = 0;
245 stream_.userBuffer[1] = 0;
246 MUTEX_INITIALIZE( &stream_.mutex );
247 showWarnings_ = true;
248 firstErrorOccurred_ = false;
253 MUTEX_DESTROY( &stream_.mutex );
256 void RtApi :: openStream( RtAudio::StreamParameters *oParams,
257 RtAudio::StreamParameters *iParams,
258 RtAudioFormat format, unsigned int sampleRate,
259 unsigned int *bufferFrames,
260 RtAudioCallback callback, void *userData,
261 RtAudio::StreamOptions *options,
262 RtAudioErrorCallback errorCallback )
264 if ( stream_.state != STREAM_CLOSED ) {
265 errorText_ = "RtApi::openStream: a stream is already open!";
266 error( RtAudioError::INVALID_USE );
270 // Clear stream information potentially left from a previously open stream.
273 if ( oParams && oParams->nChannels < 1 ) {
274 errorText_ = "RtApi::openStream: a non-NULL output StreamParameters structure cannot have an nChannels value less than one.";
275 error( RtAudioError::INVALID_USE );
279 if ( iParams && iParams->nChannels < 1 ) {
280 errorText_ = "RtApi::openStream: a non-NULL input StreamParameters structure cannot have an nChannels value less than one.";
281 error( RtAudioError::INVALID_USE );
285 if ( oParams == NULL && iParams == NULL ) {
286 errorText_ = "RtApi::openStream: input and output StreamParameters structures are both NULL!";
287 error( RtAudioError::INVALID_USE );
291 if ( formatBytes(format) == 0 ) {
292 errorText_ = "RtApi::openStream: 'format' parameter value is undefined.";
293 error( RtAudioError::INVALID_USE );
297 unsigned int nDevices = getDeviceCount();
298 unsigned int oChannels = 0;
300 oChannels = oParams->nChannels;
301 if ( oParams->deviceId >= nDevices ) {
302 errorText_ = "RtApi::openStream: output device parameter value is invalid.";
303 error( RtAudioError::INVALID_USE );
308 unsigned int iChannels = 0;
310 iChannels = iParams->nChannels;
311 if ( iParams->deviceId >= nDevices ) {
312 errorText_ = "RtApi::openStream: input device parameter value is invalid.";
313 error( RtAudioError::INVALID_USE );
320 if ( oChannels > 0 ) {
322 result = probeDeviceOpen( oParams->deviceId, OUTPUT, oChannels, oParams->firstChannel,
323 sampleRate, format, bufferFrames, options );
324 if ( result == false ) {
325 error( RtAudioError::SYSTEM_ERROR );
330 if ( iChannels > 0 ) {
332 result = probeDeviceOpen( iParams->deviceId, INPUT, iChannels, iParams->firstChannel,
333 sampleRate, format, bufferFrames, options );
334 if ( result == false ) {
335 if ( oChannels > 0 ) closeStream();
336 error( RtAudioError::SYSTEM_ERROR );
341 stream_.callbackInfo.callback = (void *) callback;
342 stream_.callbackInfo.userData = userData;
343 stream_.callbackInfo.errorCallback = (void *) errorCallback;
345 if ( options ) options->numberOfBuffers = stream_.nBuffers;
346 stream_.state = STREAM_STOPPED;
349 unsigned int RtApi :: getDefaultInputDevice( void )
351 // Should be implemented in subclasses if possible.
355 unsigned int RtApi :: getDefaultOutputDevice( void )
357 // Should be implemented in subclasses if possible.
361 void RtApi :: closeStream( void )
363 // MUST be implemented in subclasses!
367 bool RtApi :: probeDeviceOpen( unsigned int /*device*/, StreamMode /*mode*/, unsigned int /*channels*/,
368 unsigned int /*firstChannel*/, unsigned int /*sampleRate*/,
369 RtAudioFormat /*format*/, unsigned int * /*bufferSize*/,
370 RtAudio::StreamOptions * /*options*/ )
372 // MUST be implemented in subclasses!
376 void RtApi :: tickStreamTime( void )
378 // Subclasses that do not provide their own implementation of
379 // getStreamTime should call this function once per buffer I/O to
380 // provide basic stream time support.
382 stream_.streamTime += ( stream_.bufferSize * 1.0 / stream_.sampleRate );
384 #if defined( HAVE_GETTIMEOFDAY )
385 gettimeofday( &stream_.lastTickTimestamp, NULL );
389 long RtApi :: getStreamLatency( void )
393 long totalLatency = 0;
394 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
395 totalLatency = stream_.latency[0];
396 if ( stream_.mode == INPUT || stream_.mode == DUPLEX )
397 totalLatency += stream_.latency[1];
402 double RtApi :: getStreamTime( void )
406 #if defined( HAVE_GETTIMEOFDAY )
407 // Return a very accurate estimate of the stream time by
408 // adding in the elapsed time since the last tick.
412 if ( stream_.state != STREAM_RUNNING || stream_.streamTime == 0.0 )
413 return stream_.streamTime;
415 gettimeofday( &now, NULL );
416 then = stream_.lastTickTimestamp;
417 return stream_.streamTime +
418 ((now.tv_sec + 0.000001 * now.tv_usec) -
419 (then.tv_sec + 0.000001 * then.tv_usec));
421 return stream_.streamTime;
425 void RtApi :: setStreamTime( double time )
430 stream_.streamTime = time;
431 #if defined( HAVE_GETTIMEOFDAY )
432 gettimeofday( &stream_.lastTickTimestamp, NULL );
436 unsigned int RtApi :: getStreamSampleRate( void )
440 return stream_.sampleRate;
444 // *************************************************** //
446 // OS/API-specific methods.
448 // *************************************************** //
450 #if defined(__MACOSX_CORE__)
452 // The OS X CoreAudio API is designed to use a separate callback
453 // procedure for each of its audio devices. A single RtAudio duplex
454 // stream using two different devices is supported here, though it
455 // cannot be guaranteed to always behave correctly because we cannot
456 // synchronize these two callbacks.
458 // A property listener is installed for over/underrun information.
459 // However, no functionality is currently provided to allow property
460 // listeners to trigger user handlers because it is unclear what could
461 // be done if a critical stream parameter (buffer size, sample rate,
462 // device disconnect) notification arrived. The listeners entail
463 // quite a bit of extra code and most likely, a user program wouldn't
464 // be prepared for the result anyway. However, we do provide a flag
465 // to the client callback function to inform of an over/underrun.
467 // A structure to hold various information related to the CoreAudio API
470 AudioDeviceID id[2]; // device ids
471 #if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
472 AudioDeviceIOProcID procId[2];
474 UInt32 iStream[2]; // device stream index (or first if using multiple)
475 UInt32 nStreams[2]; // number of streams to use
478 pthread_cond_t condition;
479 int drainCounter; // Tracks callback counts when draining
480 bool internalDrain; // Indicates if stop is initiated from callback or not.
483 :deviceBuffer(0), drainCounter(0), internalDrain(false) { nStreams[0] = 1; nStreams[1] = 1; id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
486 RtApiCore:: RtApiCore()
488 #if defined( AVAILABLE_MAC_OS_X_VERSION_10_6_AND_LATER )
489 // This is a largely undocumented but absolutely necessary
490 // requirement starting with OS-X 10.6. If not called, queries and
491 // updates to various audio device properties are not handled
493 CFRunLoopRef theRunLoop = NULL;
494 AudioObjectPropertyAddress property = { kAudioHardwarePropertyRunLoop,
495 kAudioObjectPropertyScopeGlobal,
496 kAudioObjectPropertyElementMaster };
497 OSStatus result = AudioObjectSetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, sizeof(CFRunLoopRef), &theRunLoop);
498 if ( result != noErr ) {
499 errorText_ = "RtApiCore::RtApiCore: error setting run loop property!";
500 error( RtAudioError::WARNING );
505 RtApiCore :: ~RtApiCore()
507 // The subclass destructor gets called before the base class
508 // destructor, so close an existing stream before deallocating
509 // apiDeviceId memory.
510 if ( stream_.state != STREAM_CLOSED ) closeStream();
513 unsigned int RtApiCore :: getDeviceCount( void )
515 // Find out how many audio devices there are, if any.
517 AudioObjectPropertyAddress propertyAddress = { kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
518 OSStatus result = AudioObjectGetPropertyDataSize( kAudioObjectSystemObject, &propertyAddress, 0, NULL, &dataSize );
519 if ( result != noErr ) {
520 errorText_ = "RtApiCore::getDeviceCount: OS-X error getting device info!";
521 error( RtAudioError::WARNING );
525 return dataSize / sizeof( AudioDeviceID );
528 unsigned int RtApiCore :: getDefaultInputDevice( void )
530 unsigned int nDevices = getDeviceCount();
531 if ( nDevices <= 1 ) return 0;
534 UInt32 dataSize = sizeof( AudioDeviceID );
535 AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultInputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
536 OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id );
537 if ( result != noErr ) {
538 errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device.";
539 error( RtAudioError::WARNING );
543 dataSize *= nDevices;
544 AudioDeviceID deviceList[ nDevices ];
545 property.mSelector = kAudioHardwarePropertyDevices;
546 result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList );
547 if ( result != noErr ) {
548 errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device IDs.";
549 error( RtAudioError::WARNING );
553 for ( unsigned int i=0; i<nDevices; i++ )
554 if ( id == deviceList[i] ) return i;
556 errorText_ = "RtApiCore::getDefaultInputDevice: No default device found!";
557 error( RtAudioError::WARNING );
561 unsigned int RtApiCore :: getDefaultOutputDevice( void )
563 unsigned int nDevices = getDeviceCount();
564 if ( nDevices <= 1 ) return 0;
567 UInt32 dataSize = sizeof( AudioDeviceID );
568 AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultOutputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
569 OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id );
570 if ( result != noErr ) {
571 errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device.";
572 error( RtAudioError::WARNING );
576 dataSize = sizeof( AudioDeviceID ) * nDevices;
577 AudioDeviceID deviceList[ nDevices ];
578 property.mSelector = kAudioHardwarePropertyDevices;
579 result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList );
580 if ( result != noErr ) {
581 errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device IDs.";
582 error( RtAudioError::WARNING );
586 for ( unsigned int i=0; i<nDevices; i++ )
587 if ( id == deviceList[i] ) return i;
589 errorText_ = "RtApiCore::getDefaultOutputDevice: No default device found!";
590 error( RtAudioError::WARNING );
594 RtAudio::DeviceInfo RtApiCore :: getDeviceInfo( unsigned int device )
596 RtAudio::DeviceInfo info;
600 unsigned int nDevices = getDeviceCount();
601 if ( nDevices == 0 ) {
602 errorText_ = "RtApiCore::getDeviceInfo: no devices found!";
603 error( RtAudioError::INVALID_USE );
607 if ( device >= nDevices ) {
608 errorText_ = "RtApiCore::getDeviceInfo: device ID is invalid!";
609 error( RtAudioError::INVALID_USE );
613 AudioDeviceID deviceList[ nDevices ];
614 UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;
615 AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
616 kAudioObjectPropertyScopeGlobal,
617 kAudioObjectPropertyElementMaster };
618 OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property,
619 0, NULL, &dataSize, (void *) &deviceList );
620 if ( result != noErr ) {
621 errorText_ = "RtApiCore::getDeviceInfo: OS-X system error getting device IDs.";
622 error( RtAudioError::WARNING );
626 AudioDeviceID id = deviceList[ device ];
628 // Get the device name.
631 dataSize = sizeof( CFStringRef );
632 property.mSelector = kAudioObjectPropertyManufacturer;
633 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname );
634 if ( result != noErr ) {
635 errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device manufacturer.";
636 errorText_ = errorStream_.str();
637 error( RtAudioError::WARNING );
641 //const char *mname = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() );
642 int length = CFStringGetLength(cfname);
643 char *mname = (char *)malloc(length * 3 + 1);
644 #if defined( UNICODE ) || defined( _UNICODE )
645 CFStringGetCString(cfname, mname, length * 3 + 1, kCFStringEncodingUTF8);
647 CFStringGetCString(cfname, mname, length * 3 + 1, CFStringGetSystemEncoding());
649 info.name.append( (const char *)mname, strlen(mname) );
650 info.name.append( ": " );
654 property.mSelector = kAudioObjectPropertyName;
655 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname );
656 if ( result != noErr ) {
657 errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device name.";
658 errorText_ = errorStream_.str();
659 error( RtAudioError::WARNING );
663 //const char *name = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() );
664 length = CFStringGetLength(cfname);
665 char *name = (char *)malloc(length * 3 + 1);
666 #if defined( UNICODE ) || defined( _UNICODE )
667 CFStringGetCString(cfname, name, length * 3 + 1, kCFStringEncodingUTF8);
669 CFStringGetCString(cfname, name, length * 3 + 1, CFStringGetSystemEncoding());
671 info.name.append( (const char *)name, strlen(name) );
675 // Get the output stream "configuration".
676 AudioBufferList *bufferList = nil;
677 property.mSelector = kAudioDevicePropertyStreamConfiguration;
678 property.mScope = kAudioDevicePropertyScopeOutput;
679 // property.mElement = kAudioObjectPropertyElementWildcard;
681 result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
682 if ( result != noErr || dataSize == 0 ) {
683 errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration info for device (" << device << ").";
684 errorText_ = errorStream_.str();
685 error( RtAudioError::WARNING );
689 // Allocate the AudioBufferList.
690 bufferList = (AudioBufferList *) malloc( dataSize );
691 if ( bufferList == NULL ) {
692 errorText_ = "RtApiCore::getDeviceInfo: memory error allocating output AudioBufferList.";
693 error( RtAudioError::WARNING );
697 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
698 if ( result != noErr || dataSize == 0 ) {
700 errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration for device (" << device << ").";
701 errorText_ = errorStream_.str();
702 error( RtAudioError::WARNING );
706 // Get output channel information.
707 unsigned int i, nStreams = bufferList->mNumberBuffers;
708 for ( i=0; i<nStreams; i++ )
709 info.outputChannels += bufferList->mBuffers[i].mNumberChannels;
712 // Get the input stream "configuration".
713 property.mScope = kAudioDevicePropertyScopeInput;
714 result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
715 if ( result != noErr || dataSize == 0 ) {
716 errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration info for device (" << device << ").";
717 errorText_ = errorStream_.str();
718 error( RtAudioError::WARNING );
722 // Allocate the AudioBufferList.
723 bufferList = (AudioBufferList *) malloc( dataSize );
724 if ( bufferList == NULL ) {
725 errorText_ = "RtApiCore::getDeviceInfo: memory error allocating input AudioBufferList.";
726 error( RtAudioError::WARNING );
730 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
731 if (result != noErr || dataSize == 0) {
733 errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration for device (" << device << ").";
734 errorText_ = errorStream_.str();
735 error( RtAudioError::WARNING );
739 // Get input channel information.
740 nStreams = bufferList->mNumberBuffers;
741 for ( i=0; i<nStreams; i++ )
742 info.inputChannels += bufferList->mBuffers[i].mNumberChannels;
745 // If device opens for both playback and capture, we determine the channels.
746 if ( info.outputChannels > 0 && info.inputChannels > 0 )
747 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
749 // Probe the device sample rates.
750 bool isInput = false;
751 if ( info.outputChannels == 0 ) isInput = true;
753 // Determine the supported sample rates.
754 property.mSelector = kAudioDevicePropertyAvailableNominalSampleRates;
755 if ( isInput == false ) property.mScope = kAudioDevicePropertyScopeOutput;
756 result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
757 if ( result != kAudioHardwareNoError || dataSize == 0 ) {
758 errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rate info.";
759 errorText_ = errorStream_.str();
760 error( RtAudioError::WARNING );
764 UInt32 nRanges = dataSize / sizeof( AudioValueRange );
765 AudioValueRange rangeList[ nRanges ];
766 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &rangeList );
767 if ( result != kAudioHardwareNoError ) {
768 errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rates.";
769 errorText_ = errorStream_.str();
770 error( RtAudioError::WARNING );
774 // The sample rate reporting mechanism is a bit of a mystery. It
775 // seems that it can either return individual rates or a range of
776 // rates. I assume that if the min / max range values are the same,
777 // then that represents a single supported rate and if the min / max
778 // range values are different, the device supports an arbitrary
779 // range of values (though there might be multiple ranges, so we'll
780 // use the most conservative range).
781 Float64 minimumRate = 1.0, maximumRate = 10000000000.0;
782 bool haveValueRange = false;
783 info.sampleRates.clear();
784 for ( UInt32 i=0; i<nRanges; i++ ) {
785 if ( rangeList[i].mMinimum == rangeList[i].mMaximum ) {
786 unsigned int tmpSr = (unsigned int) rangeList[i].mMinimum;
787 info.sampleRates.push_back( tmpSr );
789 if ( !info.preferredSampleRate || ( tmpSr <= 48000 && tmpSr > info.preferredSampleRate ) )
790 info.preferredSampleRate = tmpSr;
793 haveValueRange = true;
794 if ( rangeList[i].mMinimum > minimumRate ) minimumRate = rangeList[i].mMinimum;
795 if ( rangeList[i].mMaximum < maximumRate ) maximumRate = rangeList[i].mMaximum;
799 if ( haveValueRange ) {
800 for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
801 if ( SAMPLE_RATES[k] >= (unsigned int) minimumRate && SAMPLE_RATES[k] <= (unsigned int) maximumRate ) {
802 info.sampleRates.push_back( SAMPLE_RATES[k] );
804 if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
805 info.preferredSampleRate = SAMPLE_RATES[k];
810 // Sort and remove any redundant values
811 std::sort( info.sampleRates.begin(), info.sampleRates.end() );
812 info.sampleRates.erase( unique( info.sampleRates.begin(), info.sampleRates.end() ), info.sampleRates.end() );
814 if ( info.sampleRates.size() == 0 ) {
815 errorStream_ << "RtApiCore::probeDeviceInfo: No supported sample rates found for device (" << device << ").";
816 errorText_ = errorStream_.str();
817 error( RtAudioError::WARNING );
821 // CoreAudio always uses 32-bit floating point data for PCM streams.
822 // Thus, any other "physical" formats supported by the device are of
823 // no interest to the client.
824 info.nativeFormats = RTAUDIO_FLOAT32;
826 if ( info.outputChannels > 0 )
827 if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true;
828 if ( info.inputChannels > 0 )
829 if ( getDefaultInputDevice() == device ) info.isDefaultInput = true;
835 static OSStatus callbackHandler( AudioDeviceID inDevice,
836 const AudioTimeStamp* /*inNow*/,
837 const AudioBufferList* inInputData,
838 const AudioTimeStamp* /*inInputTime*/,
839 AudioBufferList* outOutputData,
840 const AudioTimeStamp* /*inOutputTime*/,
843 CallbackInfo *info = (CallbackInfo *) infoPointer;
845 RtApiCore *object = (RtApiCore *) info->object;
846 if ( object->callbackEvent( inDevice, inInputData, outOutputData ) == false )
847 return kAudioHardwareUnspecifiedError;
849 return kAudioHardwareNoError;
852 static OSStatus xrunListener( AudioObjectID /*inDevice*/,
854 const AudioObjectPropertyAddress properties[],
855 void* handlePointer )
857 CoreHandle *handle = (CoreHandle *) handlePointer;
858 for ( UInt32 i=0; i<nAddresses; i++ ) {
859 if ( properties[i].mSelector == kAudioDeviceProcessorOverload ) {
860 if ( properties[i].mScope == kAudioDevicePropertyScopeInput )
861 handle->xrun[1] = true;
863 handle->xrun[0] = true;
867 return kAudioHardwareNoError;
870 static OSStatus rateListener( AudioObjectID inDevice,
871 UInt32 /*nAddresses*/,
872 const AudioObjectPropertyAddress /*properties*/[],
875 Float64 *rate = (Float64 *) ratePointer;
876 UInt32 dataSize = sizeof( Float64 );
877 AudioObjectPropertyAddress property = { kAudioDevicePropertyNominalSampleRate,
878 kAudioObjectPropertyScopeGlobal,
879 kAudioObjectPropertyElementMaster };
880 AudioObjectGetPropertyData( inDevice, &property, 0, NULL, &dataSize, rate );
881 return kAudioHardwareNoError;
884 bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
885 unsigned int firstChannel, unsigned int sampleRate,
886 RtAudioFormat format, unsigned int *bufferSize,
887 RtAudio::StreamOptions *options )
890 unsigned int nDevices = getDeviceCount();
891 if ( nDevices == 0 ) {
892 // This should not happen because a check is made before this function is called.
893 errorText_ = "RtApiCore::probeDeviceOpen: no devices found!";
897 if ( device >= nDevices ) {
898 // This should not happen because a check is made before this function is called.
899 errorText_ = "RtApiCore::probeDeviceOpen: device ID is invalid!";
903 AudioDeviceID deviceList[ nDevices ];
904 UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;
905 AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
906 kAudioObjectPropertyScopeGlobal,
907 kAudioObjectPropertyElementMaster };
908 OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property,
909 0, NULL, &dataSize, (void *) &deviceList );
910 if ( result != noErr ) {
911 errorText_ = "RtApiCore::probeDeviceOpen: OS-X system error getting device IDs.";
915 AudioDeviceID id = deviceList[ device ];
917 // Setup for stream mode.
918 bool isInput = false;
919 if ( mode == INPUT ) {
921 property.mScope = kAudioDevicePropertyScopeInput;
924 property.mScope = kAudioDevicePropertyScopeOutput;
926 // Get the stream "configuration".
927 AudioBufferList *bufferList = nil;
929 property.mSelector = kAudioDevicePropertyStreamConfiguration;
930 result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
931 if ( result != noErr || dataSize == 0 ) {
932 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration info for device (" << device << ").";
933 errorText_ = errorStream_.str();
937 // Allocate the AudioBufferList.
938 bufferList = (AudioBufferList *) malloc( dataSize );
939 if ( bufferList == NULL ) {
940 errorText_ = "RtApiCore::probeDeviceOpen: memory error allocating AudioBufferList.";
944 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
945 if (result != noErr || dataSize == 0) {
947 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration for device (" << device << ").";
948 errorText_ = errorStream_.str();
952 // Search for one or more streams that contain the desired number of
953 // channels. CoreAudio devices can have an arbitrary number of
954 // streams and each stream can have an arbitrary number of channels.
955 // For each stream, a single buffer of interleaved samples is
956 // provided. RtAudio prefers the use of one stream of interleaved
957 // data or multiple consecutive single-channel streams. However, we
958 // now support multiple consecutive multi-channel streams of
959 // interleaved data as well.
960 UInt32 iStream, offsetCounter = firstChannel;
961 UInt32 nStreams = bufferList->mNumberBuffers;
962 bool monoMode = false;
963 bool foundStream = false;
965 // First check that the device supports the requested number of
967 UInt32 deviceChannels = 0;
968 for ( iStream=0; iStream<nStreams; iStream++ )
969 deviceChannels += bufferList->mBuffers[iStream].mNumberChannels;
971 if ( deviceChannels < ( channels + firstChannel ) ) {
973 errorStream_ << "RtApiCore::probeDeviceOpen: the device (" << device << ") does not support the requested channel count.";
974 errorText_ = errorStream_.str();
978 // Look for a single stream meeting our needs.
979 UInt32 firstStream, streamCount = 1, streamChannels = 0, channelOffset = 0;
980 for ( iStream=0; iStream<nStreams; iStream++ ) {
981 streamChannels = bufferList->mBuffers[iStream].mNumberChannels;
982 if ( streamChannels >= channels + offsetCounter ) {
983 firstStream = iStream;
984 channelOffset = offsetCounter;
988 if ( streamChannels > offsetCounter ) break;
989 offsetCounter -= streamChannels;
992 // If we didn't find a single stream above, then we should be able
993 // to meet the channel specification with multiple streams.
994 if ( foundStream == false ) {
996 offsetCounter = firstChannel;
997 for ( iStream=0; iStream<nStreams; iStream++ ) {
998 streamChannels = bufferList->mBuffers[iStream].mNumberChannels;
999 if ( streamChannels > offsetCounter ) break;
1000 offsetCounter -= streamChannels;
1003 firstStream = iStream;
1004 channelOffset = offsetCounter;
1005 Int32 channelCounter = channels + offsetCounter - streamChannels;
1007 if ( streamChannels > 1 ) monoMode = false;
1008 while ( channelCounter > 0 ) {
1009 streamChannels = bufferList->mBuffers[++iStream].mNumberChannels;
1010 if ( streamChannels > 1 ) monoMode = false;
1011 channelCounter -= streamChannels;
1018 // Determine the buffer size.
1019 AudioValueRange bufferRange;
1020 dataSize = sizeof( AudioValueRange );
1021 property.mSelector = kAudioDevicePropertyBufferFrameSizeRange;
1022 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &bufferRange );
1024 if ( result != noErr ) {
1025 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting buffer size range for device (" << device << ").";
1026 errorText_ = errorStream_.str();
1030 if ( bufferRange.mMinimum > *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMinimum;
1031 else if ( bufferRange.mMaximum < *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMaximum;
1032 if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) *bufferSize = (unsigned long) bufferRange.mMinimum;
1034 // Set the buffer size. For multiple streams, I'm assuming we only
1035 // need to make this setting for the master channel.
1036 UInt32 theSize = (UInt32) *bufferSize;
1037 dataSize = sizeof( UInt32 );
1038 property.mSelector = kAudioDevicePropertyBufferFrameSize;
1039 result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &theSize );
1041 if ( result != noErr ) {
1042 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting the buffer size for device (" << device << ").";
1043 errorText_ = errorStream_.str();
1047 // If attempting to setup a duplex stream, the bufferSize parameter
1048 // MUST be the same in both directions!
1049 *bufferSize = theSize;
1050 if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {
1051 errorStream_ << "RtApiCore::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << device << ").";
1052 errorText_ = errorStream_.str();
1056 stream_.bufferSize = *bufferSize;
1057 stream_.nBuffers = 1;
1059 // Try to set "hog" mode ... it's not clear to me this is working.
1060 if ( options && options->flags & RTAUDIO_HOG_DEVICE ) {
1062 dataSize = sizeof( hog_pid );
1063 property.mSelector = kAudioDevicePropertyHogMode;
1064 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &hog_pid );
1065 if ( result != noErr ) {
1066 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting 'hog' state!";
1067 errorText_ = errorStream_.str();
1071 if ( hog_pid != getpid() ) {
1073 result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &hog_pid );
1074 if ( result != noErr ) {
1075 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting 'hog' state!";
1076 errorText_ = errorStream_.str();
1082 // Check and if necessary, change the sample rate for the device.
1083 Float64 nominalRate;
1084 dataSize = sizeof( Float64 );
1085 property.mSelector = kAudioDevicePropertyNominalSampleRate;
1086 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &nominalRate );
1087 if ( result != noErr ) {
1088 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting current sample rate.";
1089 errorText_ = errorStream_.str();
1093 // Only change the sample rate if off by more than 1 Hz.
1094 if ( fabs( nominalRate - (double)sampleRate ) > 1.0 ) {
1096 // Set a property listener for the sample rate change
1097 Float64 reportedRate = 0.0;
1098 AudioObjectPropertyAddress tmp = { kAudioDevicePropertyNominalSampleRate, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
1099 result = AudioObjectAddPropertyListener( id, &tmp, rateListener, (void *) &reportedRate );
1100 if ( result != noErr ) {
1101 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate property listener for device (" << device << ").";
1102 errorText_ = errorStream_.str();
1106 nominalRate = (Float64) sampleRate;
1107 result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &nominalRate );
1108 if ( result != noErr ) {
1109 AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate );
1110 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate for device (" << device << ").";
1111 errorText_ = errorStream_.str();
1115 // Now wait until the reported nominal rate is what we just set.
1116 UInt32 microCounter = 0;
1117 while ( reportedRate != nominalRate ) {
1118 microCounter += 5000;
1119 if ( microCounter > 5000000 ) break;
1123 // Remove the property listener.
1124 AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate );
1126 if ( microCounter > 5000000 ) {
1127 errorStream_ << "RtApiCore::probeDeviceOpen: timeout waiting for sample rate update for device (" << device << ").";
1128 errorText_ = errorStream_.str();
1133 // Now set the stream format for all streams. Also, check the
1134 // physical format of the device and change that if necessary.
1135 AudioStreamBasicDescription description;
1136 dataSize = sizeof( AudioStreamBasicDescription );
1137 property.mSelector = kAudioStreamPropertyVirtualFormat;
1138 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &description );
1139 if ( result != noErr ) {
1140 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream format for device (" << device << ").";
1141 errorText_ = errorStream_.str();
1145 // Set the sample rate and data format id. However, only make the
1146 // change if the sample rate is not within 1.0 of the desired
1147 // rate and the format is not linear pcm.
1148 bool updateFormat = false;
1149 if ( fabs( description.mSampleRate - (Float64)sampleRate ) > 1.0 ) {
1150 description.mSampleRate = (Float64) sampleRate;
1151 updateFormat = true;
1154 if ( description.mFormatID != kAudioFormatLinearPCM ) {
1155 description.mFormatID = kAudioFormatLinearPCM;
1156 updateFormat = true;
1159 if ( updateFormat ) {
1160 result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &description );
1161 if ( result != noErr ) {
1162 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate or data format for device (" << device << ").";
1163 errorText_ = errorStream_.str();
1168 // Now check the physical format.
1169 property.mSelector = kAudioStreamPropertyPhysicalFormat;
1170 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &description );
1171 if ( result != noErr ) {
1172 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream physical format for device (" << device << ").";
1173 errorText_ = errorStream_.str();
1177 //std::cout << "Current physical stream format:" << std::endl;
1178 //std::cout << " mBitsPerChan = " << description.mBitsPerChannel << std::endl;
1179 //std::cout << " aligned high = " << (description.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (description.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl;
1180 //std::cout << " bytesPerFrame = " << description.mBytesPerFrame << std::endl;
1181 //std::cout << " sample rate = " << description.mSampleRate << std::endl;
1183 if ( description.mFormatID != kAudioFormatLinearPCM || description.mBitsPerChannel < 16 ) {
1184 description.mFormatID = kAudioFormatLinearPCM;
1185 //description.mSampleRate = (Float64) sampleRate;
1186 AudioStreamBasicDescription testDescription = description;
1189 // We'll try higher bit rates first and then work our way down.
1190 std::vector< std::pair<UInt32, UInt32> > physicalFormats;
1191 formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsFloat) & ~kLinearPCMFormatFlagIsSignedInteger;
1192 physicalFormats.push_back( std::pair<Float32, UInt32>( 32, formatFlags ) );
1193 formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat;
1194 physicalFormats.push_back( std::pair<Float32, UInt32>( 32, formatFlags ) );
1195 physicalFormats.push_back( std::pair<Float32, UInt32>( 24, formatFlags ) ); // 24-bit packed
1196 formatFlags &= ~( kAudioFormatFlagIsPacked | kAudioFormatFlagIsAlignedHigh );
1197 physicalFormats.push_back( std::pair<Float32, UInt32>( 24.2, formatFlags ) ); // 24-bit in 4 bytes, aligned low
1198 formatFlags |= kAudioFormatFlagIsAlignedHigh;
1199 physicalFormats.push_back( std::pair<Float32, UInt32>( 24.4, formatFlags ) ); // 24-bit in 4 bytes, aligned high
1200 formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat;
1201 physicalFormats.push_back( std::pair<Float32, UInt32>( 16, formatFlags ) );
1202 physicalFormats.push_back( std::pair<Float32, UInt32>( 8, formatFlags ) );
1204 bool setPhysicalFormat = false;
1205 for( unsigned int i=0; i<physicalFormats.size(); i++ ) {
1206 testDescription = description;
1207 testDescription.mBitsPerChannel = (UInt32) physicalFormats[i].first;
1208 testDescription.mFormatFlags = physicalFormats[i].second;
1209 if ( (24 == (UInt32)physicalFormats[i].first) && ~( physicalFormats[i].second & kAudioFormatFlagIsPacked ) )
1210 testDescription.mBytesPerFrame = 4 * testDescription.mChannelsPerFrame;
1212 testDescription.mBytesPerFrame = testDescription.mBitsPerChannel/8 * testDescription.mChannelsPerFrame;
1213 testDescription.mBytesPerPacket = testDescription.mBytesPerFrame * testDescription.mFramesPerPacket;
1214 result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &testDescription );
1215 if ( result == noErr ) {
1216 setPhysicalFormat = true;
1217 //std::cout << "Updated physical stream format:" << std::endl;
1218 //std::cout << " mBitsPerChan = " << testDescription.mBitsPerChannel << std::endl;
1219 //std::cout << " aligned high = " << (testDescription.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (testDescription.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl;
1220 //std::cout << " bytesPerFrame = " << testDescription.mBytesPerFrame << std::endl;
1221 //std::cout << " sample rate = " << testDescription.mSampleRate << std::endl;
1226 if ( !setPhysicalFormat ) {
1227 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting physical data format for device (" << device << ").";
1228 errorText_ = errorStream_.str();
1231 } // done setting virtual/physical formats.
1233 // Get the stream / device latency.
1235 dataSize = sizeof( UInt32 );
1236 property.mSelector = kAudioDevicePropertyLatency;
1237 if ( AudioObjectHasProperty( id, &property ) == true ) {
1238 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &latency );
1239 if ( result == kAudioHardwareNoError ) stream_.latency[ mode ] = latency;
1241 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting device latency for device (" << device << ").";
1242 errorText_ = errorStream_.str();
1243 error( RtAudioError::WARNING );
1247 // Byte-swapping: According to AudioHardware.h, the stream data will
1248 // always be presented in native-endian format, so we should never
1249 // need to byte swap.
1250 stream_.doByteSwap[mode] = false;
1252 // From the CoreAudio documentation, PCM data must be supplied as
1254 stream_.userFormat = format;
1255 stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
1257 if ( streamCount == 1 )
1258 stream_.nDeviceChannels[mode] = description.mChannelsPerFrame;
1259 else // multiple streams
1260 stream_.nDeviceChannels[mode] = channels;
1261 stream_.nUserChannels[mode] = channels;
1262 stream_.channelOffset[mode] = channelOffset; // offset within a CoreAudio stream
1263 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
1264 else stream_.userInterleaved = true;
1265 stream_.deviceInterleaved[mode] = true;
1266 if ( monoMode == true ) stream_.deviceInterleaved[mode] = false;
1268 // Set flags for buffer conversion.
1269 stream_.doConvertBuffer[mode] = false;
1270 if ( stream_.userFormat != stream_.deviceFormat[mode] )
1271 stream_.doConvertBuffer[mode] = true;
1272 if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
1273 stream_.doConvertBuffer[mode] = true;
1274 if ( streamCount == 1 ) {
1275 if ( stream_.nUserChannels[mode] > 1 &&
1276 stream_.userInterleaved != stream_.deviceInterleaved[mode] )
1277 stream_.doConvertBuffer[mode] = true;
1279 else if ( monoMode && stream_.userInterleaved )
1280 stream_.doConvertBuffer[mode] = true;
1282 // Allocate our CoreHandle structure for the stream.
1283 CoreHandle *handle = 0;
1284 if ( stream_.apiHandle == 0 ) {
1286 handle = new CoreHandle;
1288 catch ( std::bad_alloc& ) {
1289 errorText_ = "RtApiCore::probeDeviceOpen: error allocating CoreHandle memory.";
1293 if ( pthread_cond_init( &handle->condition, NULL ) ) {
1294 errorText_ = "RtApiCore::probeDeviceOpen: error initializing pthread condition variable.";
1297 stream_.apiHandle = (void *) handle;
1300 handle = (CoreHandle *) stream_.apiHandle;
1301 handle->iStream[mode] = firstStream;
1302 handle->nStreams[mode] = streamCount;
1303 handle->id[mode] = id;
1305 // Allocate necessary internal buffers.
1306 unsigned long bufferBytes;
1307 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
1308 // stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
1309 stream_.userBuffer[mode] = (char *) malloc( bufferBytes * sizeof(char) );
1310 memset( stream_.userBuffer[mode], 0, bufferBytes * sizeof(char) );
1311 if ( stream_.userBuffer[mode] == NULL ) {
1312 errorText_ = "RtApiCore::probeDeviceOpen: error allocating user buffer memory.";
1316 // If possible, we will make use of the CoreAudio stream buffers as
1317 // "device buffers". However, we can't do this if using multiple
1319 if ( stream_.doConvertBuffer[mode] && handle->nStreams[mode] > 1 ) {
1321 bool makeBuffer = true;
1322 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
1323 if ( mode == INPUT ) {
1324 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
1325 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
1326 if ( bufferBytes <= bytesOut ) makeBuffer = false;
1331 bufferBytes *= *bufferSize;
1332 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
1333 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
1334 if ( stream_.deviceBuffer == NULL ) {
1335 errorText_ = "RtApiCore::probeDeviceOpen: error allocating device buffer memory.";
1341 stream_.sampleRate = sampleRate;
1342 stream_.device[mode] = device;
1343 stream_.state = STREAM_STOPPED;
1344 stream_.callbackInfo.object = (void *) this;
1346 // Setup the buffer conversion information structure.
1347 if ( stream_.doConvertBuffer[mode] ) {
1348 if ( streamCount > 1 ) setConvertInfo( mode, 0 );
1349 else setConvertInfo( mode, channelOffset );
1352 if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device )
1353 // Only one callback procedure per device.
1354 stream_.mode = DUPLEX;
1356 #if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
1357 result = AudioDeviceCreateIOProcID( id, callbackHandler, (void *) &stream_.callbackInfo, &handle->procId[mode] );
1359 // deprecated in favor of AudioDeviceCreateIOProcID()
1360 result = AudioDeviceAddIOProc( id, callbackHandler, (void *) &stream_.callbackInfo );
1362 if ( result != noErr ) {
1363 errorStream_ << "RtApiCore::probeDeviceOpen: system error setting callback for device (" << device << ").";
1364 errorText_ = errorStream_.str();
1367 if ( stream_.mode == OUTPUT && mode == INPUT )
1368 stream_.mode = DUPLEX;
1370 stream_.mode = mode;
1373 // Setup the device property listener for over/underload.
1374 property.mSelector = kAudioDeviceProcessorOverload;
1375 property.mScope = kAudioObjectPropertyScopeGlobal;
1376 result = AudioObjectAddPropertyListener( id, &property, xrunListener, (void *) handle );
1382 pthread_cond_destroy( &handle->condition );
1384 stream_.apiHandle = 0;
1387 for ( int i=0; i<2; i++ ) {
1388 if ( stream_.userBuffer[i] ) {
1389 free( stream_.userBuffer[i] );
1390 stream_.userBuffer[i] = 0;
1394 if ( stream_.deviceBuffer ) {
1395 free( stream_.deviceBuffer );
1396 stream_.deviceBuffer = 0;
1399 stream_.state = STREAM_CLOSED;
1403 void RtApiCore :: closeStream( void )
1405 if ( stream_.state == STREAM_CLOSED ) {
1406 errorText_ = "RtApiCore::closeStream(): no open stream to close!";
1407 error( RtAudioError::WARNING );
1411 CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
1412 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
1414 AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
1415 kAudioObjectPropertyScopeGlobal,
1416 kAudioObjectPropertyElementMaster };
1418 property.mSelector = kAudioDeviceProcessorOverload;
1419 property.mScope = kAudioObjectPropertyScopeGlobal;
1420 if (AudioObjectRemovePropertyListener( handle->id[0], &property, xrunListener, (void *) handle ) != noErr) {
1421 errorText_ = "RtApiCore::closeStream(): error removing property listener!";
1422 error( RtAudioError::WARNING );
1425 if ( stream_.state == STREAM_RUNNING )
1426 AudioDeviceStop( handle->id[0], callbackHandler );
1427 #if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
1428 AudioDeviceDestroyIOProcID( handle->id[0], handle->procId[0] );
1430 // deprecated in favor of AudioDeviceDestroyIOProcID()
1431 AudioDeviceRemoveIOProc( handle->id[0], callbackHandler );
1435 if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
1437 AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
1438 kAudioObjectPropertyScopeGlobal,
1439 kAudioObjectPropertyElementMaster };
1441 property.mSelector = kAudioDeviceProcessorOverload;
1442 property.mScope = kAudioObjectPropertyScopeGlobal;
1443 if (AudioObjectRemovePropertyListener( handle->id[1], &property, xrunListener, (void *) handle ) != noErr) {
1444 errorText_ = "RtApiCore::closeStream(): error removing property listener!";
1445 error( RtAudioError::WARNING );
1448 if ( stream_.state == STREAM_RUNNING )
1449 AudioDeviceStop( handle->id[1], callbackHandler );
1450 #if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
1451 AudioDeviceDestroyIOProcID( handle->id[1], handle->procId[1] );
1453 // deprecated in favor of AudioDeviceDestroyIOProcID()
1454 AudioDeviceRemoveIOProc( handle->id[1], callbackHandler );
1458 for ( int i=0; i<2; i++ ) {
1459 if ( stream_.userBuffer[i] ) {
1460 free( stream_.userBuffer[i] );
1461 stream_.userBuffer[i] = 0;
1465 if ( stream_.deviceBuffer ) {
1466 free( stream_.deviceBuffer );
1467 stream_.deviceBuffer = 0;
1470 // Destroy pthread condition variable.
1471 pthread_cond_destroy( &handle->condition );
1473 stream_.apiHandle = 0;
1475 stream_.mode = UNINITIALIZED;
1476 stream_.state = STREAM_CLOSED;
1479 void RtApiCore :: startStream( void )
1482 if ( stream_.state == STREAM_RUNNING ) {
1483 errorText_ = "RtApiCore::startStream(): the stream is already running!";
1484 error( RtAudioError::WARNING );
1488 OSStatus result = noErr;
1489 CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
1490 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
1492 result = AudioDeviceStart( handle->id[0], callbackHandler );
1493 if ( result != noErr ) {
1494 errorStream_ << "RtApiCore::startStream: system error (" << getErrorCode( result ) << ") starting callback procedure on device (" << stream_.device[0] << ").";
1495 errorText_ = errorStream_.str();
1500 if ( stream_.mode == INPUT ||
1501 ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
1503 result = AudioDeviceStart( handle->id[1], callbackHandler );
1504 if ( result != noErr ) {
1505 errorStream_ << "RtApiCore::startStream: system error starting input callback procedure on device (" << stream_.device[1] << ").";
1506 errorText_ = errorStream_.str();
1511 handle->drainCounter = 0;
1512 handle->internalDrain = false;
1513 stream_.state = STREAM_RUNNING;
1516 if ( result == noErr ) return;
1517 error( RtAudioError::SYSTEM_ERROR );
1520 void RtApiCore :: stopStream( void )
1523 if ( stream_.state == STREAM_STOPPED ) {
1524 errorText_ = "RtApiCore::stopStream(): the stream is already stopped!";
1525 error( RtAudioError::WARNING );
1529 OSStatus result = noErr;
1530 CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
1531 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
1533 if ( handle->drainCounter == 0 ) {
1534 handle->drainCounter = 2;
1535 pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
1538 result = AudioDeviceStop( handle->id[0], callbackHandler );
1539 if ( result != noErr ) {
1540 errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping callback procedure on device (" << stream_.device[0] << ").";
1541 errorText_ = errorStream_.str();
1546 if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
1548 result = AudioDeviceStop( handle->id[1], callbackHandler );
1549 if ( result != noErr ) {
1550 errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping input callback procedure on device (" << stream_.device[1] << ").";
1551 errorText_ = errorStream_.str();
1556 stream_.state = STREAM_STOPPED;
1559 if ( result == noErr ) return;
1560 error( RtAudioError::SYSTEM_ERROR );
1563 void RtApiCore :: abortStream( void )
1566 if ( stream_.state == STREAM_STOPPED ) {
1567 errorText_ = "RtApiCore::abortStream(): the stream is already stopped!";
1568 error( RtAudioError::WARNING );
1572 CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
1573 handle->drainCounter = 2;
1578 // This function will be called by a spawned thread when the user
1579 // callback function signals that the stream should be stopped or
1580 // aborted. It is better to handle it this way because the
1581 // callbackEvent() function probably should return before the AudioDeviceStop()
1582 // function is called.
1583 static void *coreStopStream( void *ptr )
1585 CallbackInfo *info = (CallbackInfo *) ptr;
1586 RtApiCore *object = (RtApiCore *) info->object;
1588 object->stopStream();
1589 pthread_exit( NULL );
1592 bool RtApiCore :: callbackEvent( AudioDeviceID deviceId,
1593 const AudioBufferList *inBufferList,
1594 const AudioBufferList *outBufferList )
1596 if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;
1597 if ( stream_.state == STREAM_CLOSED ) {
1598 errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";
1599 error( RtAudioError::WARNING );
1603 CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
1604 CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
1606 // Check if we were draining the stream and signal is finished.
1607 if ( handle->drainCounter > 3 ) {
1608 ThreadHandle threadId;
1610 stream_.state = STREAM_STOPPING;
1611 if ( handle->internalDrain == true )
1612 pthread_create( &threadId, NULL, coreStopStream, info );
1613 else // external call to stopStream()
1614 pthread_cond_signal( &handle->condition );
1618 AudioDeviceID outputDevice = handle->id[0];
1620 // Invoke user callback to get fresh output data UNLESS we are
1621 // draining stream or duplex mode AND the input/output devices are
1622 // different AND this function is called for the input device.
1623 if ( handle->drainCounter == 0 && ( stream_.mode != DUPLEX || deviceId == outputDevice ) ) {
1624 RtAudioCallback callback = (RtAudioCallback) info->callback;
1625 double streamTime = getStreamTime();
1626 RtAudioStreamStatus status = 0;
1627 if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
1628 status |= RTAUDIO_OUTPUT_UNDERFLOW;
1629 handle->xrun[0] = false;
1631 if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
1632 status |= RTAUDIO_INPUT_OVERFLOW;
1633 handle->xrun[1] = false;
1636 int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
1637 stream_.bufferSize, streamTime, status, info->userData );
1638 if ( cbReturnValue == 2 ) {
1639 stream_.state = STREAM_STOPPING;
1640 handle->drainCounter = 2;
1644 else if ( cbReturnValue == 1 ) {
1645 handle->drainCounter = 1;
1646 handle->internalDrain = true;
1650 if ( stream_.mode == OUTPUT || ( stream_.mode == DUPLEX && deviceId == outputDevice ) ) {
1652 if ( handle->drainCounter > 1 ) { // write zeros to the output stream
1654 if ( handle->nStreams[0] == 1 ) {
1655 memset( outBufferList->mBuffers[handle->iStream[0]].mData,
1657 outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );
1659 else { // fill multiple streams with zeros
1660 for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) {
1661 memset( outBufferList->mBuffers[handle->iStream[0]+i].mData,
1663 outBufferList->mBuffers[handle->iStream[0]+i].mDataByteSize );
1667 else if ( handle->nStreams[0] == 1 ) {
1668 if ( stream_.doConvertBuffer[0] ) { // convert directly to CoreAudio stream buffer
1669 convertBuffer( (char *) outBufferList->mBuffers[handle->iStream[0]].mData,
1670 stream_.userBuffer[0], stream_.convertInfo[0] );
1672 else { // copy from user buffer
1673 memcpy( outBufferList->mBuffers[handle->iStream[0]].mData,
1674 stream_.userBuffer[0],
1675 outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );
1678 else { // fill multiple streams
1679 Float32 *inBuffer = (Float32 *) stream_.userBuffer[0];
1680 if ( stream_.doConvertBuffer[0] ) {
1681 convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
1682 inBuffer = (Float32 *) stream_.deviceBuffer;
1685 if ( stream_.deviceInterleaved[0] == false ) { // mono mode
1686 UInt32 bufferBytes = outBufferList->mBuffers[handle->iStream[0]].mDataByteSize;
1687 for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
1688 memcpy( outBufferList->mBuffers[handle->iStream[0]+i].mData,
1689 (void *)&inBuffer[i*stream_.bufferSize], bufferBytes );
1692 else { // fill multiple multi-channel streams with interleaved data
1693 UInt32 streamChannels, channelsLeft, inJump, outJump, inOffset;
1696 bool inInterleaved = ( stream_.userInterleaved ) ? true : false;
1697 UInt32 inChannels = stream_.nUserChannels[0];
1698 if ( stream_.doConvertBuffer[0] ) {
1699 inInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode
1700 inChannels = stream_.nDeviceChannels[0];
1703 if ( inInterleaved ) inOffset = 1;
1704 else inOffset = stream_.bufferSize;
1706 channelsLeft = inChannels;
1707 for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) {
1709 out = (Float32 *) outBufferList->mBuffers[handle->iStream[0]+i].mData;
1710 streamChannels = outBufferList->mBuffers[handle->iStream[0]+i].mNumberChannels;
1713 // Account for possible channel offset in first stream
1714 if ( i == 0 && stream_.channelOffset[0] > 0 ) {
1715 streamChannels -= stream_.channelOffset[0];
1716 outJump = stream_.channelOffset[0];
1720 // Account for possible unfilled channels at end of the last stream
1721 if ( streamChannels > channelsLeft ) {
1722 outJump = streamChannels - channelsLeft;
1723 streamChannels = channelsLeft;
1726 // Determine input buffer offsets and skips
1727 if ( inInterleaved ) {
1728 inJump = inChannels;
1729 in += inChannels - channelsLeft;
1733 in += (inChannels - channelsLeft) * inOffset;
1736 for ( unsigned int i=0; i<stream_.bufferSize; i++ ) {
1737 for ( unsigned int j=0; j<streamChannels; j++ ) {
1738 *out++ = in[j*inOffset];
1743 channelsLeft -= streamChannels;
1749 // Don't bother draining input
1750 if ( handle->drainCounter ) {
1751 handle->drainCounter++;
1755 AudioDeviceID inputDevice;
1756 inputDevice = handle->id[1];
1757 if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && deviceId == inputDevice ) ) {
1759 if ( handle->nStreams[1] == 1 ) {
1760 if ( stream_.doConvertBuffer[1] ) { // convert directly from CoreAudio stream buffer
1761 convertBuffer( stream_.userBuffer[1],
1762 (char *) inBufferList->mBuffers[handle->iStream[1]].mData,
1763 stream_.convertInfo[1] );
1765 else { // copy to user buffer
1766 memcpy( stream_.userBuffer[1],
1767 inBufferList->mBuffers[handle->iStream[1]].mData,
1768 inBufferList->mBuffers[handle->iStream[1]].mDataByteSize );
1771 else { // read from multiple streams
1772 Float32 *outBuffer = (Float32 *) stream_.userBuffer[1];
1773 if ( stream_.doConvertBuffer[1] ) outBuffer = (Float32 *) stream_.deviceBuffer;
1775 if ( stream_.deviceInterleaved[1] == false ) { // mono mode
1776 UInt32 bufferBytes = inBufferList->mBuffers[handle->iStream[1]].mDataByteSize;
1777 for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
1778 memcpy( (void *)&outBuffer[i*stream_.bufferSize],
1779 inBufferList->mBuffers[handle->iStream[1]+i].mData, bufferBytes );
1782 else { // read from multiple multi-channel streams
1783 UInt32 streamChannels, channelsLeft, inJump, outJump, outOffset;
1786 bool outInterleaved = ( stream_.userInterleaved ) ? true : false;
1787 UInt32 outChannels = stream_.nUserChannels[1];
1788 if ( stream_.doConvertBuffer[1] ) {
1789 outInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode
1790 outChannels = stream_.nDeviceChannels[1];
1793 if ( outInterleaved ) outOffset = 1;
1794 else outOffset = stream_.bufferSize;
1796 channelsLeft = outChannels;
1797 for ( unsigned int i=0; i<handle->nStreams[1]; i++ ) {
1799 in = (Float32 *) inBufferList->mBuffers[handle->iStream[1]+i].mData;
1800 streamChannels = inBufferList->mBuffers[handle->iStream[1]+i].mNumberChannels;
1803 // Account for possible channel offset in first stream
1804 if ( i == 0 && stream_.channelOffset[1] > 0 ) {
1805 streamChannels -= stream_.channelOffset[1];
1806 inJump = stream_.channelOffset[1];
1810 // Account for possible unread channels at end of the last stream
1811 if ( streamChannels > channelsLeft ) {
1812 inJump = streamChannels - channelsLeft;
1813 streamChannels = channelsLeft;
1816 // Determine output buffer offsets and skips
1817 if ( outInterleaved ) {
1818 outJump = outChannels;
1819 out += outChannels - channelsLeft;
1823 out += (outChannels - channelsLeft) * outOffset;
1826 for ( unsigned int i=0; i<stream_.bufferSize; i++ ) {
1827 for ( unsigned int j=0; j<streamChannels; j++ ) {
1828 out[j*outOffset] = *in++;
1833 channelsLeft -= streamChannels;
1837 if ( stream_.doConvertBuffer[1] ) { // convert from our internal "device" buffer
1838 convertBuffer( stream_.userBuffer[1],
1839 stream_.deviceBuffer,
1840 stream_.convertInfo[1] );
1846 //MUTEX_UNLOCK( &stream_.mutex );
1848 RtApi::tickStreamTime();
1852 const char* RtApiCore :: getErrorCode( OSStatus code )
1856 case kAudioHardwareNotRunningError:
1857 return "kAudioHardwareNotRunningError";
1859 case kAudioHardwareUnspecifiedError:
1860 return "kAudioHardwareUnspecifiedError";
1862 case kAudioHardwareUnknownPropertyError:
1863 return "kAudioHardwareUnknownPropertyError";
1865 case kAudioHardwareBadPropertySizeError:
1866 return "kAudioHardwareBadPropertySizeError";
1868 case kAudioHardwareIllegalOperationError:
1869 return "kAudioHardwareIllegalOperationError";
1871 case kAudioHardwareBadObjectError:
1872 return "kAudioHardwareBadObjectError";
1874 case kAudioHardwareBadDeviceError:
1875 return "kAudioHardwareBadDeviceError";
1877 case kAudioHardwareBadStreamError:
1878 return "kAudioHardwareBadStreamError";
1880 case kAudioHardwareUnsupportedOperationError:
1881 return "kAudioHardwareUnsupportedOperationError";
1883 case kAudioDeviceUnsupportedFormatError:
1884 return "kAudioDeviceUnsupportedFormatError";
1886 case kAudioDevicePermissionsError:
1887 return "kAudioDevicePermissionsError";
1890 return "CoreAudio unknown error";
1894 //******************** End of __MACOSX_CORE__ *********************//
1897 #if defined(__UNIX_JACK__)
1899 // JACK is a low-latency audio server, originally written for the
1900 // GNU/Linux operating system and now also ported to OS-X. It can
1901 // connect a number of different applications to an audio device, as
1902 // well as allowing them to share audio between themselves.
1904 // When using JACK with RtAudio, "devices" refer to JACK clients that
1905 // have ports connected to the server. The JACK server is typically
1906 // started in a terminal as follows:
1908 // .jackd -d alsa -d hw:0
1910 // or through an interface program such as qjackctl. Many of the
1911 // parameters normally set for a stream are fixed by the JACK server
1912 // and can be specified when the JACK server is started. In
1915 // .jackd -d alsa -d hw:0 -r 44100 -p 512 -n 4
1917 // specifies a sample rate of 44100 Hz, a buffer size of 512 sample
1918 // frames, and number of buffers = 4. Once the server is running, it
1919 // is not possible to override these values. If the values are not
1920 // specified in the command-line, the JACK server uses default values.
1922 // The JACK server does not have to be running when an instance of
1923 // RtApiJack is created, though the function getDeviceCount() will
1924 // report 0 devices found until JACK has been started. When no
1925 // devices are available (i.e., the JACK server is not running), a
1926 // stream cannot be opened.
1928 #include <jack/jack.h>
1932 // A structure to hold various information related to the Jack API
1935 jack_client_t *client;
1936 jack_port_t **ports[2];
1937 std::string deviceName[2];
1939 pthread_cond_t condition;
1940 int drainCounter; // Tracks callback counts when draining
1941 bool internalDrain; // Indicates if stop is initiated from callback or not.
1944 :client(0), drainCounter(0), internalDrain(false) { ports[0] = 0; ports[1] = 0; xrun[0] = false; xrun[1] = false; }
1947 #if !defined(__RTAUDIO_DEBUG__)
1948 static void jackSilentError( const char * ) {};
1951 RtApiJack :: RtApiJack()
1952 :shouldAutoconnect_(true) {
1953 // Nothing to do here.
1954 #if !defined(__RTAUDIO_DEBUG__)
1955 // Turn off Jack's internal error reporting.
1956 jack_set_error_function( &jackSilentError );
1960 RtApiJack :: ~RtApiJack()
1962 if ( stream_.state != STREAM_CLOSED ) closeStream();
1965 unsigned int RtApiJack :: getDeviceCount( void )
1967 // See if we can become a jack client.
1968 jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption;
1969 jack_status_t *status = NULL;
1970 jack_client_t *client = jack_client_open( "RtApiJackCount", options, status );
1971 if ( client == 0 ) return 0;
1974 std::string port, previousPort;
1975 unsigned int nChannels = 0, nDevices = 0;
1976 ports = jack_get_ports( client, NULL, NULL, 0 );
1978 // Parse the port names up to the first colon (:).
1981 port = (char *) ports[ nChannels ];
1982 iColon = port.find(":");
1983 if ( iColon != std::string::npos ) {
1984 port = port.substr( 0, iColon + 1 );
1985 if ( port != previousPort ) {
1987 previousPort = port;
1990 } while ( ports[++nChannels] );
1994 jack_client_close( client );
1998 RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device )
2000 RtAudio::DeviceInfo info;
2001 info.probed = false;
2003 jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption
2004 jack_status_t *status = NULL;
2005 jack_client_t *client = jack_client_open( "RtApiJackInfo", options, status );
2006 if ( client == 0 ) {
2007 errorText_ = "RtApiJack::getDeviceInfo: Jack server not found or connection error!";
2008 error( RtAudioError::WARNING );
2013 std::string port, previousPort;
2014 unsigned int nPorts = 0, nDevices = 0;
2015 ports = jack_get_ports( client, NULL, NULL, 0 );
2017 // Parse the port names up to the first colon (:).
2020 port = (char *) ports[ nPorts ];
2021 iColon = port.find(":");
2022 if ( iColon != std::string::npos ) {
2023 port = port.substr( 0, iColon );
2024 if ( port != previousPort ) {
2025 if ( nDevices == device ) info.name = port;
2027 previousPort = port;
2030 } while ( ports[++nPorts] );
2034 if ( device >= nDevices ) {
2035 jack_client_close( client );
2036 errorText_ = "RtApiJack::getDeviceInfo: device ID is invalid!";
2037 error( RtAudioError::INVALID_USE );
2041 // Get the current jack server sample rate.
2042 info.sampleRates.clear();
2044 info.preferredSampleRate = jack_get_sample_rate( client );
2045 info.sampleRates.push_back( info.preferredSampleRate );
2047 // Count the available ports containing the client name as device
2048 // channels. Jack "input ports" equal RtAudio output channels.
2049 unsigned int nChannels = 0;
2050 ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsInput );
2052 while ( ports[ nChannels ] ) nChannels++;
2054 info.outputChannels = nChannels;
2057 // Jack "output ports" equal RtAudio input channels.
2059 ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsOutput );
2061 while ( ports[ nChannels ] ) nChannels++;
2063 info.inputChannels = nChannels;
2066 if ( info.outputChannels == 0 && info.inputChannels == 0 ) {
2067 jack_client_close(client);
2068 errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!";
2069 error( RtAudioError::WARNING );
2073 // If device opens for both playback and capture, we determine the channels.
2074 if ( info.outputChannels > 0 && info.inputChannels > 0 )
2075 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
2077 // Jack always uses 32-bit floats.
2078 info.nativeFormats = RTAUDIO_FLOAT32;
2080 // Jack doesn't provide default devices so we'll use the first available one.
2081 if ( device == 0 && info.outputChannels > 0 )
2082 info.isDefaultOutput = true;
2083 if ( device == 0 && info.inputChannels > 0 )
2084 info.isDefaultInput = true;
2086 jack_client_close(client);
2091 static int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer )
2093 CallbackInfo *info = (CallbackInfo *) infoPointer;
2095 RtApiJack *object = (RtApiJack *) info->object;
2096 if ( object->callbackEvent( (unsigned long) nframes ) == false ) return 1;
2101 // This function will be called by a spawned thread when the Jack
2102 // server signals that it is shutting down. It is necessary to handle
2103 // it this way because the jackShutdown() function must return before
2104 // the jack_deactivate() function (in closeStream()) will return.
2105 static void *jackCloseStream( void *ptr )
2107 CallbackInfo *info = (CallbackInfo *) ptr;
2108 RtApiJack *object = (RtApiJack *) info->object;
2110 object->closeStream();
2112 pthread_exit( NULL );
2114 static void jackShutdown( void *infoPointer )
2116 CallbackInfo *info = (CallbackInfo *) infoPointer;
2117 RtApiJack *object = (RtApiJack *) info->object;
2119 // Check current stream state. If stopped, then we'll assume this
2120 // was called as a result of a call to RtApiJack::stopStream (the
2121 // deactivation of a client handle causes this function to be called).
2122 // If not, we'll assume the Jack server is shutting down or some
2123 // other problem occurred and we should close the stream.
2124 if ( object->isStreamRunning() == false ) return;
2126 ThreadHandle threadId;
2127 pthread_create( &threadId, NULL, jackCloseStream, info );
2128 std::cerr << "\nRtApiJack: the Jack server is shutting down this client ... stream stopped and closed!!\n" << std::endl;
2131 static int jackXrun( void *infoPointer )
2133 JackHandle *handle = (JackHandle *) infoPointer;
2135 if ( handle->ports[0] ) handle->xrun[0] = true;
2136 if ( handle->ports[1] ) handle->xrun[1] = true;
2141 bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
2142 unsigned int firstChannel, unsigned int sampleRate,
2143 RtAudioFormat format, unsigned int *bufferSize,
2144 RtAudio::StreamOptions *options )
2146 JackHandle *handle = (JackHandle *) stream_.apiHandle;
2148 // Look for jack server and try to become a client (only do once per stream).
2149 jack_client_t *client = 0;
2150 if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) {
2151 jack_options_t jackoptions = (jack_options_t) ( JackNoStartServer ); //JackNullOption;
2152 jack_status_t *status = NULL;
2153 if ( options && !options->streamName.empty() )
2154 client = jack_client_open( options->streamName.c_str(), jackoptions, status );
2156 client = jack_client_open( "RtApiJack", jackoptions, status );
2157 if ( client == 0 ) {
2158 errorText_ = "RtApiJack::probeDeviceOpen: Jack server not found or connection error!";
2159 error( RtAudioError::WARNING );
2164 // The handle must have been created on an earlier pass.
2165 client = handle->client;
2169 std::string port, previousPort, deviceName;
2170 unsigned int nPorts = 0, nDevices = 0;
2171 ports = jack_get_ports( client, NULL, NULL, 0 );
2173 // Parse the port names up to the first colon (:).
2176 port = (char *) ports[ nPorts ];
2177 iColon = port.find(":");
2178 if ( iColon != std::string::npos ) {
2179 port = port.substr( 0, iColon );
2180 if ( port != previousPort ) {
2181 if ( nDevices == device ) deviceName = port;
2183 previousPort = port;
2186 } while ( ports[++nPorts] );
2190 if ( device >= nDevices ) {
2191 errorText_ = "RtApiJack::probeDeviceOpen: device ID is invalid!";
2195 // Count the available ports containing the client name as device
2196 // channels. Jack "input ports" equal RtAudio output channels.
2197 unsigned int nChannels = 0;
2198 unsigned long flag = JackPortIsInput;
2199 if ( mode == INPUT ) flag = JackPortIsOutput;
2200 ports = jack_get_ports( client, deviceName.c_str(), NULL, flag );
2202 while ( ports[ nChannels ] ) nChannels++;
2206 // Compare the jack ports for specified client to the requested number of channels.
2207 if ( nChannels < (channels + firstChannel) ) {
2208 errorStream_ << "RtApiJack::probeDeviceOpen: requested number of channels (" << channels << ") + offset (" << firstChannel << ") not found for specified device (" << device << ":" << deviceName << ").";
2209 errorText_ = errorStream_.str();
2213 // Check the jack server sample rate.
2214 unsigned int jackRate = jack_get_sample_rate( client );
2215 if ( sampleRate != jackRate ) {
2216 jack_client_close( client );
2217 errorStream_ << "RtApiJack::probeDeviceOpen: the requested sample rate (" << sampleRate << ") is different than the JACK server rate (" << jackRate << ").";
2218 errorText_ = errorStream_.str();
2221 stream_.sampleRate = jackRate;
2223 // Get the latency of the JACK port.
2224 ports = jack_get_ports( client, deviceName.c_str(), NULL, flag );
2225 if ( ports[ firstChannel ] ) {
2227 jack_latency_callback_mode_t cbmode = (mode == INPUT ? JackCaptureLatency : JackPlaybackLatency);
2228 // the range (usually the min and max are equal)
2229 jack_latency_range_t latrange; latrange.min = latrange.max = 0;
2230 // get the latency range
2231 jack_port_get_latency_range( jack_port_by_name( client, ports[firstChannel] ), cbmode, &latrange );
2232 // be optimistic, use the min!
2233 stream_.latency[mode] = latrange.min;
2234 //stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) );
2238 // The jack server always uses 32-bit floating-point data.
2239 stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
2240 stream_.userFormat = format;
2242 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
2243 else stream_.userInterleaved = true;
2245 // Jack always uses non-interleaved buffers.
2246 stream_.deviceInterleaved[mode] = false;
2248 // Jack always provides host byte-ordered data.
2249 stream_.doByteSwap[mode] = false;
2251 // Get the buffer size. The buffer size and number of buffers
2252 // (periods) is set when the jack server is started.
2253 stream_.bufferSize = (int) jack_get_buffer_size( client );
2254 *bufferSize = stream_.bufferSize;
2256 stream_.nDeviceChannels[mode] = channels;
2257 stream_.nUserChannels[mode] = channels;
2259 // Set flags for buffer conversion.
2260 stream_.doConvertBuffer[mode] = false;
2261 if ( stream_.userFormat != stream_.deviceFormat[mode] )
2262 stream_.doConvertBuffer[mode] = true;
2263 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
2264 stream_.nUserChannels[mode] > 1 )
2265 stream_.doConvertBuffer[mode] = true;
2267 // Allocate our JackHandle structure for the stream.
2268 if ( handle == 0 ) {
2270 handle = new JackHandle;
2272 catch ( std::bad_alloc& ) {
2273 errorText_ = "RtApiJack::probeDeviceOpen: error allocating JackHandle memory.";
2277 if ( pthread_cond_init(&handle->condition, NULL) ) {
2278 errorText_ = "RtApiJack::probeDeviceOpen: error initializing pthread condition variable.";
2281 stream_.apiHandle = (void *) handle;
2282 handle->client = client;
2284 handle->deviceName[mode] = deviceName;
2286 // Allocate necessary internal buffers.
2287 unsigned long bufferBytes;
2288 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
2289 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
2290 if ( stream_.userBuffer[mode] == NULL ) {
2291 errorText_ = "RtApiJack::probeDeviceOpen: error allocating user buffer memory.";
2295 if ( stream_.doConvertBuffer[mode] ) {
2297 bool makeBuffer = true;
2298 if ( mode == OUTPUT )
2299 bufferBytes = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
2300 else { // mode == INPUT
2301 bufferBytes = stream_.nDeviceChannels[1] * formatBytes( stream_.deviceFormat[1] );
2302 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
2303 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]);
2304 if ( bufferBytes < bytesOut ) makeBuffer = false;
2309 bufferBytes *= *bufferSize;
2310 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
2311 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
2312 if ( stream_.deviceBuffer == NULL ) {
2313 errorText_ = "RtApiJack::probeDeviceOpen: error allocating device buffer memory.";
2319 // Allocate memory for the Jack ports (channels) identifiers.
2320 handle->ports[mode] = (jack_port_t **) malloc ( sizeof (jack_port_t *) * channels );
2321 if ( handle->ports[mode] == NULL ) {
2322 errorText_ = "RtApiJack::probeDeviceOpen: error allocating port memory.";
2326 stream_.device[mode] = device;
2327 stream_.channelOffset[mode] = firstChannel;
2328 stream_.state = STREAM_STOPPED;
2329 stream_.callbackInfo.object = (void *) this;
2331 if ( stream_.mode == OUTPUT && mode == INPUT )
2332 // We had already set up the stream for output.
2333 stream_.mode = DUPLEX;
2335 stream_.mode = mode;
2336 jack_set_process_callback( handle->client, jackCallbackHandler, (void *) &stream_.callbackInfo );
2337 jack_set_xrun_callback( handle->client, jackXrun, (void *) &handle );
2338 jack_on_shutdown( handle->client, jackShutdown, (void *) &stream_.callbackInfo );
2341 // Register our ports.
2343 if ( mode == OUTPUT ) {
2344 for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
2345 snprintf( label, 64, "outport %d", i );
2346 handle->ports[0][i] = jack_port_register( handle->client, (const char *)label,
2347 JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0 );
2351 for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
2352 snprintf( label, 64, "inport %d", i );
2353 handle->ports[1][i] = jack_port_register( handle->client, (const char *)label,
2354 JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0 );
2358 // Setup the buffer conversion information structure. We don't use
2359 // buffers to do channel offsets, so we override that parameter
2361 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );
2363 if ( options && options->flags & RTAUDIO_JACK_DONT_CONNECT ) shouldAutoconnect_ = false;
2369 pthread_cond_destroy( &handle->condition );
2370 jack_client_close( handle->client );
2372 if ( handle->ports[0] ) free( handle->ports[0] );
2373 if ( handle->ports[1] ) free( handle->ports[1] );
2376 stream_.apiHandle = 0;
2379 for ( int i=0; i<2; i++ ) {
2380 if ( stream_.userBuffer[i] ) {
2381 free( stream_.userBuffer[i] );
2382 stream_.userBuffer[i] = 0;
2386 if ( stream_.deviceBuffer ) {
2387 free( stream_.deviceBuffer );
2388 stream_.deviceBuffer = 0;
2394 void RtApiJack :: closeStream( void )
2396 if ( stream_.state == STREAM_CLOSED ) {
2397 errorText_ = "RtApiJack::closeStream(): no open stream to close!";
2398 error( RtAudioError::WARNING );
2402 JackHandle *handle = (JackHandle *) stream_.apiHandle;
2405 if ( stream_.state == STREAM_RUNNING )
2406 jack_deactivate( handle->client );
2408 jack_client_close( handle->client );
2412 if ( handle->ports[0] ) free( handle->ports[0] );
2413 if ( handle->ports[1] ) free( handle->ports[1] );
2414 pthread_cond_destroy( &handle->condition );
2416 stream_.apiHandle = 0;
2419 for ( int i=0; i<2; i++ ) {
2420 if ( stream_.userBuffer[i] ) {
2421 free( stream_.userBuffer[i] );
2422 stream_.userBuffer[i] = 0;
2426 if ( stream_.deviceBuffer ) {
2427 free( stream_.deviceBuffer );
2428 stream_.deviceBuffer = 0;
2431 stream_.mode = UNINITIALIZED;
2432 stream_.state = STREAM_CLOSED;
2435 void RtApiJack :: startStream( void )
2438 if ( stream_.state == STREAM_RUNNING ) {
2439 errorText_ = "RtApiJack::startStream(): the stream is already running!";
2440 error( RtAudioError::WARNING );
2444 JackHandle *handle = (JackHandle *) stream_.apiHandle;
2445 int result = jack_activate( handle->client );
2447 errorText_ = "RtApiJack::startStream(): unable to activate JACK client!";
2453 // Get the list of available ports.
2454 if ( shouldAutoconnect_ && (stream_.mode == OUTPUT || stream_.mode == DUPLEX) ) {
2456 ports = jack_get_ports( handle->client, handle->deviceName[0].c_str(), NULL, JackPortIsInput);
2457 if ( ports == NULL) {
2458 errorText_ = "RtApiJack::startStream(): error determining available JACK input ports!";
2462 // Now make the port connections. Since RtAudio wasn't designed to
2463 // allow the user to select particular channels of a device, we'll
2464 // just open the first "nChannels" ports with offset.
2465 for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
2467 if ( ports[ stream_.channelOffset[0] + i ] )
2468 result = jack_connect( handle->client, jack_port_name( handle->ports[0][i] ), ports[ stream_.channelOffset[0] + i ] );
2471 errorText_ = "RtApiJack::startStream(): error connecting output ports!";
2478 if ( shouldAutoconnect_ && (stream_.mode == INPUT || stream_.mode == DUPLEX) ) {
2480 ports = jack_get_ports( handle->client, handle->deviceName[1].c_str(), NULL, JackPortIsOutput );
2481 if ( ports == NULL) {
2482 errorText_ = "RtApiJack::startStream(): error determining available JACK output ports!";
2486 // Now make the port connections. See note above.
2487 for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
2489 if ( ports[ stream_.channelOffset[1] + i ] )
2490 result = jack_connect( handle->client, ports[ stream_.channelOffset[1] + i ], jack_port_name( handle->ports[1][i] ) );
2493 errorText_ = "RtApiJack::startStream(): error connecting input ports!";
2500 handle->drainCounter = 0;
2501 handle->internalDrain = false;
2502 stream_.state = STREAM_RUNNING;
2505 if ( result == 0 ) return;
2506 error( RtAudioError::SYSTEM_ERROR );
2509 void RtApiJack :: stopStream( void )
2512 if ( stream_.state == STREAM_STOPPED ) {
2513 errorText_ = "RtApiJack::stopStream(): the stream is already stopped!";
2514 error( RtAudioError::WARNING );
2518 JackHandle *handle = (JackHandle *) stream_.apiHandle;
2519 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
2521 if ( handle->drainCounter == 0 ) {
2522 handle->drainCounter = 2;
2523 pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
2527 jack_deactivate( handle->client );
2528 stream_.state = STREAM_STOPPED;
2531 void RtApiJack :: abortStream( void )
2534 if ( stream_.state == STREAM_STOPPED ) {
2535 errorText_ = "RtApiJack::abortStream(): the stream is already stopped!";
2536 error( RtAudioError::WARNING );
2540 JackHandle *handle = (JackHandle *) stream_.apiHandle;
2541 handle->drainCounter = 2;
2546 // This function will be called by a spawned thread when the user
2547 // callback function signals that the stream should be stopped or
2548 // aborted. It is necessary to handle it this way because the
2549 // callbackEvent() function must return before the jack_deactivate()
2550 // function will return.
2551 static void *jackStopStream( void *ptr )
2553 CallbackInfo *info = (CallbackInfo *) ptr;
2554 RtApiJack *object = (RtApiJack *) info->object;
2556 object->stopStream();
2557 pthread_exit( NULL );
2560 bool RtApiJack :: callbackEvent( unsigned long nframes )
2562 if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;
2563 if ( stream_.state == STREAM_CLOSED ) {
2564 errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";
2565 error( RtAudioError::WARNING );
2568 if ( stream_.bufferSize != nframes ) {
2569 errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!";
2570 error( RtAudioError::WARNING );
2574 CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
2575 JackHandle *handle = (JackHandle *) stream_.apiHandle;
2577 // Check if we were draining the stream and signal is finished.
2578 if ( handle->drainCounter > 3 ) {
2579 ThreadHandle threadId;
2581 stream_.state = STREAM_STOPPING;
2582 if ( handle->internalDrain == true )
2583 pthread_create( &threadId, NULL, jackStopStream, info );
2585 pthread_cond_signal( &handle->condition );
2589 // Invoke user callback first, to get fresh output data.
2590 if ( handle->drainCounter == 0 ) {
2591 RtAudioCallback callback = (RtAudioCallback) info->callback;
2592 double streamTime = getStreamTime();
2593 RtAudioStreamStatus status = 0;
2594 if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
2595 status |= RTAUDIO_OUTPUT_UNDERFLOW;
2596 handle->xrun[0] = false;
2598 if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
2599 status |= RTAUDIO_INPUT_OVERFLOW;
2600 handle->xrun[1] = false;
2602 int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
2603 stream_.bufferSize, streamTime, status, info->userData );
2604 if ( cbReturnValue == 2 ) {
2605 stream_.state = STREAM_STOPPING;
2606 handle->drainCounter = 2;
2608 pthread_create( &id, NULL, jackStopStream, info );
2611 else if ( cbReturnValue == 1 ) {
2612 handle->drainCounter = 1;
2613 handle->internalDrain = true;
2617 jack_default_audio_sample_t *jackbuffer;
2618 unsigned long bufferBytes = nframes * sizeof( jack_default_audio_sample_t );
2619 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
2621 if ( handle->drainCounter > 1 ) { // write zeros to the output stream
2623 for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
2624 jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
2625 memset( jackbuffer, 0, bufferBytes );
2629 else if ( stream_.doConvertBuffer[0] ) {
2631 convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
2633 for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
2634 jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
2635 memcpy( jackbuffer, &stream_.deviceBuffer[i*bufferBytes], bufferBytes );
2638 else { // no buffer conversion
2639 for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
2640 jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
2641 memcpy( jackbuffer, &stream_.userBuffer[0][i*bufferBytes], bufferBytes );
2646 // Don't bother draining input
2647 if ( handle->drainCounter ) {
2648 handle->drainCounter++;
2652 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
2654 if ( stream_.doConvertBuffer[1] ) {
2655 for ( unsigned int i=0; i<stream_.nDeviceChannels[1]; i++ ) {
2656 jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );
2657 memcpy( &stream_.deviceBuffer[i*bufferBytes], jackbuffer, bufferBytes );
2659 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
2661 else { // no buffer conversion
2662 for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
2663 jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );
2664 memcpy( &stream_.userBuffer[1][i*bufferBytes], jackbuffer, bufferBytes );
2670 RtApi::tickStreamTime();
2673 //******************** End of __UNIX_JACK__ *********************//
2676 #if defined(__WINDOWS_ASIO__) // ASIO API on Windows
2678 // The ASIO API is designed around a callback scheme, so this
2679 // implementation is similar to that used for OS-X CoreAudio and Linux
2680 // Jack. The primary constraint with ASIO is that it only allows
2681 // access to a single driver at a time. Thus, it is not possible to
2682 // have more than one simultaneous RtAudio stream.
2684 // This implementation also requires a number of external ASIO files
2685 // and a few global variables. The ASIO callback scheme does not
2686 // allow for the passing of user data, so we must create a global
2687 // pointer to our callbackInfo structure.
2689 // On unix systems, we make use of a pthread condition variable.
2690 // Since there is no equivalent in Windows, I hacked something based
2691 // on information found in
2692 // http://www.cs.wustl.edu/~schmidt/win32-cv-1.html.
2694 #include "asiosys.h"
2696 #include "iasiothiscallresolver.h"
2697 #include "asiodrivers.h"
2700 static AsioDrivers drivers;
2701 static ASIOCallbacks asioCallbacks;
2702 static ASIODriverInfo driverInfo;
2703 static CallbackInfo *asioCallbackInfo;
2704 static bool asioXRun;
2707 int drainCounter; // Tracks callback counts when draining
2708 bool internalDrain; // Indicates if stop is initiated from callback or not.
2709 ASIOBufferInfo *bufferInfos;
2713 :drainCounter(0), internalDrain(false), bufferInfos(0) {}
2716 // Function declarations (definitions at end of section)
2717 static const char* getAsioErrorString( ASIOError result );
2718 static void sampleRateChanged( ASIOSampleRate sRate );
2719 static long asioMessages( long selector, long value, void* message, double* opt );
2721 RtApiAsio :: RtApiAsio()
2723 // ASIO cannot run on a multi-threaded appartment. You can call
2724 // CoInitialize beforehand, but it must be for appartment threading
2725 // (in which case, CoInitilialize will return S_FALSE here).
2726 coInitialized_ = false;
2727 HRESULT hr = CoInitialize( NULL );
2729 errorText_ = "RtApiAsio::ASIO requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)";
2730 error( RtAudioError::WARNING );
2732 coInitialized_ = true;
2734 drivers.removeCurrentDriver();
2735 driverInfo.asioVersion = 2;
2737 // See note in DirectSound implementation about GetDesktopWindow().
2738 driverInfo.sysRef = GetForegroundWindow();
2741 RtApiAsio :: ~RtApiAsio()
2743 if ( stream_.state != STREAM_CLOSED ) closeStream();
2744 if ( coInitialized_ ) CoUninitialize();
2747 unsigned int RtApiAsio :: getDeviceCount( void )
2749 return (unsigned int) drivers.asioGetNumDev();
2752 RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device )
2754 RtAudio::DeviceInfo info;
2755 info.probed = false;
2758 unsigned int nDevices = getDeviceCount();
2759 if ( nDevices == 0 ) {
2760 errorText_ = "RtApiAsio::getDeviceInfo: no devices found!";
2761 error( RtAudioError::INVALID_USE );
2765 if ( device >= nDevices ) {
2766 errorText_ = "RtApiAsio::getDeviceInfo: device ID is invalid!";
2767 error( RtAudioError::INVALID_USE );
2771 // If a stream is already open, we cannot probe other devices. Thus, use the saved results.
2772 if ( stream_.state != STREAM_CLOSED ) {
2773 if ( device >= devices_.size() ) {
2774 errorText_ = "RtApiAsio::getDeviceInfo: device ID was not present before stream was opened.";
2775 error( RtAudioError::WARNING );
2778 return devices_[ device ];
2781 char driverName[32];
2782 ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 );
2783 if ( result != ASE_OK ) {
2784 errorStream_ << "RtApiAsio::getDeviceInfo: unable to get driver name (" << getAsioErrorString( result ) << ").";
2785 errorText_ = errorStream_.str();
2786 error( RtAudioError::WARNING );
2790 info.name = driverName;
2792 if ( !drivers.loadDriver( driverName ) ) {
2793 errorStream_ << "RtApiAsio::getDeviceInfo: unable to load driver (" << driverName << ").";
2794 errorText_ = errorStream_.str();
2795 error( RtAudioError::WARNING );
2799 result = ASIOInit( &driverInfo );
2800 if ( result != ASE_OK ) {
2801 errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
2802 errorText_ = errorStream_.str();
2803 error( RtAudioError::WARNING );
2807 // Determine the device channel information.
2808 long inputChannels, outputChannels;
2809 result = ASIOGetChannels( &inputChannels, &outputChannels );
2810 if ( result != ASE_OK ) {
2811 drivers.removeCurrentDriver();
2812 errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
2813 errorText_ = errorStream_.str();
2814 error( RtAudioError::WARNING );
2818 info.outputChannels = outputChannels;
2819 info.inputChannels = inputChannels;
2820 if ( info.outputChannels > 0 && info.inputChannels > 0 )
2821 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
2823 // Determine the supported sample rates.
2824 info.sampleRates.clear();
2825 for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {
2826 result = ASIOCanSampleRate( (ASIOSampleRate) SAMPLE_RATES[i] );
2827 if ( result == ASE_OK ) {
2828 info.sampleRates.push_back( SAMPLE_RATES[i] );
2830 if ( !info.preferredSampleRate || ( SAMPLE_RATES[i] <= 48000 && SAMPLE_RATES[i] > info.preferredSampleRate ) )
2831 info.preferredSampleRate = SAMPLE_RATES[i];
2835 // Determine supported data types ... just check first channel and assume rest are the same.
2836 ASIOChannelInfo channelInfo;
2837 channelInfo.channel = 0;
2838 channelInfo.isInput = true;
2839 if ( info.inputChannels <= 0 ) channelInfo.isInput = false;
2840 result = ASIOGetChannelInfo( &channelInfo );
2841 if ( result != ASE_OK ) {
2842 drivers.removeCurrentDriver();
2843 errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting driver channel info (" << driverName << ").";
2844 errorText_ = errorStream_.str();
2845 error( RtAudioError::WARNING );
2849 info.nativeFormats = 0;
2850 if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB )
2851 info.nativeFormats |= RTAUDIO_SINT16;
2852 else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB )
2853 info.nativeFormats |= RTAUDIO_SINT32;
2854 else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB )
2855 info.nativeFormats |= RTAUDIO_FLOAT32;
2856 else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB )
2857 info.nativeFormats |= RTAUDIO_FLOAT64;
2858 else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB )
2859 info.nativeFormats |= RTAUDIO_SINT24;
2861 if ( info.outputChannels > 0 )
2862 if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true;
2863 if ( info.inputChannels > 0 )
2864 if ( getDefaultInputDevice() == device ) info.isDefaultInput = true;
2867 drivers.removeCurrentDriver();
2871 static void bufferSwitch( long index, ASIOBool /*processNow*/ )
2873 RtApiAsio *object = (RtApiAsio *) asioCallbackInfo->object;
2874 object->callbackEvent( index );
2877 void RtApiAsio :: saveDeviceInfo( void )
2881 unsigned int nDevices = getDeviceCount();
2882 devices_.resize( nDevices );
2883 for ( unsigned int i=0; i<nDevices; i++ )
2884 devices_[i] = getDeviceInfo( i );
2887 bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
2888 unsigned int firstChannel, unsigned int sampleRate,
2889 RtAudioFormat format, unsigned int *bufferSize,
2890 RtAudio::StreamOptions *options )
2891 {////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
2893 bool isDuplexInput = mode == INPUT && stream_.mode == OUTPUT;
2895 // For ASIO, a duplex stream MUST use the same driver.
2896 if ( isDuplexInput && stream_.device[0] != device ) {
2897 errorText_ = "RtApiAsio::probeDeviceOpen: an ASIO duplex stream must use the same device for input and output!";
2901 char driverName[32];
2902 ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 );
2903 if ( result != ASE_OK ) {
2904 errorStream_ << "RtApiAsio::probeDeviceOpen: unable to get driver name (" << getAsioErrorString( result ) << ").";
2905 errorText_ = errorStream_.str();
2909 // Only load the driver once for duplex stream.
2910 if ( !isDuplexInput ) {
2911 // The getDeviceInfo() function will not work when a stream is open
2912 // because ASIO does not allow multiple devices to run at the same
2913 // time. Thus, we'll probe the system before opening a stream and
2914 // save the results for use by getDeviceInfo().
2915 this->saveDeviceInfo();
2917 if ( !drivers.loadDriver( driverName ) ) {
2918 errorStream_ << "RtApiAsio::probeDeviceOpen: unable to load driver (" << driverName << ").";
2919 errorText_ = errorStream_.str();
2923 result = ASIOInit( &driverInfo );
2924 if ( result != ASE_OK ) {
2925 errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
2926 errorText_ = errorStream_.str();
2931 // keep them before any "goto error", they are used for error cleanup + goto device boundary checks
2932 bool buffersAllocated = false;
2933 AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
2934 unsigned int nChannels;
2937 // Check the device channel count.
2938 long inputChannels, outputChannels;
2939 result = ASIOGetChannels( &inputChannels, &outputChannels );
2940 if ( result != ASE_OK ) {
2941 errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
2942 errorText_ = errorStream_.str();
2946 if ( ( mode == OUTPUT && (channels+firstChannel) > (unsigned int) outputChannels) ||
2947 ( mode == INPUT && (channels+firstChannel) > (unsigned int) inputChannels) ) {
2948 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested channel count (" << channels << ") + offset (" << firstChannel << ").";
2949 errorText_ = errorStream_.str();
2952 stream_.nDeviceChannels[mode] = channels;
2953 stream_.nUserChannels[mode] = channels;
2954 stream_.channelOffset[mode] = firstChannel;
2956 // Verify the sample rate is supported.
2957 result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate );
2958 if ( result != ASE_OK ) {
2959 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested sample rate (" << sampleRate << ").";
2960 errorText_ = errorStream_.str();
2964 // Get the current sample rate
2965 ASIOSampleRate currentRate;
2966 result = ASIOGetSampleRate( ¤tRate );
2967 if ( result != ASE_OK ) {
2968 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error getting sample rate.";
2969 errorText_ = errorStream_.str();
2973 // Set the sample rate only if necessary
2974 if ( currentRate != sampleRate ) {
2975 result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate );
2976 if ( result != ASE_OK ) {
2977 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ").";
2978 errorText_ = errorStream_.str();
2983 // Determine the driver data type.
2984 ASIOChannelInfo channelInfo;
2985 channelInfo.channel = 0;
2986 if ( mode == OUTPUT ) channelInfo.isInput = false;
2987 else channelInfo.isInput = true;
2988 result = ASIOGetChannelInfo( &channelInfo );
2989 if ( result != ASE_OK ) {
2990 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting data format.";
2991 errorText_ = errorStream_.str();
2995 // Assuming WINDOWS host is always little-endian.
2996 stream_.doByteSwap[mode] = false;
2997 stream_.userFormat = format;
2998 stream_.deviceFormat[mode] = 0;
2999 if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) {
3000 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
3001 if ( channelInfo.type == ASIOSTInt16MSB ) stream_.doByteSwap[mode] = true;
3003 else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) {
3004 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
3005 if ( channelInfo.type == ASIOSTInt32MSB ) stream_.doByteSwap[mode] = true;
3007 else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) {
3008 stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
3009 if ( channelInfo.type == ASIOSTFloat32MSB ) stream_.doByteSwap[mode] = true;
3011 else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) {
3012 stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
3013 if ( channelInfo.type == ASIOSTFloat64MSB ) stream_.doByteSwap[mode] = true;
3015 else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB ) {
3016 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
3017 if ( channelInfo.type == ASIOSTInt24MSB ) stream_.doByteSwap[mode] = true;
3020 if ( stream_.deviceFormat[mode] == 0 ) {
3021 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") data format not supported by RtAudio.";
3022 errorText_ = errorStream_.str();
3026 // Set the buffer size. For a duplex stream, this will end up
3027 // setting the buffer size based on the input constraints, which
3029 long minSize, maxSize, preferSize, granularity;
3030 result = ASIOGetBufferSize( &minSize, &maxSize, &preferSize, &granularity );
3031 if ( result != ASE_OK ) {
3032 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting buffer size.";
3033 errorText_ = errorStream_.str();
3037 if ( isDuplexInput ) {
3038 // When this is the duplex input (output was opened before), then we have to use the same
3039 // buffersize as the output, because it might use the preferred buffer size, which most
3040 // likely wasn't passed as input to this. The buffer sizes have to be identically anyway,
3041 // So instead of throwing an error, make them equal. The caller uses the reference
3042 // to the "bufferSize" param as usual to set up processing buffers.
3044 *bufferSize = stream_.bufferSize;
3047 if ( *bufferSize == 0 ) *bufferSize = preferSize;
3048 else if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;
3049 else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;
3050 else if ( granularity == -1 ) {
3051 // Make sure bufferSize is a power of two.
3052 int log2_of_min_size = 0;
3053 int log2_of_max_size = 0;
3055 for ( unsigned int i = 0; i < sizeof(long) * 8; i++ ) {
3056 if ( minSize & ((long)1 << i) ) log2_of_min_size = i;
3057 if ( maxSize & ((long)1 << i) ) log2_of_max_size = i;
3060 long min_delta = std::abs( (long)*bufferSize - ((long)1 << log2_of_min_size) );
3061 int min_delta_num = log2_of_min_size;
3063 for (int i = log2_of_min_size + 1; i <= log2_of_max_size; i++) {
3064 long current_delta = std::abs( (long)*bufferSize - ((long)1 << i) );
3065 if (current_delta < min_delta) {
3066 min_delta = current_delta;
3071 *bufferSize = ( (unsigned int)1 << min_delta_num );
3072 if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;
3073 else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;
3075 else if ( granularity != 0 ) {
3076 // Set to an even multiple of granularity, rounding up.
3077 *bufferSize = (*bufferSize + granularity-1) / granularity * granularity;
3082 // we don't use it anymore, see above!
3083 // Just left it here for the case...
3084 if ( isDuplexInput && stream_.bufferSize != *bufferSize ) {
3085 errorText_ = "RtApiAsio::probeDeviceOpen: input/output buffersize discrepancy!";
3090 stream_.bufferSize = *bufferSize;
3091 stream_.nBuffers = 2;
3093 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
3094 else stream_.userInterleaved = true;
3096 // ASIO always uses non-interleaved buffers.
3097 stream_.deviceInterleaved[mode] = false;
3099 // Allocate, if necessary, our AsioHandle structure for the stream.
3100 if ( handle == 0 ) {
3102 handle = new AsioHandle;
3104 catch ( std::bad_alloc& ) {
3105 errorText_ = "RtApiAsio::probeDeviceOpen: error allocating AsioHandle memory.";
3108 handle->bufferInfos = 0;
3110 // Create a manual-reset event.
3111 handle->condition = CreateEvent( NULL, // no security
3112 TRUE, // manual-reset
3113 FALSE, // non-signaled initially
3115 stream_.apiHandle = (void *) handle;
3118 // Create the ASIO internal buffers. Since RtAudio sets up input
3119 // and output separately, we'll have to dispose of previously
3120 // created output buffers for a duplex stream.
3121 if ( mode == INPUT && stream_.mode == OUTPUT ) {
3122 ASIODisposeBuffers();
3123 if ( handle->bufferInfos ) free( handle->bufferInfos );
3126 // Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure.
3128 nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
3129 handle->bufferInfos = (ASIOBufferInfo *) malloc( nChannels * sizeof(ASIOBufferInfo) );
3130 if ( handle->bufferInfos == NULL ) {
3131 errorStream_ << "RtApiAsio::probeDeviceOpen: error allocating bufferInfo memory for driver (" << driverName << ").";
3132 errorText_ = errorStream_.str();
3136 ASIOBufferInfo *infos;
3137 infos = handle->bufferInfos;
3138 for ( i=0; i<stream_.nDeviceChannels[0]; i++, infos++ ) {
3139 infos->isInput = ASIOFalse;
3140 infos->channelNum = i + stream_.channelOffset[0];
3141 infos->buffers[0] = infos->buffers[1] = 0;
3143 for ( i=0; i<stream_.nDeviceChannels[1]; i++, infos++ ) {
3144 infos->isInput = ASIOTrue;
3145 infos->channelNum = i + stream_.channelOffset[1];
3146 infos->buffers[0] = infos->buffers[1] = 0;
3149 // prepare for callbacks
3150 stream_.sampleRate = sampleRate;
3151 stream_.device[mode] = device;
3152 stream_.mode = isDuplexInput ? DUPLEX : mode;
3154 // store this class instance before registering callbacks, that are going to use it
3155 asioCallbackInfo = &stream_.callbackInfo;
3156 stream_.callbackInfo.object = (void *) this;
3158 // Set up the ASIO callback structure and create the ASIO data buffers.
3159 asioCallbacks.bufferSwitch = &bufferSwitch;
3160 asioCallbacks.sampleRateDidChange = &sampleRateChanged;
3161 asioCallbacks.asioMessage = &asioMessages;
3162 asioCallbacks.bufferSwitchTimeInfo = NULL;
3163 result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks );
3164 if ( result != ASE_OK ) {
3165 // Standard method failed. This can happen with strict/misbehaving drivers that return valid buffer size ranges
3166 // but only accept the preferred buffer size as parameter for ASIOCreateBuffers. eg. Creatives ASIO driver
3167 // in that case, let's be naïve and try that instead
3168 *bufferSize = preferSize;
3169 stream_.bufferSize = *bufferSize;
3170 result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks );
3173 if ( result != ASE_OK ) {
3174 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") creating buffers.";
3175 errorText_ = errorStream_.str();
3178 buffersAllocated = true;
3179 stream_.state = STREAM_STOPPED;
3181 // Set flags for buffer conversion.
3182 stream_.doConvertBuffer[mode] = false;
3183 if ( stream_.userFormat != stream_.deviceFormat[mode] )
3184 stream_.doConvertBuffer[mode] = true;
3185 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
3186 stream_.nUserChannels[mode] > 1 )
3187 stream_.doConvertBuffer[mode] = true;
3189 // Allocate necessary internal buffers
3190 unsigned long bufferBytes;
3191 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
3192 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
3193 if ( stream_.userBuffer[mode] == NULL ) {
3194 errorText_ = "RtApiAsio::probeDeviceOpen: error allocating user buffer memory.";
3198 if ( stream_.doConvertBuffer[mode] ) {
3200 bool makeBuffer = true;
3201 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
3202 if ( isDuplexInput && stream_.deviceBuffer ) {
3203 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
3204 if ( bufferBytes <= bytesOut ) makeBuffer = false;
3208 bufferBytes *= *bufferSize;
3209 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
3210 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
3211 if ( stream_.deviceBuffer == NULL ) {
3212 errorText_ = "RtApiAsio::probeDeviceOpen: error allocating device buffer memory.";
3218 // Determine device latencies
3219 long inputLatency, outputLatency;
3220 result = ASIOGetLatencies( &inputLatency, &outputLatency );
3221 if ( result != ASE_OK ) {
3222 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting latency.";
3223 errorText_ = errorStream_.str();
3224 error( RtAudioError::WARNING); // warn but don't fail
3227 stream_.latency[0] = outputLatency;
3228 stream_.latency[1] = inputLatency;
3231 // Setup the buffer conversion information structure. We don't use
3232 // buffers to do channel offsets, so we override that parameter
3234 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );
3239 if ( !isDuplexInput ) {
3240 // the cleanup for error in the duplex input, is done by RtApi::openStream
3241 // So we clean up for single channel only
3243 if ( buffersAllocated )
3244 ASIODisposeBuffers();
3246 drivers.removeCurrentDriver();
3249 CloseHandle( handle->condition );
3250 if ( handle->bufferInfos )
3251 free( handle->bufferInfos );
3254 stream_.apiHandle = 0;
3258 if ( stream_.userBuffer[mode] ) {
3259 free( stream_.userBuffer[mode] );
3260 stream_.userBuffer[mode] = 0;
3263 if ( stream_.deviceBuffer ) {
3264 free( stream_.deviceBuffer );
3265 stream_.deviceBuffer = 0;
3270 }////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
3272 void RtApiAsio :: closeStream()
3274 if ( stream_.state == STREAM_CLOSED ) {
3275 errorText_ = "RtApiAsio::closeStream(): no open stream to close!";
3276 error( RtAudioError::WARNING );
3280 if ( stream_.state == STREAM_RUNNING ) {
3281 stream_.state = STREAM_STOPPED;
3284 ASIODisposeBuffers();
3285 drivers.removeCurrentDriver();
3287 AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
3289 CloseHandle( handle->condition );
3290 if ( handle->bufferInfos )
3291 free( handle->bufferInfos );
3293 stream_.apiHandle = 0;
3296 for ( int i=0; i<2; i++ ) {
3297 if ( stream_.userBuffer[i] ) {
3298 free( stream_.userBuffer[i] );
3299 stream_.userBuffer[i] = 0;
3303 if ( stream_.deviceBuffer ) {
3304 free( stream_.deviceBuffer );
3305 stream_.deviceBuffer = 0;
3308 stream_.mode = UNINITIALIZED;
3309 stream_.state = STREAM_CLOSED;
3312 bool stopThreadCalled = false;
3314 void RtApiAsio :: startStream()
3317 if ( stream_.state == STREAM_RUNNING ) {
3318 errorText_ = "RtApiAsio::startStream(): the stream is already running!";
3319 error( RtAudioError::WARNING );
3323 AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
3324 ASIOError result = ASIOStart();
3325 if ( result != ASE_OK ) {
3326 errorStream_ << "RtApiAsio::startStream: error (" << getAsioErrorString( result ) << ") starting device.";
3327 errorText_ = errorStream_.str();
3331 handle->drainCounter = 0;
3332 handle->internalDrain = false;
3333 ResetEvent( handle->condition );
3334 stream_.state = STREAM_RUNNING;
3338 stopThreadCalled = false;
3340 if ( result == ASE_OK ) return;
3341 error( RtAudioError::SYSTEM_ERROR );
3344 void RtApiAsio :: stopStream()
3347 if ( stream_.state == STREAM_STOPPED ) {
3348 errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!";
3349 error( RtAudioError::WARNING );
3353 AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
3354 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
3355 if ( handle->drainCounter == 0 ) {
3356 handle->drainCounter = 2;
3357 WaitForSingleObject( handle->condition, INFINITE ); // block until signaled
3361 stream_.state = STREAM_STOPPED;
3363 ASIOError result = ASIOStop();
3364 if ( result != ASE_OK ) {
3365 errorStream_ << "RtApiAsio::stopStream: error (" << getAsioErrorString( result ) << ") stopping device.";
3366 errorText_ = errorStream_.str();
3369 if ( result == ASE_OK ) return;
3370 error( RtAudioError::SYSTEM_ERROR );
3373 void RtApiAsio :: abortStream()
3376 if ( stream_.state == STREAM_STOPPED ) {
3377 errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!";
3378 error( RtAudioError::WARNING );
3382 // The following lines were commented-out because some behavior was
3383 // noted where the device buffers need to be zeroed to avoid
3384 // continuing sound, even when the device buffers are completely
3385 // disposed. So now, calling abort is the same as calling stop.
3386 // AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
3387 // handle->drainCounter = 2;
3391 // This function will be called by a spawned thread when the user
3392 // callback function signals that the stream should be stopped or
3393 // aborted. It is necessary to handle it this way because the
3394 // callbackEvent() function must return before the ASIOStop()
3395 // function will return.
3396 static unsigned __stdcall asioStopStream( void *ptr )
3398 CallbackInfo *info = (CallbackInfo *) ptr;
3399 RtApiAsio *object = (RtApiAsio *) info->object;
3401 object->stopStream();
3406 bool RtApiAsio :: callbackEvent( long bufferIndex )
3408 if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;
3409 if ( stream_.state == STREAM_CLOSED ) {
3410 errorText_ = "RtApiAsio::callbackEvent(): the stream is closed ... this shouldn't happen!";
3411 error( RtAudioError::WARNING );
3415 CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
3416 AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
3418 // Check if we were draining the stream and signal if finished.
3419 if ( handle->drainCounter > 3 ) {
3421 stream_.state = STREAM_STOPPING;
3422 if ( handle->internalDrain == false )
3423 SetEvent( handle->condition );
3424 else { // spawn a thread to stop the stream
3426 stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream,
3427 &stream_.callbackInfo, 0, &threadId );
3432 // Invoke user callback to get fresh output data UNLESS we are
3434 if ( handle->drainCounter == 0 ) {
3435 RtAudioCallback callback = (RtAudioCallback) info->callback;
3436 double streamTime = getStreamTime();
3437 RtAudioStreamStatus status = 0;
3438 if ( stream_.mode != INPUT && asioXRun == true ) {
3439 status |= RTAUDIO_OUTPUT_UNDERFLOW;
3442 if ( stream_.mode != OUTPUT && asioXRun == true ) {
3443 status |= RTAUDIO_INPUT_OVERFLOW;
3446 int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
3447 stream_.bufferSize, streamTime, status, info->userData );
3448 if ( cbReturnValue == 2 ) {
3449 stream_.state = STREAM_STOPPING;
3450 handle->drainCounter = 2;
3452 stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream,
3453 &stream_.callbackInfo, 0, &threadId );
3456 else if ( cbReturnValue == 1 ) {
3457 handle->drainCounter = 1;
3458 handle->internalDrain = true;
3462 unsigned int nChannels, bufferBytes, i, j;
3463 nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
3464 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
3466 bufferBytes = stream_.bufferSize * formatBytes( stream_.deviceFormat[0] );
3468 if ( handle->drainCounter > 1 ) { // write zeros to the output stream
3470 for ( i=0, j=0; i<nChannels; i++ ) {
3471 if ( handle->bufferInfos[i].isInput != ASIOTrue )
3472 memset( handle->bufferInfos[i].buffers[bufferIndex], 0, bufferBytes );
3476 else if ( stream_.doConvertBuffer[0] ) {
3478 convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
3479 if ( stream_.doByteSwap[0] )
3480 byteSwapBuffer( stream_.deviceBuffer,
3481 stream_.bufferSize * stream_.nDeviceChannels[0],
3482 stream_.deviceFormat[0] );
3484 for ( i=0, j=0; i<nChannels; i++ ) {
3485 if ( handle->bufferInfos[i].isInput != ASIOTrue )
3486 memcpy( handle->bufferInfos[i].buffers[bufferIndex],
3487 &stream_.deviceBuffer[j++*bufferBytes], bufferBytes );
3493 if ( stream_.doByteSwap[0] )
3494 byteSwapBuffer( stream_.userBuffer[0],
3495 stream_.bufferSize * stream_.nUserChannels[0],
3496 stream_.userFormat );
3498 for ( i=0, j=0; i<nChannels; i++ ) {
3499 if ( handle->bufferInfos[i].isInput != ASIOTrue )
3500 memcpy( handle->bufferInfos[i].buffers[bufferIndex],
3501 &stream_.userBuffer[0][bufferBytes*j++], bufferBytes );
3507 // Don't bother draining input
3508 if ( handle->drainCounter ) {
3509 handle->drainCounter++;
3513 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
3515 bufferBytes = stream_.bufferSize * formatBytes(stream_.deviceFormat[1]);
3517 if (stream_.doConvertBuffer[1]) {
3519 // Always interleave ASIO input data.
3520 for ( i=0, j=0; i<nChannels; i++ ) {
3521 if ( handle->bufferInfos[i].isInput == ASIOTrue )
3522 memcpy( &stream_.deviceBuffer[j++*bufferBytes],
3523 handle->bufferInfos[i].buffers[bufferIndex],
3527 if ( stream_.doByteSwap[1] )
3528 byteSwapBuffer( stream_.deviceBuffer,
3529 stream_.bufferSize * stream_.nDeviceChannels[1],
3530 stream_.deviceFormat[1] );
3531 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
3535 for ( i=0, j=0; i<nChannels; i++ ) {
3536 if ( handle->bufferInfos[i].isInput == ASIOTrue ) {
3537 memcpy( &stream_.userBuffer[1][bufferBytes*j++],
3538 handle->bufferInfos[i].buffers[bufferIndex],
3543 if ( stream_.doByteSwap[1] )
3544 byteSwapBuffer( stream_.userBuffer[1],
3545 stream_.bufferSize * stream_.nUserChannels[1],
3546 stream_.userFormat );
3551 // The following call was suggested by Malte Clasen. While the API
3552 // documentation indicates it should not be required, some device
3553 // drivers apparently do not function correctly without it.
3556 RtApi::tickStreamTime();
3560 static void sampleRateChanged( ASIOSampleRate sRate )
3562 // The ASIO documentation says that this usually only happens during
3563 // external sync. Audio processing is not stopped by the driver,
3564 // actual sample rate might not have even changed, maybe only the
3565 // sample rate status of an AES/EBU or S/PDIF digital input at the
3568 RtApi *object = (RtApi *) asioCallbackInfo->object;
3570 object->stopStream();
3572 catch ( RtAudioError &exception ) {
3573 std::cerr << "\nRtApiAsio: sampleRateChanged() error (" << exception.getMessage() << ")!\n" << std::endl;
3577 std::cerr << "\nRtApiAsio: driver reports sample rate changed to " << sRate << " ... stream stopped!!!\n" << std::endl;
3580 static long asioMessages( long selector, long value, void* /*message*/, double* /*opt*/ )
3584 switch( selector ) {
3585 case kAsioSelectorSupported:
3586 if ( value == kAsioResetRequest
3587 || value == kAsioEngineVersion
3588 || value == kAsioResyncRequest
3589 || value == kAsioLatenciesChanged
3590 // The following three were added for ASIO 2.0, you don't
3591 // necessarily have to support them.
3592 || value == kAsioSupportsTimeInfo
3593 || value == kAsioSupportsTimeCode
3594 || value == kAsioSupportsInputMonitor)
3597 case kAsioResetRequest:
3598 // Defer the task and perform the reset of the driver during the
3599 // next "safe" situation. You cannot reset the driver right now,
3600 // as this code is called from the driver. Reset the driver is
3601 // done by completely destruct is. I.e. ASIOStop(),
3602 // ASIODisposeBuffers(), Destruction Afterwards you initialize the
3604 std::cerr << "\nRtApiAsio: driver reset requested!!!" << std::endl;
3607 case kAsioResyncRequest:
3608 // This informs the application that the driver encountered some
3609 // non-fatal data loss. It is used for synchronization purposes
3610 // of different media. Added mainly to work around the Win16Mutex
3611 // problems in Windows 95/98 with the Windows Multimedia system,
3612 // which could lose data because the Mutex was held too long by
3613 // another thread. However a driver can issue it in other
3615 // std::cerr << "\nRtApiAsio: driver resync requested!!!" << std::endl;
3619 case kAsioLatenciesChanged:
3620 // This will inform the host application that the drivers were
3621 // latencies changed. Beware, it this does not mean that the
3622 // buffer sizes have changed! You might need to update internal
3624 std::cerr << "\nRtApiAsio: driver latency may have changed!!!" << std::endl;
3627 case kAsioEngineVersion:
3628 // Return the supported ASIO version of the host application. If
3629 // a host application does not implement this selector, ASIO 1.0
3630 // is assumed by the driver.
3633 case kAsioSupportsTimeInfo:
3634 // Informs the driver whether the
3635 // asioCallbacks.bufferSwitchTimeInfo() callback is supported.
3636 // For compatibility with ASIO 1.0 drivers the host application
3637 // should always support the "old" bufferSwitch method, too.
3640 case kAsioSupportsTimeCode:
3641 // Informs the driver whether application is interested in time
3642 // code info. If an application does not need to know about time
3643 // code, the driver has less work to do.
3650 static const char* getAsioErrorString( ASIOError result )
3658 static const Messages m[] =
3660 { ASE_NotPresent, "Hardware input or output is not present or available." },
3661 { ASE_HWMalfunction, "Hardware is malfunctioning." },
3662 { ASE_InvalidParameter, "Invalid input parameter." },
3663 { ASE_InvalidMode, "Invalid mode." },
3664 { ASE_SPNotAdvancing, "Sample position not advancing." },
3665 { ASE_NoClock, "Sample clock or rate cannot be determined or is not present." },
3666 { ASE_NoMemory, "Not enough memory to complete the request." }
3669 for ( unsigned int i = 0; i < sizeof(m)/sizeof(m[0]); ++i )
3670 if ( m[i].value == result ) return m[i].message;
3672 return "Unknown error.";
3675 //******************** End of __WINDOWS_ASIO__ *********************//
3679 #if defined(__WINDOWS_WASAPI__) // Windows WASAPI API
3681 // Authored by Marcus Tomlinson <themarcustomlinson@gmail.com>, April 2014
3682 // - Introduces support for the Windows WASAPI API
3683 // - Aims to deliver bit streams to and from hardware at the lowest possible latency, via the absolute minimum buffer sizes required
3684 // - Provides flexible stream configuration to an otherwise strict and inflexible WASAPI interface
3685 // - Includes automatic internal conversion of sample rate and buffer size between hardware and the user
3690 #include <audioclient.h>
3692 #include <mmdeviceapi.h>
3693 #include <functiondiscoverykeys_devpkey.h>
3695 //=============================================================================
3697 #define SAFE_RELEASE( objectPtr )\
3700 objectPtr->Release();\
3704 typedef HANDLE ( __stdcall *TAvSetMmThreadCharacteristicsPtr )( LPCWSTR TaskName, LPDWORD TaskIndex );
3706 //-----------------------------------------------------------------------------
3708 // WASAPI dictates stream sample rate, format, channel count, and in some cases, buffer size.
3709 // Therefore we must perform all necessary conversions to user buffers in order to satisfy these
3710 // requirements. WasapiBuffer ring buffers are used between HwIn->UserIn and UserOut->HwOut to
3711 // provide intermediate storage for read / write synchronization.
3725 // sets the length of the internal ring buffer
3726 void setBufferSize( unsigned int bufferSize, unsigned int formatBytes ) {
3729 buffer_ = ( char* ) calloc( bufferSize, formatBytes );
3731 bufferSize_ = bufferSize;
3736 // attempt to push a buffer into the ring buffer at the current "in" index
3737 bool pushBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format )
3739 if ( !buffer || // incoming buffer is NULL
3740 bufferSize == 0 || // incoming buffer has no data
3741 bufferSize > bufferSize_ ) // incoming buffer too large
3746 unsigned int relOutIndex = outIndex_;
3747 unsigned int inIndexEnd = inIndex_ + bufferSize;
3748 if ( relOutIndex < inIndex_ && inIndexEnd >= bufferSize_ ) {
3749 relOutIndex += bufferSize_;
3752 // "in" index can end on the "out" index but cannot begin at it
3753 if ( inIndex_ <= relOutIndex && inIndexEnd > relOutIndex ) {
3754 return false; // not enough space between "in" index and "out" index
3757 // copy buffer from external to internal
3758 int fromZeroSize = inIndex_ + bufferSize - bufferSize_;
3759 fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize;
3760 int fromInSize = bufferSize - fromZeroSize;
3765 memcpy( &( ( char* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( char ) );
3766 memcpy( buffer_, &( ( char* ) buffer )[fromInSize], fromZeroSize * sizeof( char ) );
3768 case RTAUDIO_SINT16:
3769 memcpy( &( ( short* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( short ) );
3770 memcpy( buffer_, &( ( short* ) buffer )[fromInSize], fromZeroSize * sizeof( short ) );
3772 case RTAUDIO_SINT24:
3773 memcpy( &( ( S24* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( S24 ) );
3774 memcpy( buffer_, &( ( S24* ) buffer )[fromInSize], fromZeroSize * sizeof( S24 ) );
3776 case RTAUDIO_SINT32:
3777 memcpy( &( ( int* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( int ) );
3778 memcpy( buffer_, &( ( int* ) buffer )[fromInSize], fromZeroSize * sizeof( int ) );
3780 case RTAUDIO_FLOAT32:
3781 memcpy( &( ( float* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( float ) );
3782 memcpy( buffer_, &( ( float* ) buffer )[fromInSize], fromZeroSize * sizeof( float ) );
3784 case RTAUDIO_FLOAT64:
3785 memcpy( &( ( double* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( double ) );
3786 memcpy( buffer_, &( ( double* ) buffer )[fromInSize], fromZeroSize * sizeof( double ) );
3790 // update "in" index
3791 inIndex_ += bufferSize;
3792 inIndex_ %= bufferSize_;
3797 // attempt to pull a buffer from the ring buffer from the current "out" index
3798 bool pullBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format )
3800 if ( !buffer || // incoming buffer is NULL
3801 bufferSize == 0 || // incoming buffer has no data
3802 bufferSize > bufferSize_ ) // incoming buffer too large
3807 unsigned int relInIndex = inIndex_;
3808 unsigned int outIndexEnd = outIndex_ + bufferSize;
3809 if ( relInIndex < outIndex_ && outIndexEnd >= bufferSize_ ) {
3810 relInIndex += bufferSize_;
3813 // "out" index can begin at and end on the "in" index
3814 if ( outIndex_ < relInIndex && outIndexEnd > relInIndex ) {
3815 return false; // not enough space between "out" index and "in" index
3818 // copy buffer from internal to external
3819 int fromZeroSize = outIndex_ + bufferSize - bufferSize_;
3820 fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize;
3821 int fromOutSize = bufferSize - fromZeroSize;
3826 memcpy( buffer, &( ( char* ) buffer_ )[outIndex_], fromOutSize * sizeof( char ) );
3827 memcpy( &( ( char* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( char ) );
3829 case RTAUDIO_SINT16:
3830 memcpy( buffer, &( ( short* ) buffer_ )[outIndex_], fromOutSize * sizeof( short ) );
3831 memcpy( &( ( short* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( short ) );
3833 case RTAUDIO_SINT24:
3834 memcpy( buffer, &( ( S24* ) buffer_ )[outIndex_], fromOutSize * sizeof( S24 ) );
3835 memcpy( &( ( S24* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( S24 ) );
3837 case RTAUDIO_SINT32:
3838 memcpy( buffer, &( ( int* ) buffer_ )[outIndex_], fromOutSize * sizeof( int ) );
3839 memcpy( &( ( int* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( int ) );
3841 case RTAUDIO_FLOAT32:
3842 memcpy( buffer, &( ( float* ) buffer_ )[outIndex_], fromOutSize * sizeof( float ) );
3843 memcpy( &( ( float* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( float ) );
3845 case RTAUDIO_FLOAT64:
3846 memcpy( buffer, &( ( double* ) buffer_ )[outIndex_], fromOutSize * sizeof( double ) );
3847 memcpy( &( ( double* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( double ) );
3851 // update "out" index
3852 outIndex_ += bufferSize;
3853 outIndex_ %= bufferSize_;
3860 unsigned int bufferSize_;
3861 unsigned int inIndex_;
3862 unsigned int outIndex_;
3865 //-----------------------------------------------------------------------------
3867 // In order to satisfy WASAPI's buffer requirements, we need a means of converting sample rate
3868 // between HW and the user. The convertBufferWasapi function is used to perform this conversion
3869 // between HwIn->UserIn and UserOut->HwOut during the stream callback loop.
3870 // This sample rate converter works best with conversions between one rate and its multiple.
3871 void convertBufferWasapi( char* outBuffer,
3872 const char* inBuffer,
3873 const unsigned int& channelCount,
3874 const unsigned int& inSampleRate,
3875 const unsigned int& outSampleRate,
3876 const unsigned int& inSampleCount,
3877 unsigned int& outSampleCount,
3878 const RtAudioFormat& format )
3880 // calculate the new outSampleCount and relative sampleStep
3881 float sampleRatio = ( float ) outSampleRate / inSampleRate;
3882 float sampleRatioInv = ( float ) 1 / sampleRatio;
3883 float sampleStep = 1.0f / sampleRatio;
3884 float inSampleFraction = 0.0f;
3886 outSampleCount = ( unsigned int ) std::roundf( inSampleCount * sampleRatio );
3888 // if inSampleRate is a multiple of outSampleRate (or vice versa) there's no need to interpolate
3889 if ( floor( sampleRatio ) == sampleRatio || floor( sampleRatioInv ) == sampleRatioInv )
3891 // frame-by-frame, copy each relative input sample into it's corresponding output sample
3892 for ( unsigned int outSample = 0; outSample < outSampleCount; outSample++ )
3894 unsigned int inSample = ( unsigned int ) inSampleFraction;
3899 memcpy( &( ( char* ) outBuffer )[ outSample * channelCount ], &( ( char* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( char ) );
3901 case RTAUDIO_SINT16:
3902 memcpy( &( ( short* ) outBuffer )[ outSample * channelCount ], &( ( short* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( short ) );
3904 case RTAUDIO_SINT24:
3905 memcpy( &( ( S24* ) outBuffer )[ outSample * channelCount ], &( ( S24* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( S24 ) );
3907 case RTAUDIO_SINT32:
3908 memcpy( &( ( int* ) outBuffer )[ outSample * channelCount ], &( ( int* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( int ) );
3910 case RTAUDIO_FLOAT32:
3911 memcpy( &( ( float* ) outBuffer )[ outSample * channelCount ], &( ( float* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( float ) );
3913 case RTAUDIO_FLOAT64:
3914 memcpy( &( ( double* ) outBuffer )[ outSample * channelCount ], &( ( double* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( double ) );
3918 // jump to next in sample
3919 inSampleFraction += sampleStep;
3922 else // else interpolate
3924 // frame-by-frame, copy each relative input sample into it's corresponding output sample
3925 for ( unsigned int outSample = 0; outSample < outSampleCount; outSample++ )
3927 unsigned int inSample = ( unsigned int ) inSampleFraction;
3928 float inSampleDec = inSampleFraction - inSample;
3929 unsigned int frameInSample = inSample * channelCount;
3930 unsigned int frameOutSample = outSample * channelCount;
3936 for ( unsigned int channel = 0; channel < channelCount; channel++ )
3938 char fromSample = ( ( char* ) inBuffer )[ frameInSample + channel ];
3939 char toSample = ( ( char* ) inBuffer )[ frameInSample + channelCount + channel ];
3940 char sampleDiff = ( char ) ( ( toSample - fromSample ) * inSampleDec );
3941 ( ( char* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;
3945 case RTAUDIO_SINT16:
3947 for ( unsigned int channel = 0; channel < channelCount; channel++ )
3949 short fromSample = ( ( short* ) inBuffer )[ frameInSample + channel ];
3950 short toSample = ( ( short* ) inBuffer )[ frameInSample + channelCount + channel ];
3951 short sampleDiff = ( short ) ( ( toSample - fromSample ) * inSampleDec );
3952 ( ( short* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;
3956 case RTAUDIO_SINT24:
3958 for ( unsigned int channel = 0; channel < channelCount; channel++ )
3960 int fromSample = ( ( S24* ) inBuffer )[ frameInSample + channel ].asInt();
3961 int toSample = ( ( S24* ) inBuffer )[ frameInSample + channelCount + channel ].asInt();
3962 int sampleDiff = ( int ) ( ( toSample - fromSample ) * inSampleDec );
3963 ( ( S24* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;
3967 case RTAUDIO_SINT32:
3969 for ( unsigned int channel = 0; channel < channelCount; channel++ )
3971 int fromSample = ( ( int* ) inBuffer )[ frameInSample + channel ];
3972 int toSample = ( ( int* ) inBuffer )[ frameInSample + channelCount + channel ];
3973 int sampleDiff = ( int ) ( ( toSample - fromSample ) * inSampleDec );
3974 ( ( int* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;
3978 case RTAUDIO_FLOAT32:
3980 for ( unsigned int channel = 0; channel < channelCount; channel++ )
3982 float fromSample = ( ( float* ) inBuffer )[ frameInSample + channel ];
3983 float toSample = ( ( float* ) inBuffer )[ frameInSample + channelCount + channel ];
3984 float sampleDiff = ( toSample - fromSample ) * inSampleDec;
3985 ( ( float* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;
3989 case RTAUDIO_FLOAT64:
3991 for ( unsigned int channel = 0; channel < channelCount; channel++ )
3993 double fromSample = ( ( double* ) inBuffer )[ frameInSample + channel ];
3994 double toSample = ( ( double* ) inBuffer )[ frameInSample + channelCount + channel ];
3995 double sampleDiff = ( toSample - fromSample ) * inSampleDec;
3996 ( ( double* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;
4002 // jump to next in sample
4003 inSampleFraction += sampleStep;
4008 //-----------------------------------------------------------------------------
4010 // A structure to hold various information related to the WASAPI implementation.
4013 IAudioClient* captureAudioClient;
4014 IAudioClient* renderAudioClient;
4015 IAudioCaptureClient* captureClient;
4016 IAudioRenderClient* renderClient;
4017 HANDLE captureEvent;
4021 : captureAudioClient( NULL ),
4022 renderAudioClient( NULL ),
4023 captureClient( NULL ),
4024 renderClient( NULL ),
4025 captureEvent( NULL ),
4026 renderEvent( NULL ) {}
4029 //=============================================================================
4031 RtApiWasapi::RtApiWasapi()
4032 : coInitialized_( false ), deviceEnumerator_( NULL )
4034 // WASAPI can run either apartment or multi-threaded
4035 HRESULT hr = CoInitialize( NULL );
4036 if ( !FAILED( hr ) )
4037 coInitialized_ = true;
4039 // Instantiate device enumerator
4040 hr = CoCreateInstance( __uuidof( MMDeviceEnumerator ), NULL,
4041 CLSCTX_ALL, __uuidof( IMMDeviceEnumerator ),
4042 ( void** ) &deviceEnumerator_ );
4044 if ( FAILED( hr ) ) {
4045 errorText_ = "RtApiWasapi::RtApiWasapi: Unable to instantiate device enumerator";
4046 error( RtAudioError::DRIVER_ERROR );
4050 //-----------------------------------------------------------------------------
4052 RtApiWasapi::~RtApiWasapi()
4054 if ( stream_.state != STREAM_CLOSED )
4057 SAFE_RELEASE( deviceEnumerator_ );
4059 // If this object previously called CoInitialize()
4060 if ( coInitialized_ )
4064 //=============================================================================
4066 unsigned int RtApiWasapi::getDeviceCount( void )
4068 unsigned int captureDeviceCount = 0;
4069 unsigned int renderDeviceCount = 0;
4071 IMMDeviceCollection* captureDevices = NULL;
4072 IMMDeviceCollection* renderDevices = NULL;
4074 // Count capture devices
4076 HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
4077 if ( FAILED( hr ) ) {
4078 errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device collection.";
4082 hr = captureDevices->GetCount( &captureDeviceCount );
4083 if ( FAILED( hr ) ) {
4084 errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device count.";
4088 // Count render devices
4089 hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
4090 if ( FAILED( hr ) ) {
4091 errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device collection.";
4095 hr = renderDevices->GetCount( &renderDeviceCount );
4096 if ( FAILED( hr ) ) {
4097 errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device count.";
4102 // release all references
4103 SAFE_RELEASE( captureDevices );
4104 SAFE_RELEASE( renderDevices );
4106 if ( errorText_.empty() )
4107 return captureDeviceCount + renderDeviceCount;
4109 error( RtAudioError::DRIVER_ERROR );
4113 //-----------------------------------------------------------------------------
4115 RtAudio::DeviceInfo RtApiWasapi::getDeviceInfo( unsigned int device )
4117 RtAudio::DeviceInfo info;
4118 unsigned int captureDeviceCount = 0;
4119 unsigned int renderDeviceCount = 0;
4120 std::string defaultDeviceName;
4121 bool isCaptureDevice = false;
4123 PROPVARIANT deviceNameProp;
4124 PROPVARIANT defaultDeviceNameProp;
4126 IMMDeviceCollection* captureDevices = NULL;
4127 IMMDeviceCollection* renderDevices = NULL;
4128 IMMDevice* devicePtr = NULL;
4129 IMMDevice* defaultDevicePtr = NULL;
4130 IAudioClient* audioClient = NULL;
4131 IPropertyStore* devicePropStore = NULL;
4132 IPropertyStore* defaultDevicePropStore = NULL;
4134 WAVEFORMATEX* deviceFormat = NULL;
4135 WAVEFORMATEX* closestMatchFormat = NULL;
4138 info.probed = false;
4140 // Count capture devices
4142 RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
4143 HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
4144 if ( FAILED( hr ) ) {
4145 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device collection.";
4149 hr = captureDevices->GetCount( &captureDeviceCount );
4150 if ( FAILED( hr ) ) {
4151 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device count.";
4155 // Count render devices
4156 hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
4157 if ( FAILED( hr ) ) {
4158 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device collection.";
4162 hr = renderDevices->GetCount( &renderDeviceCount );
4163 if ( FAILED( hr ) ) {
4164 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device count.";
4168 // validate device index
4169 if ( device >= captureDeviceCount + renderDeviceCount ) {
4170 errorText_ = "RtApiWasapi::getDeviceInfo: Invalid device index.";
4171 errorType = RtAudioError::INVALID_USE;
4175 // determine whether index falls within capture or render devices
4176 if ( device >= renderDeviceCount ) {
4177 hr = captureDevices->Item( device - renderDeviceCount, &devicePtr );
4178 if ( FAILED( hr ) ) {
4179 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device handle.";
4182 isCaptureDevice = true;
4185 hr = renderDevices->Item( device, &devicePtr );
4186 if ( FAILED( hr ) ) {
4187 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device handle.";
4190 isCaptureDevice = false;
4193 // get default device name
4194 if ( isCaptureDevice ) {
4195 hr = deviceEnumerator_->GetDefaultAudioEndpoint( eCapture, eConsole, &defaultDevicePtr );
4196 if ( FAILED( hr ) ) {
4197 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default capture device handle.";
4202 hr = deviceEnumerator_->GetDefaultAudioEndpoint( eRender, eConsole, &defaultDevicePtr );
4203 if ( FAILED( hr ) ) {
4204 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default render device handle.";
4209 hr = defaultDevicePtr->OpenPropertyStore( STGM_READ, &defaultDevicePropStore );
4210 if ( FAILED( hr ) ) {
4211 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open default device property store.";
4214 PropVariantInit( &defaultDeviceNameProp );
4216 hr = defaultDevicePropStore->GetValue( PKEY_Device_FriendlyName, &defaultDeviceNameProp );
4217 if ( FAILED( hr ) ) {
4218 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default device property: PKEY_Device_FriendlyName.";
4222 defaultDeviceName = convertCharPointerToStdString(defaultDeviceNameProp.pwszVal);
4225 hr = devicePtr->OpenPropertyStore( STGM_READ, &devicePropStore );
4226 if ( FAILED( hr ) ) {
4227 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open device property store.";
4231 PropVariantInit( &deviceNameProp );
4233 hr = devicePropStore->GetValue( PKEY_Device_FriendlyName, &deviceNameProp );
4234 if ( FAILED( hr ) ) {
4235 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device property: PKEY_Device_FriendlyName.";
4239 info.name =convertCharPointerToStdString(deviceNameProp.pwszVal);
4242 if ( isCaptureDevice ) {
4243 info.isDefaultInput = info.name == defaultDeviceName;
4244 info.isDefaultOutput = false;
4247 info.isDefaultInput = false;
4248 info.isDefaultOutput = info.name == defaultDeviceName;
4252 hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, NULL, ( void** ) &audioClient );
4253 if ( FAILED( hr ) ) {
4254 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device audio client.";
4258 hr = audioClient->GetMixFormat( &deviceFormat );
4259 if ( FAILED( hr ) ) {
4260 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device mix format.";
4264 if ( isCaptureDevice ) {
4265 info.inputChannels = deviceFormat->nChannels;
4266 info.outputChannels = 0;
4267 info.duplexChannels = 0;
4270 info.inputChannels = 0;
4271 info.outputChannels = deviceFormat->nChannels;
4272 info.duplexChannels = 0;
4276 info.sampleRates.clear();
4278 // allow support for all sample rates as we have a built-in sample rate converter
4279 for ( unsigned int i = 0; i < MAX_SAMPLE_RATES; i++ ) {
4280 info.sampleRates.push_back( SAMPLE_RATES[i] );
4282 info.preferredSampleRate = deviceFormat->nSamplesPerSec;
4285 info.nativeFormats = 0;
4287 if ( deviceFormat->wFormatTag == WAVE_FORMAT_IEEE_FLOAT ||
4288 ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
4289 ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_IEEE_FLOAT ) )
4291 if ( deviceFormat->wBitsPerSample == 32 ) {
4292 info.nativeFormats |= RTAUDIO_FLOAT32;
4294 else if ( deviceFormat->wBitsPerSample == 64 ) {
4295 info.nativeFormats |= RTAUDIO_FLOAT64;
4298 else if ( deviceFormat->wFormatTag == WAVE_FORMAT_PCM ||
4299 ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
4300 ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_PCM ) )
4302 if ( deviceFormat->wBitsPerSample == 8 ) {
4303 info.nativeFormats |= RTAUDIO_SINT8;
4305 else if ( deviceFormat->wBitsPerSample == 16 ) {
4306 info.nativeFormats |= RTAUDIO_SINT16;
4308 else if ( deviceFormat->wBitsPerSample == 24 ) {
4309 info.nativeFormats |= RTAUDIO_SINT24;
4311 else if ( deviceFormat->wBitsPerSample == 32 ) {
4312 info.nativeFormats |= RTAUDIO_SINT32;
4320 // release all references
4321 PropVariantClear( &deviceNameProp );
4322 PropVariantClear( &defaultDeviceNameProp );
4324 SAFE_RELEASE( captureDevices );
4325 SAFE_RELEASE( renderDevices );
4326 SAFE_RELEASE( devicePtr );
4327 SAFE_RELEASE( defaultDevicePtr );
4328 SAFE_RELEASE( audioClient );
4329 SAFE_RELEASE( devicePropStore );
4330 SAFE_RELEASE( defaultDevicePropStore );
4332 CoTaskMemFree( deviceFormat );
4333 CoTaskMemFree( closestMatchFormat );
4335 if ( !errorText_.empty() )
4340 //-----------------------------------------------------------------------------
4342 unsigned int RtApiWasapi::getDefaultOutputDevice( void )
4344 for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {
4345 if ( getDeviceInfo( i ).isDefaultOutput ) {
4353 //-----------------------------------------------------------------------------
4355 unsigned int RtApiWasapi::getDefaultInputDevice( void )
4357 for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {
4358 if ( getDeviceInfo( i ).isDefaultInput ) {
4366 //-----------------------------------------------------------------------------
4368 void RtApiWasapi::closeStream( void )
4370 if ( stream_.state == STREAM_CLOSED ) {
4371 errorText_ = "RtApiWasapi::closeStream: No open stream to close.";
4372 error( RtAudioError::WARNING );
4376 if ( stream_.state != STREAM_STOPPED )
4379 // clean up stream memory
4380 SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient )
4381 SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient )
4383 SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureClient )
4384 SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderClient )
4386 if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent )
4387 CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent );
4389 if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent )
4390 CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent );
4392 delete ( WasapiHandle* ) stream_.apiHandle;
4393 stream_.apiHandle = NULL;
4395 for ( int i = 0; i < 2; i++ ) {
4396 if ( stream_.userBuffer[i] ) {
4397 free( stream_.userBuffer[i] );
4398 stream_.userBuffer[i] = 0;
4402 if ( stream_.deviceBuffer ) {
4403 free( stream_.deviceBuffer );
4404 stream_.deviceBuffer = 0;
4407 // update stream state
4408 stream_.state = STREAM_CLOSED;
4411 //-----------------------------------------------------------------------------
4413 void RtApiWasapi::startStream( void )
4417 if ( stream_.state == STREAM_RUNNING ) {
4418 errorText_ = "RtApiWasapi::startStream: The stream is already running.";
4419 error( RtAudioError::WARNING );
4423 // update stream state
4424 stream_.state = STREAM_RUNNING;
4426 // create WASAPI stream thread
4427 stream_.callbackInfo.thread = ( ThreadHandle ) CreateThread( NULL, 0, runWasapiThread, this, CREATE_SUSPENDED, NULL );
4429 if ( !stream_.callbackInfo.thread ) {
4430 errorText_ = "RtApiWasapi::startStream: Unable to instantiate callback thread.";
4431 error( RtAudioError::THREAD_ERROR );
4434 SetThreadPriority( ( void* ) stream_.callbackInfo.thread, stream_.callbackInfo.priority );
4435 ResumeThread( ( void* ) stream_.callbackInfo.thread );
4439 //-----------------------------------------------------------------------------
4441 void RtApiWasapi::stopStream( void )
4445 if ( stream_.state == STREAM_STOPPED ) {
4446 errorText_ = "RtApiWasapi::stopStream: The stream is already stopped.";
4447 error( RtAudioError::WARNING );
4451 // inform stream thread by setting stream state to STREAM_STOPPING
4452 stream_.state = STREAM_STOPPING;
4454 // wait until stream thread is stopped
4455 while( stream_.state != STREAM_STOPPED ) {
4459 // Wait for the last buffer to play before stopping.
4460 Sleep( 1000 * stream_.bufferSize / stream_.sampleRate );
4462 // stop capture client if applicable
4463 if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) {
4464 HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop();
4465 if ( FAILED( hr ) ) {
4466 errorText_ = "RtApiWasapi::stopStream: Unable to stop capture stream.";
4467 error( RtAudioError::DRIVER_ERROR );
4472 // stop render client if applicable
4473 if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) {
4474 HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop();
4475 if ( FAILED( hr ) ) {
4476 errorText_ = "RtApiWasapi::stopStream: Unable to stop render stream.";
4477 error( RtAudioError::DRIVER_ERROR );
4482 // close thread handle
4483 if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
4484 errorText_ = "RtApiWasapi::stopStream: Unable to close callback thread.";
4485 error( RtAudioError::THREAD_ERROR );
4489 stream_.callbackInfo.thread = (ThreadHandle) NULL;
4492 //-----------------------------------------------------------------------------
4494 void RtApiWasapi::abortStream( void )
4498 if ( stream_.state == STREAM_STOPPED ) {
4499 errorText_ = "RtApiWasapi::abortStream: The stream is already stopped.";
4500 error( RtAudioError::WARNING );
4504 // inform stream thread by setting stream state to STREAM_STOPPING
4505 stream_.state = STREAM_STOPPING;
4507 // wait until stream thread is stopped
4508 while ( stream_.state != STREAM_STOPPED ) {
4512 // stop capture client if applicable
4513 if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) {
4514 HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop();
4515 if ( FAILED( hr ) ) {
4516 errorText_ = "RtApiWasapi::abortStream: Unable to stop capture stream.";
4517 error( RtAudioError::DRIVER_ERROR );
4522 // stop render client if applicable
4523 if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) {
4524 HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop();
4525 if ( FAILED( hr ) ) {
4526 errorText_ = "RtApiWasapi::abortStream: Unable to stop render stream.";
4527 error( RtAudioError::DRIVER_ERROR );
4532 // close thread handle
4533 if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
4534 errorText_ = "RtApiWasapi::abortStream: Unable to close callback thread.";
4535 error( RtAudioError::THREAD_ERROR );
4539 stream_.callbackInfo.thread = (ThreadHandle) NULL;
4542 //-----------------------------------------------------------------------------
4544 bool RtApiWasapi::probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
4545 unsigned int firstChannel, unsigned int sampleRate,
4546 RtAudioFormat format, unsigned int* bufferSize,
4547 RtAudio::StreamOptions* options )
4549 bool methodResult = FAILURE;
4550 unsigned int captureDeviceCount = 0;
4551 unsigned int renderDeviceCount = 0;
4553 IMMDeviceCollection* captureDevices = NULL;
4554 IMMDeviceCollection* renderDevices = NULL;
4555 IMMDevice* devicePtr = NULL;
4556 WAVEFORMATEX* deviceFormat = NULL;
4557 unsigned int bufferBytes;
4558 stream_.state = STREAM_STOPPED;
4560 // create API Handle if not already created
4561 if ( !stream_.apiHandle )
4562 stream_.apiHandle = ( void* ) new WasapiHandle();
4564 // Count capture devices
4566 RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
4567 HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
4568 if ( FAILED( hr ) ) {
4569 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device collection.";
4573 hr = captureDevices->GetCount( &captureDeviceCount );
4574 if ( FAILED( hr ) ) {
4575 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device count.";
4579 // Count render devices
4580 hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
4581 if ( FAILED( hr ) ) {
4582 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device collection.";
4586 hr = renderDevices->GetCount( &renderDeviceCount );
4587 if ( FAILED( hr ) ) {
4588 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device count.";
4592 // validate device index
4593 if ( device >= captureDeviceCount + renderDeviceCount ) {
4594 errorType = RtAudioError::INVALID_USE;
4595 errorText_ = "RtApiWasapi::probeDeviceOpen: Invalid device index.";
4599 // determine whether index falls within capture or render devices
4600 if ( device >= renderDeviceCount ) {
4601 if ( mode != INPUT ) {
4602 errorType = RtAudioError::INVALID_USE;
4603 errorText_ = "RtApiWasapi::probeDeviceOpen: Capture device selected as output device.";
4607 // retrieve captureAudioClient from devicePtr
4608 IAudioClient*& captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
4610 hr = captureDevices->Item( device - renderDeviceCount, &devicePtr );
4611 if ( FAILED( hr ) ) {
4612 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device handle.";
4616 hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
4617 NULL, ( void** ) &captureAudioClient );
4618 if ( FAILED( hr ) ) {
4619 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client.";
4623 hr = captureAudioClient->GetMixFormat( &deviceFormat );
4624 if ( FAILED( hr ) ) {
4625 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format.";
4629 stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
4630 captureAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
4633 if ( mode != OUTPUT ) {
4634 errorType = RtAudioError::INVALID_USE;
4635 errorText_ = "RtApiWasapi::probeDeviceOpen: Render device selected as input device.";
4639 // retrieve renderAudioClient from devicePtr
4640 IAudioClient*& renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
4642 hr = renderDevices->Item( device, &devicePtr );
4643 if ( FAILED( hr ) ) {
4644 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device handle.";
4648 hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
4649 NULL, ( void** ) &renderAudioClient );
4650 if ( FAILED( hr ) ) {
4651 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client.";
4655 hr = renderAudioClient->GetMixFormat( &deviceFormat );
4656 if ( FAILED( hr ) ) {
4657 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format.";
4661 stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
4662 renderAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
4666 if ( ( stream_.mode == OUTPUT && mode == INPUT ) ||
4667 ( stream_.mode == INPUT && mode == OUTPUT ) ) {
4668 stream_.mode = DUPLEX;
4671 stream_.mode = mode;
4674 stream_.device[mode] = device;
4675 stream_.doByteSwap[mode] = false;
4676 stream_.sampleRate = sampleRate;
4677 stream_.bufferSize = *bufferSize;
4678 stream_.nBuffers = 1;
4679 stream_.nUserChannels[mode] = channels;
4680 stream_.channelOffset[mode] = firstChannel;
4681 stream_.userFormat = format;
4682 stream_.deviceFormat[mode] = getDeviceInfo( device ).nativeFormats;
4684 if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
4685 stream_.userInterleaved = false;
4687 stream_.userInterleaved = true;
4688 stream_.deviceInterleaved[mode] = true;
4690 // Set flags for buffer conversion.
4691 stream_.doConvertBuffer[mode] = false;
4692 if ( stream_.userFormat != stream_.deviceFormat[mode] ||
4693 stream_.nUserChannels != stream_.nDeviceChannels )
4694 stream_.doConvertBuffer[mode] = true;
4695 else if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
4696 stream_.nUserChannels[mode] > 1 )
4697 stream_.doConvertBuffer[mode] = true;
4699 if ( stream_.doConvertBuffer[mode] )
4700 setConvertInfo( mode, 0 );
4702 // Allocate necessary internal buffers
4703 bufferBytes = stream_.nUserChannels[mode] * stream_.bufferSize * formatBytes( stream_.userFormat );
4705 stream_.userBuffer[mode] = ( char* ) calloc( bufferBytes, 1 );
4706 if ( !stream_.userBuffer[mode] ) {
4707 errorType = RtAudioError::MEMORY_ERROR;
4708 errorText_ = "RtApiWasapi::probeDeviceOpen: Error allocating user buffer memory.";
4712 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME )
4713 stream_.callbackInfo.priority = 15;
4715 stream_.callbackInfo.priority = 0;
4717 ///! TODO: RTAUDIO_MINIMIZE_LATENCY // Provide stream buffers directly to callback
4718 ///! TODO: RTAUDIO_HOG_DEVICE // Exclusive mode
4720 methodResult = SUCCESS;
4724 SAFE_RELEASE( captureDevices );
4725 SAFE_RELEASE( renderDevices );
4726 SAFE_RELEASE( devicePtr );
4727 CoTaskMemFree( deviceFormat );
4729 // if method failed, close the stream
4730 if ( methodResult == FAILURE )
4733 if ( !errorText_.empty() )
4735 return methodResult;
4738 //=============================================================================
4740 DWORD WINAPI RtApiWasapi::runWasapiThread( void* wasapiPtr )
4743 ( ( RtApiWasapi* ) wasapiPtr )->wasapiThread();
4748 DWORD WINAPI RtApiWasapi::stopWasapiThread( void* wasapiPtr )
4751 ( ( RtApiWasapi* ) wasapiPtr )->stopStream();
4756 DWORD WINAPI RtApiWasapi::abortWasapiThread( void* wasapiPtr )
4759 ( ( RtApiWasapi* ) wasapiPtr )->abortStream();
4764 //-----------------------------------------------------------------------------
4766 void RtApiWasapi::wasapiThread()
4768 // as this is a new thread, we must CoInitialize it
4769 CoInitialize( NULL );
4773 IAudioClient* captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
4774 IAudioClient* renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
4775 IAudioCaptureClient* captureClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureClient;
4776 IAudioRenderClient* renderClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderClient;
4777 HANDLE captureEvent = ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent;
4778 HANDLE renderEvent = ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent;
4780 WAVEFORMATEX* captureFormat = NULL;
4781 WAVEFORMATEX* renderFormat = NULL;
4782 float captureSrRatio = 0.0f;
4783 float renderSrRatio = 0.0f;
4784 WasapiBuffer captureBuffer;
4785 WasapiBuffer renderBuffer;
4787 // declare local stream variables
4788 RtAudioCallback callback = ( RtAudioCallback ) stream_.callbackInfo.callback;
4789 BYTE* streamBuffer = NULL;
4790 unsigned long captureFlags = 0;
4791 unsigned int bufferFrameCount = 0;
4792 unsigned int numFramesPadding = 0;
4793 unsigned int convBufferSize = 0;
4794 bool callbackPushed = false;
4795 bool callbackPulled = false;
4796 bool callbackStopped = false;
4797 int callbackResult = 0;
4799 // convBuffer is used to store converted buffers between WASAPI and the user
4800 char* convBuffer = NULL;
4801 unsigned int convBuffSize = 0;
4802 unsigned int deviceBuffSize = 0;
4805 RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
4807 // Attempt to assign "Pro Audio" characteristic to thread
4808 HMODULE AvrtDll = LoadLibrary( (LPCTSTR) "AVRT.dll" );
4810 DWORD taskIndex = 0;
4811 TAvSetMmThreadCharacteristicsPtr AvSetMmThreadCharacteristicsPtr = ( TAvSetMmThreadCharacteristicsPtr ) GetProcAddress( AvrtDll, "AvSetMmThreadCharacteristicsW" );
4812 AvSetMmThreadCharacteristicsPtr( L"Pro Audio", &taskIndex );
4813 FreeLibrary( AvrtDll );
4816 // start capture stream if applicable
4817 if ( captureAudioClient ) {
4818 hr = captureAudioClient->GetMixFormat( &captureFormat );
4819 if ( FAILED( hr ) ) {
4820 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
4824 captureSrRatio = ( ( float ) captureFormat->nSamplesPerSec / stream_.sampleRate );
4826 // initialize capture stream according to desire buffer size
4827 float desiredBufferSize = stream_.bufferSize * captureSrRatio;
4828 REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / captureFormat->nSamplesPerSec );
4830 if ( !captureClient ) {
4831 hr = captureAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
4832 AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
4833 desiredBufferPeriod,
4834 desiredBufferPeriod,
4837 if ( FAILED( hr ) ) {
4838 errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize capture audio client.";
4842 hr = captureAudioClient->GetService( __uuidof( IAudioCaptureClient ),
4843 ( void** ) &captureClient );
4844 if ( FAILED( hr ) ) {
4845 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture client handle.";
4849 // configure captureEvent to trigger on every available capture buffer
4850 captureEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
4851 if ( !captureEvent ) {
4852 errorType = RtAudioError::SYSTEM_ERROR;
4853 errorText_ = "RtApiWasapi::wasapiThread: Unable to create capture event.";
4857 hr = captureAudioClient->SetEventHandle( captureEvent );
4858 if ( FAILED( hr ) ) {
4859 errorText_ = "RtApiWasapi::wasapiThread: Unable to set capture event handle.";
4863 ( ( WasapiHandle* ) stream_.apiHandle )->captureClient = captureClient;
4864 ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent = captureEvent;
4867 unsigned int inBufferSize = 0;
4868 hr = captureAudioClient->GetBufferSize( &inBufferSize );
4869 if ( FAILED( hr ) ) {
4870 errorText_ = "RtApiWasapi::wasapiThread: Unable to get capture buffer size.";
4874 // scale outBufferSize according to stream->user sample rate ratio
4875 unsigned int outBufferSize = ( unsigned int ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT];
4876 inBufferSize *= stream_.nDeviceChannels[INPUT];
4878 // set captureBuffer size
4879 captureBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[INPUT] ) );
4881 // reset the capture stream
4882 hr = captureAudioClient->Reset();
4883 if ( FAILED( hr ) ) {
4884 errorText_ = "RtApiWasapi::wasapiThread: Unable to reset capture stream.";
4888 // start the capture stream
4889 hr = captureAudioClient->Start();
4890 if ( FAILED( hr ) ) {
4891 errorText_ = "RtApiWasapi::wasapiThread: Unable to start capture stream.";
4896 // start render stream if applicable
4897 if ( renderAudioClient ) {
4898 hr = renderAudioClient->GetMixFormat( &renderFormat );
4899 if ( FAILED( hr ) ) {
4900 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
4904 renderSrRatio = ( ( float ) renderFormat->nSamplesPerSec / stream_.sampleRate );
4906 // initialize render stream according to desire buffer size
4907 float desiredBufferSize = stream_.bufferSize * renderSrRatio;
4908 REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / renderFormat->nSamplesPerSec );
4910 if ( !renderClient ) {
4911 hr = renderAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
4912 AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
4913 desiredBufferPeriod,
4914 desiredBufferPeriod,
4917 if ( FAILED( hr ) ) {
4918 errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize render audio client.";
4922 hr = renderAudioClient->GetService( __uuidof( IAudioRenderClient ),
4923 ( void** ) &renderClient );
4924 if ( FAILED( hr ) ) {
4925 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render client handle.";
4929 // configure renderEvent to trigger on every available render buffer
4930 renderEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
4931 if ( !renderEvent ) {
4932 errorType = RtAudioError::SYSTEM_ERROR;
4933 errorText_ = "RtApiWasapi::wasapiThread: Unable to create render event.";
4937 hr = renderAudioClient->SetEventHandle( renderEvent );
4938 if ( FAILED( hr ) ) {
4939 errorText_ = "RtApiWasapi::wasapiThread: Unable to set render event handle.";
4943 ( ( WasapiHandle* ) stream_.apiHandle )->renderClient = renderClient;
4944 ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent = renderEvent;
4947 unsigned int outBufferSize = 0;
4948 hr = renderAudioClient->GetBufferSize( &outBufferSize );
4949 if ( FAILED( hr ) ) {
4950 errorText_ = "RtApiWasapi::wasapiThread: Unable to get render buffer size.";
4954 // scale inBufferSize according to user->stream sample rate ratio
4955 unsigned int inBufferSize = ( unsigned int ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT];
4956 outBufferSize *= stream_.nDeviceChannels[OUTPUT];
4958 // set renderBuffer size
4959 renderBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[OUTPUT] ) );
4961 // reset the render stream
4962 hr = renderAudioClient->Reset();
4963 if ( FAILED( hr ) ) {
4964 errorText_ = "RtApiWasapi::wasapiThread: Unable to reset render stream.";
4968 // start the render stream
4969 hr = renderAudioClient->Start();
4970 if ( FAILED( hr ) ) {
4971 errorText_ = "RtApiWasapi::wasapiThread: Unable to start render stream.";
4976 if ( stream_.mode == INPUT ) {
4977 convBuffSize = ( size_t ) std::roundf( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
4978 deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
4980 else if ( stream_.mode == OUTPUT ) {
4981 convBuffSize = ( size_t ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
4982 deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
4984 else if ( stream_.mode == DUPLEX ) {
4985 convBuffSize = std::max( ( size_t ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
4986 ( size_t ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
4987 deviceBuffSize = std::max( stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
4988 stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
4991 convBuffer = ( char* ) malloc( convBuffSize );
4992 stream_.deviceBuffer = ( char* ) malloc( deviceBuffSize );
4993 if ( !convBuffer || !stream_.deviceBuffer ) {
4994 errorType = RtAudioError::MEMORY_ERROR;
4995 errorText_ = "RtApiWasapi::wasapiThread: Error allocating device buffer memory.";
4999 // stream process loop
5000 while ( stream_.state != STREAM_STOPPING ) {
5001 if ( !callbackPulled ) {
5004 // 1. Pull callback buffer from inputBuffer
5005 // 2. If 1. was successful: Convert callback buffer to user sample rate and channel count
5006 // Convert callback buffer to user format
5008 if ( captureAudioClient ) {
5009 // Pull callback buffer from inputBuffer
5010 callbackPulled = captureBuffer.pullBuffer( convBuffer,
5011 ( unsigned int ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT],
5012 stream_.deviceFormat[INPUT] );
5014 if ( callbackPulled ) {
5015 // Convert callback buffer to user sample rate
5016 convertBufferWasapi( stream_.deviceBuffer,
5018 stream_.nDeviceChannels[INPUT],
5019 captureFormat->nSamplesPerSec,
5021 ( unsigned int ) ( stream_.bufferSize * captureSrRatio ),
5023 stream_.deviceFormat[INPUT] );
5025 if ( stream_.doConvertBuffer[INPUT] ) {
5026 // Convert callback buffer to user format
5027 convertBuffer( stream_.userBuffer[INPUT],
5028 stream_.deviceBuffer,
5029 stream_.convertInfo[INPUT] );
5032 // no further conversion, simple copy deviceBuffer to userBuffer
5033 memcpy( stream_.userBuffer[INPUT],
5034 stream_.deviceBuffer,
5035 stream_.bufferSize * stream_.nUserChannels[INPUT] * formatBytes( stream_.userFormat ) );
5040 // if there is no capture stream, set callbackPulled flag
5041 callbackPulled = true;
5046 // 1. Execute user callback method
5047 // 2. Handle return value from callback
5049 // if callback has not requested the stream to stop
5050 if ( callbackPulled && !callbackStopped ) {
5051 // Execute user callback method
5052 callbackResult = callback( stream_.userBuffer[OUTPUT],
5053 stream_.userBuffer[INPUT],
5056 captureFlags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY ? RTAUDIO_INPUT_OVERFLOW : 0,
5057 stream_.callbackInfo.userData );
5059 // Handle return value from callback
5060 if ( callbackResult == 1 ) {
5061 // instantiate a thread to stop this thread
5062 HANDLE threadHandle = CreateThread( NULL, 0, stopWasapiThread, this, 0, NULL );
5063 if ( !threadHandle ) {
5064 errorType = RtAudioError::THREAD_ERROR;
5065 errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream stop thread.";
5068 else if ( !CloseHandle( threadHandle ) ) {
5069 errorType = RtAudioError::THREAD_ERROR;
5070 errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream stop thread handle.";
5074 callbackStopped = true;
5076 else if ( callbackResult == 2 ) {
5077 // instantiate a thread to stop this thread
5078 HANDLE threadHandle = CreateThread( NULL, 0, abortWasapiThread, this, 0, NULL );
5079 if ( !threadHandle ) {
5080 errorType = RtAudioError::THREAD_ERROR;
5081 errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream abort thread.";
5084 else if ( !CloseHandle( threadHandle ) ) {
5085 errorType = RtAudioError::THREAD_ERROR;
5086 errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream abort thread handle.";
5090 callbackStopped = true;
5097 // 1. Convert callback buffer to stream format
5098 // 2. Convert callback buffer to stream sample rate and channel count
5099 // 3. Push callback buffer into outputBuffer
5101 if ( renderAudioClient && callbackPulled ) {
5102 if ( stream_.doConvertBuffer[OUTPUT] ) {
5103 // Convert callback buffer to stream format
5104 convertBuffer( stream_.deviceBuffer,
5105 stream_.userBuffer[OUTPUT],
5106 stream_.convertInfo[OUTPUT] );
5110 // Convert callback buffer to stream sample rate
5111 convertBufferWasapi( convBuffer,
5112 stream_.deviceBuffer,
5113 stream_.nDeviceChannels[OUTPUT],
5115 renderFormat->nSamplesPerSec,
5118 stream_.deviceFormat[OUTPUT] );
5120 // Push callback buffer into outputBuffer
5121 callbackPushed = renderBuffer.pushBuffer( convBuffer,
5122 convBufferSize * stream_.nDeviceChannels[OUTPUT],
5123 stream_.deviceFormat[OUTPUT] );
5126 // if there is no render stream, set callbackPushed flag
5127 callbackPushed = true;
5132 // 1. Get capture buffer from stream
5133 // 2. Push capture buffer into inputBuffer
5134 // 3. If 2. was successful: Release capture buffer
5136 if ( captureAudioClient ) {
5137 // if the callback input buffer was not pulled from captureBuffer, wait for next capture event
5138 if ( !callbackPulled ) {
5139 WaitForSingleObject( captureEvent, INFINITE );
5142 // Get capture buffer from stream
5143 hr = captureClient->GetBuffer( &streamBuffer,
5145 &captureFlags, NULL, NULL );
5146 if ( FAILED( hr ) ) {
5147 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture buffer.";
5151 if ( bufferFrameCount != 0 ) {
5152 // Push capture buffer into inputBuffer
5153 if ( captureBuffer.pushBuffer( ( char* ) streamBuffer,
5154 bufferFrameCount * stream_.nDeviceChannels[INPUT],
5155 stream_.deviceFormat[INPUT] ) )
5157 // Release capture buffer
5158 hr = captureClient->ReleaseBuffer( bufferFrameCount );
5159 if ( FAILED( hr ) ) {
5160 errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
5166 // Inform WASAPI that capture was unsuccessful
5167 hr = captureClient->ReleaseBuffer( 0 );
5168 if ( FAILED( hr ) ) {
5169 errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
5176 // Inform WASAPI that capture was unsuccessful
5177 hr = captureClient->ReleaseBuffer( 0 );
5178 if ( FAILED( hr ) ) {
5179 errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
5187 // 1. Get render buffer from stream
5188 // 2. Pull next buffer from outputBuffer
5189 // 3. If 2. was successful: Fill render buffer with next buffer
5190 // Release render buffer
5192 if ( renderAudioClient ) {
5193 // if the callback output buffer was not pushed to renderBuffer, wait for next render event
5194 if ( callbackPulled && !callbackPushed ) {
5195 WaitForSingleObject( renderEvent, INFINITE );
5198 // Get render buffer from stream
5199 hr = renderAudioClient->GetBufferSize( &bufferFrameCount );
5200 if ( FAILED( hr ) ) {
5201 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer size.";
5205 hr = renderAudioClient->GetCurrentPadding( &numFramesPadding );
5206 if ( FAILED( hr ) ) {
5207 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer padding.";
5211 bufferFrameCount -= numFramesPadding;
5213 if ( bufferFrameCount != 0 ) {
5214 hr = renderClient->GetBuffer( bufferFrameCount, &streamBuffer );
5215 if ( FAILED( hr ) ) {
5216 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer.";
5220 // Pull next buffer from outputBuffer
5221 // Fill render buffer with next buffer
5222 if ( renderBuffer.pullBuffer( ( char* ) streamBuffer,
5223 bufferFrameCount * stream_.nDeviceChannels[OUTPUT],
5224 stream_.deviceFormat[OUTPUT] ) )
5226 // Release render buffer
5227 hr = renderClient->ReleaseBuffer( bufferFrameCount, 0 );
5228 if ( FAILED( hr ) ) {
5229 errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
5235 // Inform WASAPI that render was unsuccessful
5236 hr = renderClient->ReleaseBuffer( 0, 0 );
5237 if ( FAILED( hr ) ) {
5238 errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
5245 // Inform WASAPI that render was unsuccessful
5246 hr = renderClient->ReleaseBuffer( 0, 0 );
5247 if ( FAILED( hr ) ) {
5248 errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
5254 // if the callback buffer was pushed renderBuffer reset callbackPulled flag
5255 if ( callbackPushed ) {
5256 callbackPulled = false;
5258 RtApi::tickStreamTime();
5265 CoTaskMemFree( captureFormat );
5266 CoTaskMemFree( renderFormat );
5268 free ( convBuffer );
5272 // update stream state
5273 stream_.state = STREAM_STOPPED;
5275 if ( errorText_.empty() )
5281 //******************** End of __WINDOWS_WASAPI__ *********************//
5285 #if defined(__WINDOWS_DS__) // Windows DirectSound API
5287 // Modified by Robin Davies, October 2005
5288 // - Improvements to DirectX pointer chasing.
5289 // - Bug fix for non-power-of-two Asio granularity used by Edirol PCR-A30.
5290 // - Auto-call CoInitialize for DSOUND and ASIO platforms.
5291 // Various revisions for RtAudio 4.0 by Gary Scavone, April 2007
5292 // Changed device query structure for RtAudio 4.0.7, January 2010
5294 #include <mmsystem.h>
5298 #include <algorithm>
5300 #if defined(__MINGW32__)
5301 // missing from latest mingw winapi
5302 #define WAVE_FORMAT_96M08 0x00010000 /* 96 kHz, Mono, 8-bit */
5303 #define WAVE_FORMAT_96S08 0x00020000 /* 96 kHz, Stereo, 8-bit */
5304 #define WAVE_FORMAT_96M16 0x00040000 /* 96 kHz, Mono, 16-bit */
5305 #define WAVE_FORMAT_96S16 0x00080000 /* 96 kHz, Stereo, 16-bit */
5308 #define MINIMUM_DEVICE_BUFFER_SIZE 32768
5310 #ifdef _MSC_VER // if Microsoft Visual C++
5311 #pragma comment( lib, "winmm.lib" ) // then, auto-link winmm.lib. Otherwise, it has to be added manually.
5314 static inline DWORD dsPointerBetween( DWORD pointer, DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize )
5316 if ( pointer > bufferSize ) pointer -= bufferSize;
5317 if ( laterPointer < earlierPointer ) laterPointer += bufferSize;
5318 if ( pointer < earlierPointer ) pointer += bufferSize;
5319 return pointer >= earlierPointer && pointer < laterPointer;
5322 // A structure to hold various information related to the DirectSound
5323 // API implementation.
5325 unsigned int drainCounter; // Tracks callback counts when draining
5326 bool internalDrain; // Indicates if stop is initiated from callback or not.
5330 UINT bufferPointer[2];
5331 DWORD dsBufferSize[2];
5332 DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by.
5336 :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; }
5339 // Declarations for utility functions, callbacks, and structures
5340 // specific to the DirectSound implementation.
5341 static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
5342 LPCTSTR description,
5346 static const char* getErrorString( int code );
5348 static unsigned __stdcall callbackHandler( void *ptr );
5357 : found(false) { validId[0] = false; validId[1] = false; }
5360 struct DsProbeData {
5362 std::vector<struct DsDevice>* dsDevices;
5365 RtApiDs :: RtApiDs()
5367 // Dsound will run both-threaded. If CoInitialize fails, then just
5368 // accept whatever the mainline chose for a threading model.
5369 coInitialized_ = false;
5370 HRESULT hr = CoInitialize( NULL );
5371 if ( !FAILED( hr ) ) coInitialized_ = true;
5374 RtApiDs :: ~RtApiDs()
5376 if ( stream_.state != STREAM_CLOSED ) closeStream();
5377 if ( coInitialized_ ) CoUninitialize(); // balanced call.
5380 // The DirectSound default output is always the first device.
5381 unsigned int RtApiDs :: getDefaultOutputDevice( void )
5386 // The DirectSound default input is always the first input device,
5387 // which is the first capture device enumerated.
5388 unsigned int RtApiDs :: getDefaultInputDevice( void )
5393 unsigned int RtApiDs :: getDeviceCount( void )
5395 // Set query flag for previously found devices to false, so that we
5396 // can check for any devices that have disappeared.
5397 for ( unsigned int i=0; i<dsDevices.size(); i++ )
5398 dsDevices[i].found = false;
5400 // Query DirectSound devices.
5401 struct DsProbeData probeInfo;
5402 probeInfo.isInput = false;
5403 probeInfo.dsDevices = &dsDevices;
5404 HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );
5405 if ( FAILED( result ) ) {
5406 errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!";
5407 errorText_ = errorStream_.str();
5408 error( RtAudioError::WARNING );
5411 // Query DirectSoundCapture devices.
5412 probeInfo.isInput = true;
5413 result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );
5414 if ( FAILED( result ) ) {
5415 errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!";
5416 errorText_ = errorStream_.str();
5417 error( RtAudioError::WARNING );
5420 // Clean out any devices that may have disappeared (code update submitted by Eli Zehngut).
5421 for ( unsigned int i=0; i<dsDevices.size(); ) {
5422 if ( dsDevices[i].found == false ) dsDevices.erase( dsDevices.begin() + i );
5426 return static_cast<unsigned int>(dsDevices.size());
5429 RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device )
5431 RtAudio::DeviceInfo info;
5432 info.probed = false;
5434 if ( dsDevices.size() == 0 ) {
5435 // Force a query of all devices
5437 if ( dsDevices.size() == 0 ) {
5438 errorText_ = "RtApiDs::getDeviceInfo: no devices found!";
5439 error( RtAudioError::INVALID_USE );
5444 if ( device >= dsDevices.size() ) {
5445 errorText_ = "RtApiDs::getDeviceInfo: device ID is invalid!";
5446 error( RtAudioError::INVALID_USE );
5451 if ( dsDevices[ device ].validId[0] == false ) goto probeInput;
5453 LPDIRECTSOUND output;
5455 result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );
5456 if ( FAILED( result ) ) {
5457 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!";
5458 errorText_ = errorStream_.str();
5459 error( RtAudioError::WARNING );
5463 outCaps.dwSize = sizeof( outCaps );
5464 result = output->GetCaps( &outCaps );
5465 if ( FAILED( result ) ) {
5467 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting capabilities!";
5468 errorText_ = errorStream_.str();
5469 error( RtAudioError::WARNING );
5473 // Get output channel information.
5474 info.outputChannels = ( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1;
5476 // Get sample rate information.
5477 info.sampleRates.clear();
5478 for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
5479 if ( SAMPLE_RATES[k] >= (unsigned int) outCaps.dwMinSecondarySampleRate &&
5480 SAMPLE_RATES[k] <= (unsigned int) outCaps.dwMaxSecondarySampleRate ) {
5481 info.sampleRates.push_back( SAMPLE_RATES[k] );
5483 if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
5484 info.preferredSampleRate = SAMPLE_RATES[k];
5488 // Get format information.
5489 if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT ) info.nativeFormats |= RTAUDIO_SINT16;
5490 if ( outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) info.nativeFormats |= RTAUDIO_SINT8;
5494 if ( getDefaultOutputDevice() == device )
5495 info.isDefaultOutput = true;
5497 if ( dsDevices[ device ].validId[1] == false ) {
5498 info.name = dsDevices[ device ].name;
5505 LPDIRECTSOUNDCAPTURE input;
5506 result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );
5507 if ( FAILED( result ) ) {
5508 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";
5509 errorText_ = errorStream_.str();
5510 error( RtAudioError::WARNING );
5515 inCaps.dwSize = sizeof( inCaps );
5516 result = input->GetCaps( &inCaps );
5517 if ( FAILED( result ) ) {
5519 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsDevices[ device ].name << ")!";
5520 errorText_ = errorStream_.str();
5521 error( RtAudioError::WARNING );
5525 // Get input channel information.
5526 info.inputChannels = inCaps.dwChannels;
5528 // Get sample rate and format information.
5529 std::vector<unsigned int> rates;
5530 if ( inCaps.dwChannels >= 2 ) {
5531 if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.nativeFormats |= RTAUDIO_SINT16;
5532 if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.nativeFormats |= RTAUDIO_SINT16;
5533 if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.nativeFormats |= RTAUDIO_SINT16;
5534 if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.nativeFormats |= RTAUDIO_SINT16;
5535 if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.nativeFormats |= RTAUDIO_SINT8;
5536 if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.nativeFormats |= RTAUDIO_SINT8;
5537 if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.nativeFormats |= RTAUDIO_SINT8;
5538 if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.nativeFormats |= RTAUDIO_SINT8;
5540 if ( info.nativeFormats & RTAUDIO_SINT16 ) {
5541 if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) rates.push_back( 11025 );
5542 if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) rates.push_back( 22050 );
5543 if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) rates.push_back( 44100 );
5544 if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) rates.push_back( 96000 );
5546 else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
5547 if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) rates.push_back( 11025 );
5548 if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) rates.push_back( 22050 );
5549 if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) rates.push_back( 44100 );
5550 if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) rates.push_back( 96000 );
5553 else if ( inCaps.dwChannels == 1 ) {
5554 if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.nativeFormats |= RTAUDIO_SINT16;
5555 if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.nativeFormats |= RTAUDIO_SINT16;
5556 if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.nativeFormats |= RTAUDIO_SINT16;
5557 if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.nativeFormats |= RTAUDIO_SINT16;
5558 if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.nativeFormats |= RTAUDIO_SINT8;
5559 if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.nativeFormats |= RTAUDIO_SINT8;
5560 if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.nativeFormats |= RTAUDIO_SINT8;
5561 if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.nativeFormats |= RTAUDIO_SINT8;
5563 if ( info.nativeFormats & RTAUDIO_SINT16 ) {
5564 if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) rates.push_back( 11025 );
5565 if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) rates.push_back( 22050 );
5566 if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) rates.push_back( 44100 );
5567 if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) rates.push_back( 96000 );
5569 else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
5570 if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) rates.push_back( 11025 );
5571 if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) rates.push_back( 22050 );
5572 if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) rates.push_back( 44100 );
5573 if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) rates.push_back( 96000 );
5576 else info.inputChannels = 0; // technically, this would be an error
5580 if ( info.inputChannels == 0 ) return info;
5582 // Copy the supported rates to the info structure but avoid duplication.
5584 for ( unsigned int i=0; i<rates.size(); i++ ) {
5586 for ( unsigned int j=0; j<info.sampleRates.size(); j++ ) {
5587 if ( rates[i] == info.sampleRates[j] ) {
5592 if ( found == false ) info.sampleRates.push_back( rates[i] );
5594 std::sort( info.sampleRates.begin(), info.sampleRates.end() );
5596 // If device opens for both playback and capture, we determine the channels.
5597 if ( info.outputChannels > 0 && info.inputChannels > 0 )
5598 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
5600 if ( device == 0 ) info.isDefaultInput = true;
5602 // Copy name and return.
5603 info.name = dsDevices[ device ].name;
5608 bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
5609 unsigned int firstChannel, unsigned int sampleRate,
5610 RtAudioFormat format, unsigned int *bufferSize,
5611 RtAudio::StreamOptions *options )
5613 if ( channels + firstChannel > 2 ) {
5614 errorText_ = "RtApiDs::probeDeviceOpen: DirectSound does not support more than 2 channels per device.";
5618 size_t nDevices = dsDevices.size();
5619 if ( nDevices == 0 ) {
5620 // This should not happen because a check is made before this function is called.
5621 errorText_ = "RtApiDs::probeDeviceOpen: no devices found!";
5625 if ( device >= nDevices ) {
5626 // This should not happen because a check is made before this function is called.
5627 errorText_ = "RtApiDs::probeDeviceOpen: device ID is invalid!";
5631 if ( mode == OUTPUT ) {
5632 if ( dsDevices[ device ].validId[0] == false ) {
5633 errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support output!";
5634 errorText_ = errorStream_.str();
5638 else { // mode == INPUT
5639 if ( dsDevices[ device ].validId[1] == false ) {
5640 errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support input!";
5641 errorText_ = errorStream_.str();
5646 // According to a note in PortAudio, using GetDesktopWindow()
5647 // instead of GetForegroundWindow() is supposed to avoid problems
5648 // that occur when the application's window is not the foreground
5649 // window. Also, if the application window closes before the
5650 // DirectSound buffer, DirectSound can crash. In the past, I had
5651 // problems when using GetDesktopWindow() but it seems fine now
5652 // (January 2010). I'll leave it commented here.
5653 // HWND hWnd = GetForegroundWindow();
5654 HWND hWnd = GetDesktopWindow();
5656 // Check the numberOfBuffers parameter and limit the lowest value to
5657 // two. This is a judgement call and a value of two is probably too
5658 // low for capture, but it should work for playback.
5660 if ( options ) nBuffers = options->numberOfBuffers;
5661 if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) nBuffers = 2;
5662 if ( nBuffers < 2 ) nBuffers = 3;
5664 // Check the lower range of the user-specified buffer size and set
5665 // (arbitrarily) to a lower bound of 32.
5666 if ( *bufferSize < 32 ) *bufferSize = 32;
5668 // Create the wave format structure. The data format setting will
5669 // be determined later.
5670 WAVEFORMATEX waveFormat;
5671 ZeroMemory( &waveFormat, sizeof(WAVEFORMATEX) );
5672 waveFormat.wFormatTag = WAVE_FORMAT_PCM;
5673 waveFormat.nChannels = channels + firstChannel;
5674 waveFormat.nSamplesPerSec = (unsigned long) sampleRate;
5676 // Determine the device buffer size. By default, we'll use the value
5677 // defined above (32K), but we will grow it to make allowances for
5678 // very large software buffer sizes.
5679 DWORD dsBufferSize = MINIMUM_DEVICE_BUFFER_SIZE;
5680 DWORD dsPointerLeadTime = 0;
5682 void *ohandle = 0, *bhandle = 0;
5684 if ( mode == OUTPUT ) {
5686 LPDIRECTSOUND output;
5687 result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );
5688 if ( FAILED( result ) ) {
5689 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!";
5690 errorText_ = errorStream_.str();
5695 outCaps.dwSize = sizeof( outCaps );
5696 result = output->GetCaps( &outCaps );
5697 if ( FAILED( result ) ) {
5699 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting capabilities (" << dsDevices[ device ].name << ")!";
5700 errorText_ = errorStream_.str();
5704 // Check channel information.
5705 if ( channels + firstChannel == 2 && !( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ) {
5706 errorStream_ << "RtApiDs::getDeviceInfo: the output device (" << dsDevices[ device ].name << ") does not support stereo playback.";
5707 errorText_ = errorStream_.str();
5711 // Check format information. Use 16-bit format unless not
5712 // supported or user requests 8-bit.
5713 if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT &&
5714 !( format == RTAUDIO_SINT8 && outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) ) {
5715 waveFormat.wBitsPerSample = 16;
5716 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
5719 waveFormat.wBitsPerSample = 8;
5720 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
5722 stream_.userFormat = format;
5724 // Update wave format structure and buffer information.
5725 waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
5726 waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
5727 dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
5729 // If the user wants an even bigger buffer, increase the device buffer size accordingly.
5730 while ( dsPointerLeadTime * 2U > dsBufferSize )
5733 // Set cooperative level to DSSCL_EXCLUSIVE ... sound stops when window focus changes.
5734 // result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE );
5735 // Set cooperative level to DSSCL_PRIORITY ... sound remains when window focus changes.
5736 result = output->SetCooperativeLevel( hWnd, DSSCL_PRIORITY );
5737 if ( FAILED( result ) ) {
5739 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting cooperative level (" << dsDevices[ device ].name << ")!";
5740 errorText_ = errorStream_.str();
5744 // Even though we will write to the secondary buffer, we need to
5745 // access the primary buffer to set the correct output format
5746 // (since the default is 8-bit, 22 kHz!). Setup the DS primary
5747 // buffer description.
5748 DSBUFFERDESC bufferDescription;
5749 ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
5750 bufferDescription.dwSize = sizeof( DSBUFFERDESC );
5751 bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER;
5753 // Obtain the primary buffer
5754 LPDIRECTSOUNDBUFFER buffer;
5755 result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
5756 if ( FAILED( result ) ) {
5758 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") accessing primary buffer (" << dsDevices[ device ].name << ")!";
5759 errorText_ = errorStream_.str();
5763 // Set the primary DS buffer sound format.
5764 result = buffer->SetFormat( &waveFormat );
5765 if ( FAILED( result ) ) {
5767 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting primary buffer format (" << dsDevices[ device ].name << ")!";
5768 errorText_ = errorStream_.str();
5772 // Setup the secondary DS buffer description.
5773 ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
5774 bufferDescription.dwSize = sizeof( DSBUFFERDESC );
5775 bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
5776 DSBCAPS_GLOBALFOCUS |
5777 DSBCAPS_GETCURRENTPOSITION2 |
5778 DSBCAPS_LOCHARDWARE ); // Force hardware mixing
5779 bufferDescription.dwBufferBytes = dsBufferSize;
5780 bufferDescription.lpwfxFormat = &waveFormat;
5782 // Try to create the secondary DS buffer. If that doesn't work,
5783 // try to use software mixing. Otherwise, there's a problem.
5784 result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
5785 if ( FAILED( result ) ) {
5786 bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
5787 DSBCAPS_GLOBALFOCUS |
5788 DSBCAPS_GETCURRENTPOSITION2 |
5789 DSBCAPS_LOCSOFTWARE ); // Force software mixing
5790 result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
5791 if ( FAILED( result ) ) {
5793 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating secondary buffer (" << dsDevices[ device ].name << ")!";
5794 errorText_ = errorStream_.str();
5799 // Get the buffer size ... might be different from what we specified.
5801 dsbcaps.dwSize = sizeof( DSBCAPS );
5802 result = buffer->GetCaps( &dsbcaps );
5803 if ( FAILED( result ) ) {
5806 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";
5807 errorText_ = errorStream_.str();
5811 dsBufferSize = dsbcaps.dwBufferBytes;
5813 // Lock the DS buffer
5816 result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );
5817 if ( FAILED( result ) ) {
5820 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking buffer (" << dsDevices[ device ].name << ")!";
5821 errorText_ = errorStream_.str();
5825 // Zero the DS buffer
5826 ZeroMemory( audioPtr, dataLen );
5828 // Unlock the DS buffer
5829 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
5830 if ( FAILED( result ) ) {
5833 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking buffer (" << dsDevices[ device ].name << ")!";
5834 errorText_ = errorStream_.str();
5838 ohandle = (void *) output;
5839 bhandle = (void *) buffer;
5842 if ( mode == INPUT ) {
5844 LPDIRECTSOUNDCAPTURE input;
5845 result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );
5846 if ( FAILED( result ) ) {
5847 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";
5848 errorText_ = errorStream_.str();
5853 inCaps.dwSize = sizeof( inCaps );
5854 result = input->GetCaps( &inCaps );
5855 if ( FAILED( result ) ) {
5857 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsDevices[ device ].name << ")!";
5858 errorText_ = errorStream_.str();
5862 // Check channel information.
5863 if ( inCaps.dwChannels < channels + firstChannel ) {
5864 errorText_ = "RtApiDs::getDeviceInfo: the input device does not support requested input channels.";
5868 // Check format information. Use 16-bit format unless user
5870 DWORD deviceFormats;
5871 if ( channels + firstChannel == 2 ) {
5872 deviceFormats = WAVE_FORMAT_1S08 | WAVE_FORMAT_2S08 | WAVE_FORMAT_4S08 | WAVE_FORMAT_96S08;
5873 if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
5874 waveFormat.wBitsPerSample = 8;
5875 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
5877 else { // assume 16-bit is supported
5878 waveFormat.wBitsPerSample = 16;
5879 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
5882 else { // channel == 1
5883 deviceFormats = WAVE_FORMAT_1M08 | WAVE_FORMAT_2M08 | WAVE_FORMAT_4M08 | WAVE_FORMAT_96M08;
5884 if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
5885 waveFormat.wBitsPerSample = 8;
5886 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
5888 else { // assume 16-bit is supported
5889 waveFormat.wBitsPerSample = 16;
5890 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
5893 stream_.userFormat = format;
5895 // Update wave format structure and buffer information.
5896 waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
5897 waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
5898 dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
5900 // If the user wants an even bigger buffer, increase the device buffer size accordingly.
5901 while ( dsPointerLeadTime * 2U > dsBufferSize )
5904 // Setup the secondary DS buffer description.
5905 DSCBUFFERDESC bufferDescription;
5906 ZeroMemory( &bufferDescription, sizeof( DSCBUFFERDESC ) );
5907 bufferDescription.dwSize = sizeof( DSCBUFFERDESC );
5908 bufferDescription.dwFlags = 0;
5909 bufferDescription.dwReserved = 0;
5910 bufferDescription.dwBufferBytes = dsBufferSize;
5911 bufferDescription.lpwfxFormat = &waveFormat;
5913 // Create the capture buffer.
5914 LPDIRECTSOUNDCAPTUREBUFFER buffer;
5915 result = input->CreateCaptureBuffer( &bufferDescription, &buffer, NULL );
5916 if ( FAILED( result ) ) {
5918 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating input buffer (" << dsDevices[ device ].name << ")!";
5919 errorText_ = errorStream_.str();
5923 // Get the buffer size ... might be different from what we specified.
5925 dscbcaps.dwSize = sizeof( DSCBCAPS );
5926 result = buffer->GetCaps( &dscbcaps );
5927 if ( FAILED( result ) ) {
5930 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";
5931 errorText_ = errorStream_.str();
5935 dsBufferSize = dscbcaps.dwBufferBytes;
5937 // NOTE: We could have a problem here if this is a duplex stream
5938 // and the play and capture hardware buffer sizes are different
5939 // (I'm actually not sure if that is a problem or not).
5940 // Currently, we are not verifying that.
5942 // Lock the capture buffer
5945 result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );
5946 if ( FAILED( result ) ) {
5949 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking input buffer (" << dsDevices[ device ].name << ")!";
5950 errorText_ = errorStream_.str();
5955 ZeroMemory( audioPtr, dataLen );
5957 // Unlock the buffer
5958 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
5959 if ( FAILED( result ) ) {
5962 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking input buffer (" << dsDevices[ device ].name << ")!";
5963 errorText_ = errorStream_.str();
5967 ohandle = (void *) input;
5968 bhandle = (void *) buffer;
5971 // Set various stream parameters
5972 DsHandle *handle = 0;
5973 stream_.nDeviceChannels[mode] = channels + firstChannel;
5974 stream_.nUserChannels[mode] = channels;
5975 stream_.bufferSize = *bufferSize;
5976 stream_.channelOffset[mode] = firstChannel;
5977 stream_.deviceInterleaved[mode] = true;
5978 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
5979 else stream_.userInterleaved = true;
5981 // Set flag for buffer conversion
5982 stream_.doConvertBuffer[mode] = false;
5983 if (stream_.nUserChannels[mode] != stream_.nDeviceChannels[mode])
5984 stream_.doConvertBuffer[mode] = true;
5985 if (stream_.userFormat != stream_.deviceFormat[mode])
5986 stream_.doConvertBuffer[mode] = true;
5987 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
5988 stream_.nUserChannels[mode] > 1 )
5989 stream_.doConvertBuffer[mode] = true;
5991 // Allocate necessary internal buffers
5992 long bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
5993 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
5994 if ( stream_.userBuffer[mode] == NULL ) {
5995 errorText_ = "RtApiDs::probeDeviceOpen: error allocating user buffer memory.";
5999 if ( stream_.doConvertBuffer[mode] ) {
6001 bool makeBuffer = true;
6002 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
6003 if ( mode == INPUT ) {
6004 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
6005 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
6006 if ( bufferBytes <= (long) bytesOut ) makeBuffer = false;
6011 bufferBytes *= *bufferSize;
6012 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
6013 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
6014 if ( stream_.deviceBuffer == NULL ) {
6015 errorText_ = "RtApiDs::probeDeviceOpen: error allocating device buffer memory.";
6021 // Allocate our DsHandle structures for the stream.
6022 if ( stream_.apiHandle == 0 ) {
6024 handle = new DsHandle;
6026 catch ( std::bad_alloc& ) {
6027 errorText_ = "RtApiDs::probeDeviceOpen: error allocating AsioHandle memory.";
6031 // Create a manual-reset event.
6032 handle->condition = CreateEvent( NULL, // no security
6033 TRUE, // manual-reset
6034 FALSE, // non-signaled initially
6036 stream_.apiHandle = (void *) handle;
6039 handle = (DsHandle *) stream_.apiHandle;
6040 handle->id[mode] = ohandle;
6041 handle->buffer[mode] = bhandle;
6042 handle->dsBufferSize[mode] = dsBufferSize;
6043 handle->dsPointerLeadTime[mode] = dsPointerLeadTime;
6045 stream_.device[mode] = device;
6046 stream_.state = STREAM_STOPPED;
6047 if ( stream_.mode == OUTPUT && mode == INPUT )
6048 // We had already set up an output stream.
6049 stream_.mode = DUPLEX;
6051 stream_.mode = mode;
6052 stream_.nBuffers = nBuffers;
6053 stream_.sampleRate = sampleRate;
6055 // Setup the buffer conversion information structure.
6056 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
6058 // Setup the callback thread.
6059 if ( stream_.callbackInfo.isRunning == false ) {
6061 stream_.callbackInfo.isRunning = true;
6062 stream_.callbackInfo.object = (void *) this;
6063 stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &callbackHandler,
6064 &stream_.callbackInfo, 0, &threadId );
6065 if ( stream_.callbackInfo.thread == 0 ) {
6066 errorText_ = "RtApiDs::probeDeviceOpen: error creating callback thread!";
6070 // Boost DS thread priority
6071 SetThreadPriority( (HANDLE) stream_.callbackInfo.thread, THREAD_PRIORITY_HIGHEST );
6077 if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
6078 LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
6079 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6080 if ( buffer ) buffer->Release();
6083 if ( handle->buffer[1] ) {
6084 LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
6085 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6086 if ( buffer ) buffer->Release();
6089 CloseHandle( handle->condition );
6091 stream_.apiHandle = 0;
6094 for ( int i=0; i<2; i++ ) {
6095 if ( stream_.userBuffer[i] ) {
6096 free( stream_.userBuffer[i] );
6097 stream_.userBuffer[i] = 0;
6101 if ( stream_.deviceBuffer ) {
6102 free( stream_.deviceBuffer );
6103 stream_.deviceBuffer = 0;
6106 stream_.state = STREAM_CLOSED;
6110 void RtApiDs :: closeStream()
6112 if ( stream_.state == STREAM_CLOSED ) {
6113 errorText_ = "RtApiDs::closeStream(): no open stream to close!";
6114 error( RtAudioError::WARNING );
6118 // Stop the callback thread.
6119 stream_.callbackInfo.isRunning = false;
6120 WaitForSingleObject( (HANDLE) stream_.callbackInfo.thread, INFINITE );
6121 CloseHandle( (HANDLE) stream_.callbackInfo.thread );
6123 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6125 if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
6126 LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
6127 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6134 if ( handle->buffer[1] ) {
6135 LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
6136 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6143 CloseHandle( handle->condition );
6145 stream_.apiHandle = 0;
6148 for ( int i=0; i<2; i++ ) {
6149 if ( stream_.userBuffer[i] ) {
6150 free( stream_.userBuffer[i] );
6151 stream_.userBuffer[i] = 0;
6155 if ( stream_.deviceBuffer ) {
6156 free( stream_.deviceBuffer );
6157 stream_.deviceBuffer = 0;
6160 stream_.mode = UNINITIALIZED;
6161 stream_.state = STREAM_CLOSED;
6164 void RtApiDs :: startStream()
6167 if ( stream_.state == STREAM_RUNNING ) {
6168 errorText_ = "RtApiDs::startStream(): the stream is already running!";
6169 error( RtAudioError::WARNING );
6173 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6175 // Increase scheduler frequency on lesser windows (a side-effect of
6176 // increasing timer accuracy). On greater windows (Win2K or later),
6177 // this is already in effect.
6178 timeBeginPeriod( 1 );
6180 buffersRolling = false;
6181 duplexPrerollBytes = 0;
6183 if ( stream_.mode == DUPLEX ) {
6184 // 0.5 seconds of silence in DUPLEX mode while the devices spin up and synchronize.
6185 duplexPrerollBytes = (int) ( 0.5 * stream_.sampleRate * formatBytes( stream_.deviceFormat[1] ) * stream_.nDeviceChannels[1] );
6189 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
6191 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6192 result = buffer->Play( 0, 0, DSBPLAY_LOOPING );
6193 if ( FAILED( result ) ) {
6194 errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting output buffer!";
6195 errorText_ = errorStream_.str();
6200 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
6202 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6203 result = buffer->Start( DSCBSTART_LOOPING );
6204 if ( FAILED( result ) ) {
6205 errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting input buffer!";
6206 errorText_ = errorStream_.str();
6211 handle->drainCounter = 0;
6212 handle->internalDrain = false;
6213 ResetEvent( handle->condition );
6214 stream_.state = STREAM_RUNNING;
6217 if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );
6220 void RtApiDs :: stopStream()
6223 if ( stream_.state == STREAM_STOPPED ) {
6224 errorText_ = "RtApiDs::stopStream(): the stream is already stopped!";
6225 error( RtAudioError::WARNING );
6232 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6233 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
6234 if ( handle->drainCounter == 0 ) {
6235 handle->drainCounter = 2;
6236 WaitForSingleObject( handle->condition, INFINITE ); // block until signaled
6239 stream_.state = STREAM_STOPPED;
6241 MUTEX_LOCK( &stream_.mutex );
6243 // Stop the buffer and clear memory
6244 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6245 result = buffer->Stop();
6246 if ( FAILED( result ) ) {
6247 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping output buffer!";
6248 errorText_ = errorStream_.str();
6252 // Lock the buffer and clear it so that if we start to play again,
6253 // we won't have old data playing.
6254 result = buffer->Lock( 0, handle->dsBufferSize[0], &audioPtr, &dataLen, NULL, NULL, 0 );
6255 if ( FAILED( result ) ) {
6256 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking output buffer!";
6257 errorText_ = errorStream_.str();
6261 // Zero the DS buffer
6262 ZeroMemory( audioPtr, dataLen );
6264 // Unlock the DS buffer
6265 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
6266 if ( FAILED( result ) ) {
6267 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking output buffer!";
6268 errorText_ = errorStream_.str();
6272 // If we start playing again, we must begin at beginning of buffer.
6273 handle->bufferPointer[0] = 0;
6276 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
6277 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6281 stream_.state = STREAM_STOPPED;
6283 if ( stream_.mode != DUPLEX )
6284 MUTEX_LOCK( &stream_.mutex );
6286 result = buffer->Stop();
6287 if ( FAILED( result ) ) {
6288 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping input buffer!";
6289 errorText_ = errorStream_.str();
6293 // Lock the buffer and clear it so that if we start to play again,
6294 // we won't have old data playing.
6295 result = buffer->Lock( 0, handle->dsBufferSize[1], &audioPtr, &dataLen, NULL, NULL, 0 );
6296 if ( FAILED( result ) ) {
6297 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking input buffer!";
6298 errorText_ = errorStream_.str();
6302 // Zero the DS buffer
6303 ZeroMemory( audioPtr, dataLen );
6305 // Unlock the DS buffer
6306 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
6307 if ( FAILED( result ) ) {
6308 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking input buffer!";
6309 errorText_ = errorStream_.str();
6313 // If we start recording again, we must begin at beginning of buffer.
6314 handle->bufferPointer[1] = 0;
6318 timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows.
6319 MUTEX_UNLOCK( &stream_.mutex );
6321 if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );
6324 void RtApiDs :: abortStream()
6327 if ( stream_.state == STREAM_STOPPED ) {
6328 errorText_ = "RtApiDs::abortStream(): the stream is already stopped!";
6329 error( RtAudioError::WARNING );
6333 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6334 handle->drainCounter = 2;
6339 void RtApiDs :: callbackEvent()
6341 if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) {
6342 Sleep( 50 ); // sleep 50 milliseconds
6346 if ( stream_.state == STREAM_CLOSED ) {
6347 errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!";
6348 error( RtAudioError::WARNING );
6352 CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
6353 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6355 // Check if we were draining the stream and signal is finished.
6356 if ( handle->drainCounter > stream_.nBuffers + 2 ) {
6358 stream_.state = STREAM_STOPPING;
6359 if ( handle->internalDrain == false )
6360 SetEvent( handle->condition );
6366 // Invoke user callback to get fresh output data UNLESS we are
6368 if ( handle->drainCounter == 0 ) {
6369 RtAudioCallback callback = (RtAudioCallback) info->callback;
6370 double streamTime = getStreamTime();
6371 RtAudioStreamStatus status = 0;
6372 if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
6373 status |= RTAUDIO_OUTPUT_UNDERFLOW;
6374 handle->xrun[0] = false;
6376 if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
6377 status |= RTAUDIO_INPUT_OVERFLOW;
6378 handle->xrun[1] = false;
6380 int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
6381 stream_.bufferSize, streamTime, status, info->userData );
6382 if ( cbReturnValue == 2 ) {
6383 stream_.state = STREAM_STOPPING;
6384 handle->drainCounter = 2;
6388 else if ( cbReturnValue == 1 ) {
6389 handle->drainCounter = 1;
6390 handle->internalDrain = true;
6395 DWORD currentWritePointer, safeWritePointer;
6396 DWORD currentReadPointer, safeReadPointer;
6397 UINT nextWritePointer;
6399 LPVOID buffer1 = NULL;
6400 LPVOID buffer2 = NULL;
6401 DWORD bufferSize1 = 0;
6402 DWORD bufferSize2 = 0;
6407 MUTEX_LOCK( &stream_.mutex );
6408 if ( stream_.state == STREAM_STOPPED ) {
6409 MUTEX_UNLOCK( &stream_.mutex );
6413 if ( buffersRolling == false ) {
6414 if ( stream_.mode == DUPLEX ) {
6415 //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
6417 // It takes a while for the devices to get rolling. As a result,
6418 // there's no guarantee that the capture and write device pointers
6419 // will move in lockstep. Wait here for both devices to start
6420 // rolling, and then set our buffer pointers accordingly.
6421 // e.g. Crystal Drivers: the capture buffer starts up 5700 to 9600
6422 // bytes later than the write buffer.
6424 // Stub: a serious risk of having a pre-emptive scheduling round
6425 // take place between the two GetCurrentPosition calls... but I'm
6426 // really not sure how to solve the problem. Temporarily boost to
6427 // Realtime priority, maybe; but I'm not sure what priority the
6428 // DirectSound service threads run at. We *should* be roughly
6429 // within a ms or so of correct.
6431 LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6432 LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6434 DWORD startSafeWritePointer, startSafeReadPointer;
6436 result = dsWriteBuffer->GetCurrentPosition( NULL, &startSafeWritePointer );
6437 if ( FAILED( result ) ) {
6438 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
6439 errorText_ = errorStream_.str();
6440 MUTEX_UNLOCK( &stream_.mutex );
6441 error( RtAudioError::SYSTEM_ERROR );
6444 result = dsCaptureBuffer->GetCurrentPosition( NULL, &startSafeReadPointer );
6445 if ( FAILED( result ) ) {
6446 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
6447 errorText_ = errorStream_.str();
6448 MUTEX_UNLOCK( &stream_.mutex );
6449 error( RtAudioError::SYSTEM_ERROR );
6453 result = dsWriteBuffer->GetCurrentPosition( NULL, &safeWritePointer );
6454 if ( FAILED( result ) ) {
6455 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
6456 errorText_ = errorStream_.str();
6457 MUTEX_UNLOCK( &stream_.mutex );
6458 error( RtAudioError::SYSTEM_ERROR );
6461 result = dsCaptureBuffer->GetCurrentPosition( NULL, &safeReadPointer );
6462 if ( FAILED( result ) ) {
6463 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
6464 errorText_ = errorStream_.str();
6465 MUTEX_UNLOCK( &stream_.mutex );
6466 error( RtAudioError::SYSTEM_ERROR );
6469 if ( safeWritePointer != startSafeWritePointer && safeReadPointer != startSafeReadPointer ) break;
6473 //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
6475 handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];
6476 if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];
6477 handle->bufferPointer[1] = safeReadPointer;
6479 else if ( stream_.mode == OUTPUT ) {
6481 // Set the proper nextWritePosition after initial startup.
6482 LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6483 result = dsWriteBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer );
6484 if ( FAILED( result ) ) {
6485 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
6486 errorText_ = errorStream_.str();
6487 MUTEX_UNLOCK( &stream_.mutex );
6488 error( RtAudioError::SYSTEM_ERROR );
6491 handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];
6492 if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];
6495 buffersRolling = true;
6498 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
6500 LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6502 if ( handle->drainCounter > 1 ) { // write zeros to the output stream
6503 bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
6504 bufferBytes *= formatBytes( stream_.userFormat );
6505 memset( stream_.userBuffer[0], 0, bufferBytes );
6508 // Setup parameters and do buffer conversion if necessary.
6509 if ( stream_.doConvertBuffer[0] ) {
6510 buffer = stream_.deviceBuffer;
6511 convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
6512 bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[0];
6513 bufferBytes *= formatBytes( stream_.deviceFormat[0] );
6516 buffer = stream_.userBuffer[0];
6517 bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
6518 bufferBytes *= formatBytes( stream_.userFormat );
6521 // No byte swapping necessary in DirectSound implementation.
6523 // Ahhh ... windoze. 16-bit data is signed but 8-bit data is
6524 // unsigned. So, we need to convert our signed 8-bit data here to
6526 if ( stream_.deviceFormat[0] == RTAUDIO_SINT8 )
6527 for ( int i=0; i<bufferBytes; i++ ) buffer[i] = (unsigned char) ( buffer[i] + 128 );
6529 DWORD dsBufferSize = handle->dsBufferSize[0];
6530 nextWritePointer = handle->bufferPointer[0];
6532 DWORD endWrite, leadPointer;
6534 // Find out where the read and "safe write" pointers are.
6535 result = dsBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer );
6536 if ( FAILED( result ) ) {
6537 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
6538 errorText_ = errorStream_.str();
6539 MUTEX_UNLOCK( &stream_.mutex );
6540 error( RtAudioError::SYSTEM_ERROR );
6544 // We will copy our output buffer into the region between
6545 // safeWritePointer and leadPointer. If leadPointer is not
6546 // beyond the next endWrite position, wait until it is.
6547 leadPointer = safeWritePointer + handle->dsPointerLeadTime[0];
6548 //std::cout << "safeWritePointer = " << safeWritePointer << ", leadPointer = " << leadPointer << ", nextWritePointer = " << nextWritePointer << std::endl;
6549 if ( leadPointer > dsBufferSize ) leadPointer -= dsBufferSize;
6550 if ( leadPointer < nextWritePointer ) leadPointer += dsBufferSize; // unwrap offset
6551 endWrite = nextWritePointer + bufferBytes;
6553 // Check whether the entire write region is behind the play pointer.
6554 if ( leadPointer >= endWrite ) break;
6556 // If we are here, then we must wait until the leadPointer advances
6557 // beyond the end of our next write region. We use the
6558 // Sleep() function to suspend operation until that happens.
6559 double millis = ( endWrite - leadPointer ) * 1000.0;
6560 millis /= ( formatBytes( stream_.deviceFormat[0]) * stream_.nDeviceChannels[0] * stream_.sampleRate);
6561 if ( millis < 1.0 ) millis = 1.0;
6562 Sleep( (DWORD) millis );
6565 if ( dsPointerBetween( nextWritePointer, safeWritePointer, currentWritePointer, dsBufferSize )
6566 || dsPointerBetween( endWrite, safeWritePointer, currentWritePointer, dsBufferSize ) ) {
6567 // We've strayed into the forbidden zone ... resync the read pointer.
6568 handle->xrun[0] = true;
6569 nextWritePointer = safeWritePointer + handle->dsPointerLeadTime[0] - bufferBytes;
6570 if ( nextWritePointer >= dsBufferSize ) nextWritePointer -= dsBufferSize;
6571 handle->bufferPointer[0] = nextWritePointer;
6572 endWrite = nextWritePointer + bufferBytes;
6575 // Lock free space in the buffer
6576 result = dsBuffer->Lock( nextWritePointer, bufferBytes, &buffer1,
6577 &bufferSize1, &buffer2, &bufferSize2, 0 );
6578 if ( FAILED( result ) ) {
6579 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking buffer during playback!";
6580 errorText_ = errorStream_.str();
6581 MUTEX_UNLOCK( &stream_.mutex );
6582 error( RtAudioError::SYSTEM_ERROR );
6586 // Copy our buffer into the DS buffer
6587 CopyMemory( buffer1, buffer, bufferSize1 );
6588 if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 );
6590 // Update our buffer offset and unlock sound buffer
6591 dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
6592 if ( FAILED( result ) ) {
6593 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!";
6594 errorText_ = errorStream_.str();
6595 MUTEX_UNLOCK( &stream_.mutex );
6596 error( RtAudioError::SYSTEM_ERROR );
6599 nextWritePointer = ( nextWritePointer + bufferSize1 + bufferSize2 ) % dsBufferSize;
6600 handle->bufferPointer[0] = nextWritePointer;
6603 // Don't bother draining input
6604 if ( handle->drainCounter ) {
6605 handle->drainCounter++;
6609 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
6611 // Setup parameters.
6612 if ( stream_.doConvertBuffer[1] ) {
6613 buffer = stream_.deviceBuffer;
6614 bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[1];
6615 bufferBytes *= formatBytes( stream_.deviceFormat[1] );
6618 buffer = stream_.userBuffer[1];
6619 bufferBytes = stream_.bufferSize * stream_.nUserChannels[1];
6620 bufferBytes *= formatBytes( stream_.userFormat );
6623 LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6624 long nextReadPointer = handle->bufferPointer[1];
6625 DWORD dsBufferSize = handle->dsBufferSize[1];
6627 // Find out where the write and "safe read" pointers are.
6628 result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer );
6629 if ( FAILED( result ) ) {
6630 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
6631 errorText_ = errorStream_.str();
6632 MUTEX_UNLOCK( &stream_.mutex );
6633 error( RtAudioError::SYSTEM_ERROR );
6637 if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset
6638 DWORD endRead = nextReadPointer + bufferBytes;
6640 // Handling depends on whether we are INPUT or DUPLEX.
6641 // If we're in INPUT mode then waiting is a good thing. If we're in DUPLEX mode,
6642 // then a wait here will drag the write pointers into the forbidden zone.
6644 // In DUPLEX mode, rather than wait, we will back off the read pointer until
6645 // it's in a safe position. This causes dropouts, but it seems to be the only
6646 // practical way to sync up the read and write pointers reliably, given the
6647 // the very complex relationship between phase and increment of the read and write
6650 // In order to minimize audible dropouts in DUPLEX mode, we will
6651 // provide a pre-roll period of 0.5 seconds in which we return
6652 // zeros from the read buffer while the pointers sync up.
6654 if ( stream_.mode == DUPLEX ) {
6655 if ( safeReadPointer < endRead ) {
6656 if ( duplexPrerollBytes <= 0 ) {
6657 // Pre-roll time over. Be more agressive.
6658 int adjustment = endRead-safeReadPointer;
6660 handle->xrun[1] = true;
6662 // - large adjustments: we've probably run out of CPU cycles, so just resync exactly,
6663 // and perform fine adjustments later.
6664 // - small adjustments: back off by twice as much.
6665 if ( adjustment >= 2*bufferBytes )
6666 nextReadPointer = safeReadPointer-2*bufferBytes;
6668 nextReadPointer = safeReadPointer-bufferBytes-adjustment;
6670 if ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;
6674 // In pre=roll time. Just do it.
6675 nextReadPointer = safeReadPointer - bufferBytes;
6676 while ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;
6678 endRead = nextReadPointer + bufferBytes;
6681 else { // mode == INPUT
6682 while ( safeReadPointer < endRead && stream_.callbackInfo.isRunning ) {
6683 // See comments for playback.
6684 double millis = (endRead - safeReadPointer) * 1000.0;
6685 millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate);
6686 if ( millis < 1.0 ) millis = 1.0;
6687 Sleep( (DWORD) millis );
6689 // Wake up and find out where we are now.
6690 result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer );
6691 if ( FAILED( result ) ) {
6692 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
6693 errorText_ = errorStream_.str();
6694 MUTEX_UNLOCK( &stream_.mutex );
6695 error( RtAudioError::SYSTEM_ERROR );
6699 if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset
6703 // Lock free space in the buffer
6704 result = dsBuffer->Lock( nextReadPointer, bufferBytes, &buffer1,
6705 &bufferSize1, &buffer2, &bufferSize2, 0 );
6706 if ( FAILED( result ) ) {
6707 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking capture buffer!";
6708 errorText_ = errorStream_.str();
6709 MUTEX_UNLOCK( &stream_.mutex );
6710 error( RtAudioError::SYSTEM_ERROR );
6714 if ( duplexPrerollBytes <= 0 ) {
6715 // Copy our buffer into the DS buffer
6716 CopyMemory( buffer, buffer1, bufferSize1 );
6717 if ( buffer2 != NULL ) CopyMemory( buffer+bufferSize1, buffer2, bufferSize2 );
6720 memset( buffer, 0, bufferSize1 );
6721 if ( buffer2 != NULL ) memset( buffer + bufferSize1, 0, bufferSize2 );
6722 duplexPrerollBytes -= bufferSize1 + bufferSize2;
6725 // Update our buffer offset and unlock sound buffer
6726 nextReadPointer = ( nextReadPointer + bufferSize1 + bufferSize2 ) % dsBufferSize;
6727 dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
6728 if ( FAILED( result ) ) {
6729 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking capture buffer!";
6730 errorText_ = errorStream_.str();
6731 MUTEX_UNLOCK( &stream_.mutex );
6732 error( RtAudioError::SYSTEM_ERROR );
6735 handle->bufferPointer[1] = nextReadPointer;
6737 // No byte swapping necessary in DirectSound implementation.
6739 // If necessary, convert 8-bit data from unsigned to signed.
6740 if ( stream_.deviceFormat[1] == RTAUDIO_SINT8 )
6741 for ( int j=0; j<bufferBytes; j++ ) buffer[j] = (signed char) ( buffer[j] - 128 );
6743 // Do buffer conversion if necessary.
6744 if ( stream_.doConvertBuffer[1] )
6745 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
6749 MUTEX_UNLOCK( &stream_.mutex );
6750 RtApi::tickStreamTime();
6753 // Definitions for utility functions and callbacks
6754 // specific to the DirectSound implementation.
6756 static unsigned __stdcall callbackHandler( void *ptr )
6758 CallbackInfo *info = (CallbackInfo *) ptr;
6759 RtApiDs *object = (RtApiDs *) info->object;
6760 bool* isRunning = &info->isRunning;
6762 while ( *isRunning == true ) {
6763 object->callbackEvent();
6770 static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
6771 LPCTSTR description,
6775 struct DsProbeData& probeInfo = *(struct DsProbeData*) lpContext;
6776 std::vector<struct DsDevice>& dsDevices = *probeInfo.dsDevices;
6779 bool validDevice = false;
6780 if ( probeInfo.isInput == true ) {
6782 LPDIRECTSOUNDCAPTURE object;
6784 hr = DirectSoundCaptureCreate( lpguid, &object, NULL );
6785 if ( hr != DS_OK ) return TRUE;
6787 caps.dwSize = sizeof(caps);
6788 hr = object->GetCaps( &caps );
6789 if ( hr == DS_OK ) {
6790 if ( caps.dwChannels > 0 && caps.dwFormats > 0 )
6797 LPDIRECTSOUND object;
6798 hr = DirectSoundCreate( lpguid, &object, NULL );
6799 if ( hr != DS_OK ) return TRUE;
6801 caps.dwSize = sizeof(caps);
6802 hr = object->GetCaps( &caps );
6803 if ( hr == DS_OK ) {
6804 if ( caps.dwFlags & DSCAPS_PRIMARYMONO || caps.dwFlags & DSCAPS_PRIMARYSTEREO )
6810 // If good device, then save its name and guid.
6811 std::string name = convertCharPointerToStdString( description );
6812 //if ( name == "Primary Sound Driver" || name == "Primary Sound Capture Driver" )
6813 if ( lpguid == NULL )
6814 name = "Default Device";
6815 if ( validDevice ) {
6816 for ( unsigned int i=0; i<dsDevices.size(); i++ ) {
6817 if ( dsDevices[i].name == name ) {
6818 dsDevices[i].found = true;
6819 if ( probeInfo.isInput ) {
6820 dsDevices[i].id[1] = lpguid;
6821 dsDevices[i].validId[1] = true;
6824 dsDevices[i].id[0] = lpguid;
6825 dsDevices[i].validId[0] = true;
6833 device.found = true;
6834 if ( probeInfo.isInput ) {
6835 device.id[1] = lpguid;
6836 device.validId[1] = true;
6839 device.id[0] = lpguid;
6840 device.validId[0] = true;
6842 dsDevices.push_back( device );
6848 static const char* getErrorString( int code )
6852 case DSERR_ALLOCATED:
6853 return "Already allocated";
6855 case DSERR_CONTROLUNAVAIL:
6856 return "Control unavailable";
6858 case DSERR_INVALIDPARAM:
6859 return "Invalid parameter";
6861 case DSERR_INVALIDCALL:
6862 return "Invalid call";
6865 return "Generic error";
6867 case DSERR_PRIOLEVELNEEDED:
6868 return "Priority level needed";
6870 case DSERR_OUTOFMEMORY:
6871 return "Out of memory";
6873 case DSERR_BADFORMAT:
6874 return "The sample rate or the channel format is not supported";
6876 case DSERR_UNSUPPORTED:
6877 return "Not supported";
6879 case DSERR_NODRIVER:
6882 case DSERR_ALREADYINITIALIZED:
6883 return "Already initialized";
6885 case DSERR_NOAGGREGATION:
6886 return "No aggregation";
6888 case DSERR_BUFFERLOST:
6889 return "Buffer lost";
6891 case DSERR_OTHERAPPHASPRIO:
6892 return "Another application already has priority";
6894 case DSERR_UNINITIALIZED:
6895 return "Uninitialized";
6898 return "DirectSound unknown error";
6901 //******************** End of __WINDOWS_DS__ *********************//
6905 #if defined(__LINUX_ALSA__)
6907 #include <alsa/asoundlib.h>
6910 // A structure to hold various information related to the ALSA API
6913 snd_pcm_t *handles[2];
6916 pthread_cond_t runnable_cv;
6920 :synchronized(false), runnable(false) { xrun[0] = false; xrun[1] = false; }
6923 static void *alsaCallbackHandler( void * ptr );
6925 RtApiAlsa :: RtApiAlsa()
6927 // Nothing to do here.
6930 RtApiAlsa :: ~RtApiAlsa()
6932 if ( stream_.state != STREAM_CLOSED ) closeStream();
6935 unsigned int RtApiAlsa :: getDeviceCount( void )
6937 unsigned nDevices = 0;
6938 int result, subdevice, card;
6942 // Count cards and devices
6944 snd_card_next( &card );
6945 while ( card >= 0 ) {
6946 sprintf( name, "hw:%d", card );
6947 result = snd_ctl_open( &handle, name, 0 );
6949 errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << ".";
6950 errorText_ = errorStream_.str();
6951 error( RtAudioError::WARNING );
6956 result = snd_ctl_pcm_next_device( handle, &subdevice );
6958 errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
6959 errorText_ = errorStream_.str();
6960 error( RtAudioError::WARNING );
6963 if ( subdevice < 0 )
6968 snd_ctl_close( handle );
6969 snd_card_next( &card );
6972 result = snd_ctl_open( &handle, "default", 0 );
6975 snd_ctl_close( handle );
6981 RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device )
6983 RtAudio::DeviceInfo info;
6984 info.probed = false;
6986 unsigned nDevices = 0;
6987 int result, subdevice, card;
6991 // Count cards and devices
6994 snd_card_next( &card );
6995 while ( card >= 0 ) {
6996 sprintf( name, "hw:%d", card );
6997 result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
6999 errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << ".";
7000 errorText_ = errorStream_.str();
7001 error( RtAudioError::WARNING );
7006 result = snd_ctl_pcm_next_device( chandle, &subdevice );
7008 errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
7009 errorText_ = errorStream_.str();
7010 error( RtAudioError::WARNING );
7013 if ( subdevice < 0 ) break;
7014 if ( nDevices == device ) {
7015 sprintf( name, "hw:%d,%d", card, subdevice );
7021 snd_ctl_close( chandle );
7022 snd_card_next( &card );
7025 result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
7026 if ( result == 0 ) {
7027 if ( nDevices == device ) {
7028 strcpy( name, "default" );
7034 if ( nDevices == 0 ) {
7035 errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!";
7036 error( RtAudioError::INVALID_USE );
7040 if ( device >= nDevices ) {
7041 errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!";
7042 error( RtAudioError::INVALID_USE );
7048 // If a stream is already open, we cannot probe the stream devices.
7049 // Thus, use the saved results.
7050 if ( stream_.state != STREAM_CLOSED &&
7051 ( stream_.device[0] == device || stream_.device[1] == device ) ) {
7052 snd_ctl_close( chandle );
7053 if ( device >= devices_.size() ) {
7054 errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened.";
7055 error( RtAudioError::WARNING );
7058 return devices_[ device ];
7061 int openMode = SND_PCM_ASYNC;
7062 snd_pcm_stream_t stream;
7063 snd_pcm_info_t *pcminfo;
7064 snd_pcm_info_alloca( &pcminfo );
7066 snd_pcm_hw_params_t *params;
7067 snd_pcm_hw_params_alloca( ¶ms );
7069 // First try for playback unless default device (which has subdev -1)
7070 stream = SND_PCM_STREAM_PLAYBACK;
7071 snd_pcm_info_set_stream( pcminfo, stream );
7072 if ( subdevice != -1 ) {
7073 snd_pcm_info_set_device( pcminfo, subdevice );
7074 snd_pcm_info_set_subdevice( pcminfo, 0 );
7076 result = snd_ctl_pcm_info( chandle, pcminfo );
7078 // Device probably doesn't support playback.
7083 result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK );
7085 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
7086 errorText_ = errorStream_.str();
7087 error( RtAudioError::WARNING );
7091 // The device is open ... fill the parameter structure.
7092 result = snd_pcm_hw_params_any( phandle, params );
7094 snd_pcm_close( phandle );
7095 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
7096 errorText_ = errorStream_.str();
7097 error( RtAudioError::WARNING );
7101 // Get output channel information.
7103 result = snd_pcm_hw_params_get_channels_max( params, &value );
7105 snd_pcm_close( phandle );
7106 errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << ".";
7107 errorText_ = errorStream_.str();
7108 error( RtAudioError::WARNING );
7111 info.outputChannels = value;
7112 snd_pcm_close( phandle );
7115 stream = SND_PCM_STREAM_CAPTURE;
7116 snd_pcm_info_set_stream( pcminfo, stream );
7118 // Now try for capture unless default device (with subdev = -1)
7119 if ( subdevice != -1 ) {
7120 result = snd_ctl_pcm_info( chandle, pcminfo );
7121 snd_ctl_close( chandle );
7123 // Device probably doesn't support capture.
7124 if ( info.outputChannels == 0 ) return info;
7125 goto probeParameters;
7129 snd_ctl_close( chandle );
7131 result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
7133 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
7134 errorText_ = errorStream_.str();
7135 error( RtAudioError::WARNING );
7136 if ( info.outputChannels == 0 ) return info;
7137 goto probeParameters;
7140 // The device is open ... fill the parameter structure.
7141 result = snd_pcm_hw_params_any( phandle, params );
7143 snd_pcm_close( phandle );
7144 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
7145 errorText_ = errorStream_.str();
7146 error( RtAudioError::WARNING );
7147 if ( info.outputChannels == 0 ) return info;
7148 goto probeParameters;
7151 result = snd_pcm_hw_params_get_channels_max( params, &value );
7153 snd_pcm_close( phandle );
7154 errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << ".";
7155 errorText_ = errorStream_.str();
7156 error( RtAudioError::WARNING );
7157 if ( info.outputChannels == 0 ) return info;
7158 goto probeParameters;
7160 info.inputChannels = value;
7161 snd_pcm_close( phandle );
7163 // If device opens for both playback and capture, we determine the channels.
7164 if ( info.outputChannels > 0 && info.inputChannels > 0 )
7165 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
7167 // ALSA doesn't provide default devices so we'll use the first available one.
7168 if ( device == 0 && info.outputChannels > 0 )
7169 info.isDefaultOutput = true;
7170 if ( device == 0 && info.inputChannels > 0 )
7171 info.isDefaultInput = true;
7174 // At this point, we just need to figure out the supported data
7175 // formats and sample rates. We'll proceed by opening the device in
7176 // the direction with the maximum number of channels, or playback if
7177 // they are equal. This might limit our sample rate options, but so
7180 if ( info.outputChannels >= info.inputChannels )
7181 stream = SND_PCM_STREAM_PLAYBACK;
7183 stream = SND_PCM_STREAM_CAPTURE;
7184 snd_pcm_info_set_stream( pcminfo, stream );
7186 result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
7188 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
7189 errorText_ = errorStream_.str();
7190 error( RtAudioError::WARNING );
7194 // The device is open ... fill the parameter structure.
7195 result = snd_pcm_hw_params_any( phandle, params );
7197 snd_pcm_close( phandle );
7198 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
7199 errorText_ = errorStream_.str();
7200 error( RtAudioError::WARNING );
7204 // Test our discrete set of sample rate values.
7205 info.sampleRates.clear();
7206 for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {
7207 if ( snd_pcm_hw_params_test_rate( phandle, params, SAMPLE_RATES[i], 0 ) == 0 ) {
7208 info.sampleRates.push_back( SAMPLE_RATES[i] );
7210 if ( !info.preferredSampleRate || ( SAMPLE_RATES[i] <= 48000 && SAMPLE_RATES[i] > info.preferredSampleRate ) )
7211 info.preferredSampleRate = SAMPLE_RATES[i];
7214 if ( info.sampleRates.size() == 0 ) {
7215 snd_pcm_close( phandle );
7216 errorStream_ << "RtApiAlsa::getDeviceInfo: no supported sample rates found for device (" << name << ").";
7217 errorText_ = errorStream_.str();
7218 error( RtAudioError::WARNING );
7222 // Probe the supported data formats ... we don't care about endian-ness just yet
7223 snd_pcm_format_t format;
7224 info.nativeFormats = 0;
7225 format = SND_PCM_FORMAT_S8;
7226 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7227 info.nativeFormats |= RTAUDIO_SINT8;
7228 format = SND_PCM_FORMAT_S16;
7229 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7230 info.nativeFormats |= RTAUDIO_SINT16;
7231 format = SND_PCM_FORMAT_S24;
7232 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7233 info.nativeFormats |= RTAUDIO_SINT24;
7234 format = SND_PCM_FORMAT_S32;
7235 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7236 info.nativeFormats |= RTAUDIO_SINT32;
7237 format = SND_PCM_FORMAT_FLOAT;
7238 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7239 info.nativeFormats |= RTAUDIO_FLOAT32;
7240 format = SND_PCM_FORMAT_FLOAT64;
7241 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7242 info.nativeFormats |= RTAUDIO_FLOAT64;
7244 // Check that we have at least one supported format
7245 if ( info.nativeFormats == 0 ) {
7246 snd_pcm_close( phandle );
7247 errorStream_ << "RtApiAlsa::getDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio.";
7248 errorText_ = errorStream_.str();
7249 error( RtAudioError::WARNING );
7253 // Get the device name
7255 result = snd_card_get_name( card, &cardname );
7256 if ( result >= 0 ) {
7257 sprintf( name, "hw:%s,%d", cardname, subdevice );
7262 // That's all ... close the device and return
7263 snd_pcm_close( phandle );
7268 void RtApiAlsa :: saveDeviceInfo( void )
7272 unsigned int nDevices = getDeviceCount();
7273 devices_.resize( nDevices );
7274 for ( unsigned int i=0; i<nDevices; i++ )
7275 devices_[i] = getDeviceInfo( i );
7278 bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
7279 unsigned int firstChannel, unsigned int sampleRate,
7280 RtAudioFormat format, unsigned int *bufferSize,
7281 RtAudio::StreamOptions *options )
7284 #if defined(__RTAUDIO_DEBUG__)
7286 snd_output_stdio_attach(&out, stderr, 0);
7289 // I'm not using the "plug" interface ... too much inconsistent behavior.
7291 unsigned nDevices = 0;
7292 int result, subdevice, card;
7296 if ( options && options->flags & RTAUDIO_ALSA_USE_DEFAULT )
7297 snprintf(name, sizeof(name), "%s", "default");
7299 // Count cards and devices
7301 snd_card_next( &card );
7302 while ( card >= 0 ) {
7303 sprintf( name, "hw:%d", card );
7304 result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
7306 errorStream_ << "RtApiAlsa::probeDeviceOpen: control open, card = " << card << ", " << snd_strerror( result ) << ".";
7307 errorText_ = errorStream_.str();
7312 result = snd_ctl_pcm_next_device( chandle, &subdevice );
7313 if ( result < 0 ) break;
7314 if ( subdevice < 0 ) break;
7315 if ( nDevices == device ) {
7316 sprintf( name, "hw:%d,%d", card, subdevice );
7317 snd_ctl_close( chandle );
7322 snd_ctl_close( chandle );
7323 snd_card_next( &card );
7326 result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
7327 if ( result == 0 ) {
7328 if ( nDevices == device ) {
7329 strcpy( name, "default" );
7335 if ( nDevices == 0 ) {
7336 // This should not happen because a check is made before this function is called.
7337 errorText_ = "RtApiAlsa::probeDeviceOpen: no devices found!";
7341 if ( device >= nDevices ) {
7342 // This should not happen because a check is made before this function is called.
7343 errorText_ = "RtApiAlsa::probeDeviceOpen: device ID is invalid!";
7350 // The getDeviceInfo() function will not work for a device that is
7351 // already open. Thus, we'll probe the system before opening a
7352 // stream and save the results for use by getDeviceInfo().
7353 if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) // only do once
7354 this->saveDeviceInfo();
7356 snd_pcm_stream_t stream;
7357 if ( mode == OUTPUT )
7358 stream = SND_PCM_STREAM_PLAYBACK;
7360 stream = SND_PCM_STREAM_CAPTURE;
7363 int openMode = SND_PCM_ASYNC;
7364 result = snd_pcm_open( &phandle, name, stream, openMode );
7366 if ( mode == OUTPUT )
7367 errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for output.";
7369 errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for input.";
7370 errorText_ = errorStream_.str();
7374 // Fill the parameter structure.
7375 snd_pcm_hw_params_t *hw_params;
7376 snd_pcm_hw_params_alloca( &hw_params );
7377 result = snd_pcm_hw_params_any( phandle, hw_params );
7379 snd_pcm_close( phandle );
7380 errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") parameters, " << snd_strerror( result ) << ".";
7381 errorText_ = errorStream_.str();
7385 #if defined(__RTAUDIO_DEBUG__)
7386 fprintf( stderr, "\nRtApiAlsa: dump hardware params just after device open:\n\n" );
7387 snd_pcm_hw_params_dump( hw_params, out );
7390 // Set access ... check user preference.
7391 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) {
7392 stream_.userInterleaved = false;
7393 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
7395 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
7396 stream_.deviceInterleaved[mode] = true;
7399 stream_.deviceInterleaved[mode] = false;
7402 stream_.userInterleaved = true;
7403 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
7405 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
7406 stream_.deviceInterleaved[mode] = false;
7409 stream_.deviceInterleaved[mode] = true;
7413 snd_pcm_close( phandle );
7414 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") access, " << snd_strerror( result ) << ".";
7415 errorText_ = errorStream_.str();
7419 // Determine how to set the device format.
7420 stream_.userFormat = format;
7421 snd_pcm_format_t deviceFormat = SND_PCM_FORMAT_UNKNOWN;
7423 if ( format == RTAUDIO_SINT8 )
7424 deviceFormat = SND_PCM_FORMAT_S8;
7425 else if ( format == RTAUDIO_SINT16 )
7426 deviceFormat = SND_PCM_FORMAT_S16;
7427 else if ( format == RTAUDIO_SINT24 )
7428 deviceFormat = SND_PCM_FORMAT_S24;
7429 else if ( format == RTAUDIO_SINT32 )
7430 deviceFormat = SND_PCM_FORMAT_S32;
7431 else if ( format == RTAUDIO_FLOAT32 )
7432 deviceFormat = SND_PCM_FORMAT_FLOAT;
7433 else if ( format == RTAUDIO_FLOAT64 )
7434 deviceFormat = SND_PCM_FORMAT_FLOAT64;
7436 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat) == 0) {
7437 stream_.deviceFormat[mode] = format;
7441 // The user requested format is not natively supported by the device.
7442 deviceFormat = SND_PCM_FORMAT_FLOAT64;
7443 if ( snd_pcm_hw_params_test_format( phandle, hw_params, deviceFormat ) == 0 ) {
7444 stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
7448 deviceFormat = SND_PCM_FORMAT_FLOAT;
7449 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7450 stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
7454 deviceFormat = SND_PCM_FORMAT_S32;
7455 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7456 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
7460 deviceFormat = SND_PCM_FORMAT_S24;
7461 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7462 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
7466 deviceFormat = SND_PCM_FORMAT_S16;
7467 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7468 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
7472 deviceFormat = SND_PCM_FORMAT_S8;
7473 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7474 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
7478 // If we get here, no supported format was found.
7479 snd_pcm_close( phandle );
7480 errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device " << device << " data format not supported by RtAudio.";
7481 errorText_ = errorStream_.str();
7485 result = snd_pcm_hw_params_set_format( phandle, hw_params, deviceFormat );
7487 snd_pcm_close( phandle );
7488 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") data format, " << snd_strerror( result ) << ".";
7489 errorText_ = errorStream_.str();
7493 // Determine whether byte-swaping is necessary.
7494 stream_.doByteSwap[mode] = false;
7495 if ( deviceFormat != SND_PCM_FORMAT_S8 ) {
7496 result = snd_pcm_format_cpu_endian( deviceFormat );
7498 stream_.doByteSwap[mode] = true;
7499 else if (result < 0) {
7500 snd_pcm_close( phandle );
7501 errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") endian-ness, " << snd_strerror( result ) << ".";
7502 errorText_ = errorStream_.str();
7507 // Set the sample rate.
7508 result = snd_pcm_hw_params_set_rate_near( phandle, hw_params, (unsigned int*) &sampleRate, 0 );
7510 snd_pcm_close( phandle );
7511 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting sample rate on device (" << name << "), " << snd_strerror( result ) << ".";
7512 errorText_ = errorStream_.str();
7516 // Determine the number of channels for this device. We support a possible
7517 // minimum device channel number > than the value requested by the user.
7518 stream_.nUserChannels[mode] = channels;
7520 result = snd_pcm_hw_params_get_channels_max( hw_params, &value );
7521 unsigned int deviceChannels = value;
7522 if ( result < 0 || deviceChannels < channels + firstChannel ) {
7523 snd_pcm_close( phandle );
7524 errorStream_ << "RtApiAlsa::probeDeviceOpen: requested channel parameters not supported by device (" << name << "), " << snd_strerror( result ) << ".";
7525 errorText_ = errorStream_.str();
7529 result = snd_pcm_hw_params_get_channels_min( hw_params, &value );
7531 snd_pcm_close( phandle );
7532 errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting minimum channels for device (" << name << "), " << snd_strerror( result ) << ".";
7533 errorText_ = errorStream_.str();
7536 deviceChannels = value;
7537 if ( deviceChannels < channels + firstChannel ) deviceChannels = channels + firstChannel;
7538 stream_.nDeviceChannels[mode] = deviceChannels;
7540 // Set the device channels.
7541 result = snd_pcm_hw_params_set_channels( phandle, hw_params, deviceChannels );
7543 snd_pcm_close( phandle );
7544 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting channels for device (" << name << "), " << snd_strerror( result ) << ".";
7545 errorText_ = errorStream_.str();
7549 // Set the buffer (or period) size.
7551 snd_pcm_uframes_t periodSize = *bufferSize;
7552 result = snd_pcm_hw_params_set_period_size_near( phandle, hw_params, &periodSize, &dir );
7554 snd_pcm_close( phandle );
7555 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting period size for device (" << name << "), " << snd_strerror( result ) << ".";
7556 errorText_ = errorStream_.str();
7559 *bufferSize = periodSize;
7561 // Set the buffer number, which in ALSA is referred to as the "period".
7562 unsigned int periods = 0;
7563 if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2;
7564 if ( options && options->numberOfBuffers > 0 ) periods = options->numberOfBuffers;
7565 if ( periods < 2 ) periods = 4; // a fairly safe default value
7566 result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir );
7568 snd_pcm_close( phandle );
7569 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << ".";
7570 errorText_ = errorStream_.str();
7574 // If attempting to setup a duplex stream, the bufferSize parameter
7575 // MUST be the same in both directions!
7576 if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {
7577 snd_pcm_close( phandle );
7578 errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ").";
7579 errorText_ = errorStream_.str();
7583 stream_.bufferSize = *bufferSize;
7585 // Install the hardware configuration
7586 result = snd_pcm_hw_params( phandle, hw_params );
7588 snd_pcm_close( phandle );
7589 errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing hardware configuration on device (" << name << "), " << snd_strerror( result ) << ".";
7590 errorText_ = errorStream_.str();
7594 #if defined(__RTAUDIO_DEBUG__)
7595 fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n");
7596 snd_pcm_hw_params_dump( hw_params, out );
7599 // Set the software configuration to fill buffers with zeros and prevent device stopping on xruns.
7600 snd_pcm_sw_params_t *sw_params = NULL;
7601 snd_pcm_sw_params_alloca( &sw_params );
7602 snd_pcm_sw_params_current( phandle, sw_params );
7603 snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize );
7604 snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, ULONG_MAX );
7605 snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 );
7607 // The following two settings were suggested by Theo Veenker
7608 //snd_pcm_sw_params_set_avail_min( phandle, sw_params, *bufferSize );
7609 //snd_pcm_sw_params_set_xfer_align( phandle, sw_params, 1 );
7611 // here are two options for a fix
7612 //snd_pcm_sw_params_set_silence_size( phandle, sw_params, ULONG_MAX );
7613 snd_pcm_uframes_t val;
7614 snd_pcm_sw_params_get_boundary( sw_params, &val );
7615 snd_pcm_sw_params_set_silence_size( phandle, sw_params, val );
7617 result = snd_pcm_sw_params( phandle, sw_params );
7619 snd_pcm_close( phandle );
7620 errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing software configuration on device (" << name << "), " << snd_strerror( result ) << ".";
7621 errorText_ = errorStream_.str();
7625 #if defined(__RTAUDIO_DEBUG__)
7626 fprintf(stderr, "\nRtApiAlsa: dump software params after installation:\n\n");
7627 snd_pcm_sw_params_dump( sw_params, out );
7630 // Set flags for buffer conversion
7631 stream_.doConvertBuffer[mode] = false;
7632 if ( stream_.userFormat != stream_.deviceFormat[mode] )
7633 stream_.doConvertBuffer[mode] = true;
7634 if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
7635 stream_.doConvertBuffer[mode] = true;
7636 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
7637 stream_.nUserChannels[mode] > 1 )
7638 stream_.doConvertBuffer[mode] = true;
7640 // Allocate the ApiHandle if necessary and then save.
7641 AlsaHandle *apiInfo = 0;
7642 if ( stream_.apiHandle == 0 ) {
7644 apiInfo = (AlsaHandle *) new AlsaHandle;
7646 catch ( std::bad_alloc& ) {
7647 errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating AlsaHandle memory.";
7651 if ( pthread_cond_init( &apiInfo->runnable_cv, NULL ) ) {
7652 errorText_ = "RtApiAlsa::probeDeviceOpen: error initializing pthread condition variable.";
7656 stream_.apiHandle = (void *) apiInfo;
7657 apiInfo->handles[0] = 0;
7658 apiInfo->handles[1] = 0;
7661 apiInfo = (AlsaHandle *) stream_.apiHandle;
7663 apiInfo->handles[mode] = phandle;
7666 // Allocate necessary internal buffers.
7667 unsigned long bufferBytes;
7668 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
7669 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
7670 if ( stream_.userBuffer[mode] == NULL ) {
7671 errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating user buffer memory.";
7675 if ( stream_.doConvertBuffer[mode] ) {
7677 bool makeBuffer = true;
7678 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
7679 if ( mode == INPUT ) {
7680 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
7681 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
7682 if ( bufferBytes <= bytesOut ) makeBuffer = false;
7687 bufferBytes *= *bufferSize;
7688 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
7689 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
7690 if ( stream_.deviceBuffer == NULL ) {
7691 errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating device buffer memory.";
7697 stream_.sampleRate = sampleRate;
7698 stream_.nBuffers = periods;
7699 stream_.device[mode] = device;
7700 stream_.state = STREAM_STOPPED;
7702 // Setup the buffer conversion information structure.
7703 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
7705 // Setup thread if necessary.
7706 if ( stream_.mode == OUTPUT && mode == INPUT ) {
7707 // We had already set up an output stream.
7708 stream_.mode = DUPLEX;
7709 // Link the streams if possible.
7710 apiInfo->synchronized = false;
7711 if ( snd_pcm_link( apiInfo->handles[0], apiInfo->handles[1] ) == 0 )
7712 apiInfo->synchronized = true;
7714 errorText_ = "RtApiAlsa::probeDeviceOpen: unable to synchronize input and output devices.";
7715 error( RtAudioError::WARNING );
7719 stream_.mode = mode;
7721 // Setup callback thread.
7722 stream_.callbackInfo.object = (void *) this;
7724 // Set the thread attributes for joinable and realtime scheduling
7725 // priority (optional). The higher priority will only take affect
7726 // if the program is run as root or suid. Note, under Linux
7727 // processes with CAP_SYS_NICE privilege, a user can change
7728 // scheduling policy and priority (thus need not be root). See
7729 // POSIX "capabilities".
7730 pthread_attr_t attr;
7731 pthread_attr_init( &attr );
7732 pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
7734 #ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
7735 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
7736 // We previously attempted to increase the audio callback priority
7737 // to SCHED_RR here via the attributes. However, while no errors
7738 // were reported in doing so, it did not work. So, now this is
7739 // done in the alsaCallbackHandler function.
7740 stream_.callbackInfo.doRealtime = true;
7741 int priority = options->priority;
7742 int min = sched_get_priority_min( SCHED_RR );
7743 int max = sched_get_priority_max( SCHED_RR );
7744 if ( priority < min ) priority = min;
7745 else if ( priority > max ) priority = max;
7746 stream_.callbackInfo.priority = priority;
7750 stream_.callbackInfo.isRunning = true;
7751 result = pthread_create( &stream_.callbackInfo.thread, &attr, alsaCallbackHandler, &stream_.callbackInfo );
7752 pthread_attr_destroy( &attr );
7754 stream_.callbackInfo.isRunning = false;
7755 errorText_ = "RtApiAlsa::error creating callback thread!";
7764 pthread_cond_destroy( &apiInfo->runnable_cv );
7765 if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
7766 if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
7768 stream_.apiHandle = 0;
7771 if ( phandle) snd_pcm_close( phandle );
7773 for ( int i=0; i<2; i++ ) {
7774 if ( stream_.userBuffer[i] ) {
7775 free( stream_.userBuffer[i] );
7776 stream_.userBuffer[i] = 0;
7780 if ( stream_.deviceBuffer ) {
7781 free( stream_.deviceBuffer );
7782 stream_.deviceBuffer = 0;
7785 stream_.state = STREAM_CLOSED;
7789 void RtApiAlsa :: closeStream()
7791 if ( stream_.state == STREAM_CLOSED ) {
7792 errorText_ = "RtApiAlsa::closeStream(): no open stream to close!";
7793 error( RtAudioError::WARNING );
7797 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
7798 stream_.callbackInfo.isRunning = false;
7799 MUTEX_LOCK( &stream_.mutex );
7800 if ( stream_.state == STREAM_STOPPED ) {
7801 apiInfo->runnable = true;
7802 pthread_cond_signal( &apiInfo->runnable_cv );
7804 MUTEX_UNLOCK( &stream_.mutex );
7805 pthread_join( stream_.callbackInfo.thread, NULL );
7807 if ( stream_.state == STREAM_RUNNING ) {
7808 stream_.state = STREAM_STOPPED;
7809 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
7810 snd_pcm_drop( apiInfo->handles[0] );
7811 if ( stream_.mode == INPUT || stream_.mode == DUPLEX )
7812 snd_pcm_drop( apiInfo->handles[1] );
7816 pthread_cond_destroy( &apiInfo->runnable_cv );
7817 if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
7818 if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
7820 stream_.apiHandle = 0;
7823 for ( int i=0; i<2; i++ ) {
7824 if ( stream_.userBuffer[i] ) {
7825 free( stream_.userBuffer[i] );
7826 stream_.userBuffer[i] = 0;
7830 if ( stream_.deviceBuffer ) {
7831 free( stream_.deviceBuffer );
7832 stream_.deviceBuffer = 0;
7835 stream_.mode = UNINITIALIZED;
7836 stream_.state = STREAM_CLOSED;
7839 void RtApiAlsa :: startStream()
7841 // This method calls snd_pcm_prepare if the device isn't already in that state.
7844 if ( stream_.state == STREAM_RUNNING ) {
7845 errorText_ = "RtApiAlsa::startStream(): the stream is already running!";
7846 error( RtAudioError::WARNING );
7850 MUTEX_LOCK( &stream_.mutex );
7853 snd_pcm_state_t state;
7854 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
7855 snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
7856 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
7857 state = snd_pcm_state( handle[0] );
7858 if ( state != SND_PCM_STATE_PREPARED ) {
7859 result = snd_pcm_prepare( handle[0] );
7861 errorStream_ << "RtApiAlsa::startStream: error preparing output pcm device, " << snd_strerror( result ) << ".";
7862 errorText_ = errorStream_.str();
7868 if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
7869 result = snd_pcm_drop(handle[1]); // fix to remove stale data received since device has been open
7870 state = snd_pcm_state( handle[1] );
7871 if ( state != SND_PCM_STATE_PREPARED ) {
7872 result = snd_pcm_prepare( handle[1] );
7874 errorStream_ << "RtApiAlsa::startStream: error preparing input pcm device, " << snd_strerror( result ) << ".";
7875 errorText_ = errorStream_.str();
7881 stream_.state = STREAM_RUNNING;
7884 apiInfo->runnable = true;
7885 pthread_cond_signal( &apiInfo->runnable_cv );
7886 MUTEX_UNLOCK( &stream_.mutex );
7888 if ( result >= 0 ) return;
7889 error( RtAudioError::SYSTEM_ERROR );
7892 void RtApiAlsa :: stopStream()
7895 if ( stream_.state == STREAM_STOPPED ) {
7896 errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!";
7897 error( RtAudioError::WARNING );
7901 stream_.state = STREAM_STOPPED;
7902 MUTEX_LOCK( &stream_.mutex );
7905 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
7906 snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
7907 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
7908 if ( apiInfo->synchronized )
7909 result = snd_pcm_drop( handle[0] );
7911 result = snd_pcm_drain( handle[0] );
7913 errorStream_ << "RtApiAlsa::stopStream: error draining output pcm device, " << snd_strerror( result ) << ".";
7914 errorText_ = errorStream_.str();
7919 if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
7920 result = snd_pcm_drop( handle[1] );
7922 errorStream_ << "RtApiAlsa::stopStream: error stopping input pcm device, " << snd_strerror( result ) << ".";
7923 errorText_ = errorStream_.str();
7929 apiInfo->runnable = false; // fixes high CPU usage when stopped
7930 MUTEX_UNLOCK( &stream_.mutex );
7932 if ( result >= 0 ) return;
7933 error( RtAudioError::SYSTEM_ERROR );
7936 void RtApiAlsa :: abortStream()
7939 if ( stream_.state == STREAM_STOPPED ) {
7940 errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!";
7941 error( RtAudioError::WARNING );
7945 stream_.state = STREAM_STOPPED;
7946 MUTEX_LOCK( &stream_.mutex );
7949 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
7950 snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
7951 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
7952 result = snd_pcm_drop( handle[0] );
7954 errorStream_ << "RtApiAlsa::abortStream: error aborting output pcm device, " << snd_strerror( result ) << ".";
7955 errorText_ = errorStream_.str();
7960 if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
7961 result = snd_pcm_drop( handle[1] );
7963 errorStream_ << "RtApiAlsa::abortStream: error aborting input pcm device, " << snd_strerror( result ) << ".";
7964 errorText_ = errorStream_.str();
7970 apiInfo->runnable = false; // fixes high CPU usage when stopped
7971 MUTEX_UNLOCK( &stream_.mutex );
7973 if ( result >= 0 ) return;
7974 error( RtAudioError::SYSTEM_ERROR );
7977 void RtApiAlsa :: callbackEvent()
7979 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
7980 if ( stream_.state == STREAM_STOPPED ) {
7981 MUTEX_LOCK( &stream_.mutex );
7982 while ( !apiInfo->runnable )
7983 pthread_cond_wait( &apiInfo->runnable_cv, &stream_.mutex );
7985 if ( stream_.state != STREAM_RUNNING ) {
7986 MUTEX_UNLOCK( &stream_.mutex );
7989 MUTEX_UNLOCK( &stream_.mutex );
7992 if ( stream_.state == STREAM_CLOSED ) {
7993 errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!";
7994 error( RtAudioError::WARNING );
7998 int doStopStream = 0;
7999 RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
8000 double streamTime = getStreamTime();
8001 RtAudioStreamStatus status = 0;
8002 if ( stream_.mode != INPUT && apiInfo->xrun[0] == true ) {
8003 status |= RTAUDIO_OUTPUT_UNDERFLOW;
8004 apiInfo->xrun[0] = false;
8006 if ( stream_.mode != OUTPUT && apiInfo->xrun[1] == true ) {
8007 status |= RTAUDIO_INPUT_OVERFLOW;
8008 apiInfo->xrun[1] = false;
8010 doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
8011 stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
8013 if ( doStopStream == 2 ) {
8018 MUTEX_LOCK( &stream_.mutex );
8020 // The state might change while waiting on a mutex.
8021 if ( stream_.state == STREAM_STOPPED ) goto unlock;
8027 snd_pcm_sframes_t frames;
8028 RtAudioFormat format;
8029 handle = (snd_pcm_t **) apiInfo->handles;
8031 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
8033 // Setup parameters.
8034 if ( stream_.doConvertBuffer[1] ) {
8035 buffer = stream_.deviceBuffer;
8036 channels = stream_.nDeviceChannels[1];
8037 format = stream_.deviceFormat[1];
8040 buffer = stream_.userBuffer[1];
8041 channels = stream_.nUserChannels[1];
8042 format = stream_.userFormat;
8045 // Read samples from device in interleaved/non-interleaved format.
8046 if ( stream_.deviceInterleaved[1] )
8047 result = snd_pcm_readi( handle[1], buffer, stream_.bufferSize );
8049 void *bufs[channels];
8050 size_t offset = stream_.bufferSize * formatBytes( format );
8051 for ( int i=0; i<channels; i++ )
8052 bufs[i] = (void *) (buffer + (i * offset));
8053 result = snd_pcm_readn( handle[1], bufs, stream_.bufferSize );
8056 if ( result < (int) stream_.bufferSize ) {
8057 // Either an error or overrun occured.
8058 if ( result == -EPIPE ) {
8059 snd_pcm_state_t state = snd_pcm_state( handle[1] );
8060 if ( state == SND_PCM_STATE_XRUN ) {
8061 apiInfo->xrun[1] = true;
8062 result = snd_pcm_prepare( handle[1] );
8064 errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after overrun, " << snd_strerror( result ) << ".";
8065 errorText_ = errorStream_.str();
8069 errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
8070 errorText_ = errorStream_.str();
8074 errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << ".";
8075 errorText_ = errorStream_.str();
8077 error( RtAudioError::WARNING );
8081 // Do byte swapping if necessary.
8082 if ( stream_.doByteSwap[1] )
8083 byteSwapBuffer( buffer, stream_.bufferSize * channels, format );
8085 // Do buffer conversion if necessary.
8086 if ( stream_.doConvertBuffer[1] )
8087 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
8089 // Check stream latency
8090 result = snd_pcm_delay( handle[1], &frames );
8091 if ( result == 0 && frames > 0 ) stream_.latency[1] = frames;
8096 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
8098 // Setup parameters and do buffer conversion if necessary.
8099 if ( stream_.doConvertBuffer[0] ) {
8100 buffer = stream_.deviceBuffer;
8101 convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
8102 channels = stream_.nDeviceChannels[0];
8103 format = stream_.deviceFormat[0];
8106 buffer = stream_.userBuffer[0];
8107 channels = stream_.nUserChannels[0];
8108 format = stream_.userFormat;
8111 // Do byte swapping if necessary.
8112 if ( stream_.doByteSwap[0] )
8113 byteSwapBuffer(buffer, stream_.bufferSize * channels, format);
8115 // Write samples to device in interleaved/non-interleaved format.
8116 if ( stream_.deviceInterleaved[0] )
8117 result = snd_pcm_writei( handle[0], buffer, stream_.bufferSize );
8119 void *bufs[channels];
8120 size_t offset = stream_.bufferSize * formatBytes( format );
8121 for ( int i=0; i<channels; i++ )
8122 bufs[i] = (void *) (buffer + (i * offset));
8123 result = snd_pcm_writen( handle[0], bufs, stream_.bufferSize );
8126 if ( result < (int) stream_.bufferSize ) {
8127 // Either an error or underrun occured.
8128 if ( result == -EPIPE ) {
8129 snd_pcm_state_t state = snd_pcm_state( handle[0] );
8130 if ( state == SND_PCM_STATE_XRUN ) {
8131 apiInfo->xrun[0] = true;
8132 result = snd_pcm_prepare( handle[0] );
8134 errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after underrun, " << snd_strerror( result ) << ".";
8135 errorText_ = errorStream_.str();
8138 errorText_ = "RtApiAlsa::callbackEvent: audio write error, underrun.";
8141 errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
8142 errorText_ = errorStream_.str();
8146 errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << ".";
8147 errorText_ = errorStream_.str();
8149 error( RtAudioError::WARNING );
8153 // Check stream latency
8154 result = snd_pcm_delay( handle[0], &frames );
8155 if ( result == 0 && frames > 0 ) stream_.latency[0] = frames;
8159 MUTEX_UNLOCK( &stream_.mutex );
8161 RtApi::tickStreamTime();
8162 if ( doStopStream == 1 ) this->stopStream();
8165 static void *alsaCallbackHandler( void *ptr )
8167 CallbackInfo *info = (CallbackInfo *) ptr;
8168 RtApiAlsa *object = (RtApiAlsa *) info->object;
8169 bool *isRunning = &info->isRunning;
8171 #ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
8172 if ( info->doRealtime ) {
8173 pthread_t tID = pthread_self(); // ID of this thread
8174 sched_param prio = { info->priority }; // scheduling priority of thread
8175 pthread_setschedparam( tID, SCHED_RR, &prio );
8179 while ( *isRunning == true ) {
8180 pthread_testcancel();
8181 object->callbackEvent();
8184 pthread_exit( NULL );
8187 //******************** End of __LINUX_ALSA__ *********************//
8190 #if defined(__LINUX_PULSE__)
8192 // Code written by Peter Meerwald, pmeerw@pmeerw.net
8193 // and Tristan Matthews.
8195 #include <pulse/error.h>
8196 #include <pulse/simple.h>
8199 static const unsigned int SUPPORTED_SAMPLERATES[] = { 8000, 16000, 22050, 32000,
8200 44100, 48000, 96000, 0};
8202 struct rtaudio_pa_format_mapping_t {
8203 RtAudioFormat rtaudio_format;
8204 pa_sample_format_t pa_format;
8207 static const rtaudio_pa_format_mapping_t supported_sampleformats[] = {
8208 {RTAUDIO_SINT16, PA_SAMPLE_S16LE},
8209 {RTAUDIO_SINT32, PA_SAMPLE_S32LE},
8210 {RTAUDIO_FLOAT32, PA_SAMPLE_FLOAT32LE},
8211 {0, PA_SAMPLE_INVALID}};
8213 struct PulseAudioHandle {
8217 pthread_cond_t runnable_cv;
8219 PulseAudioHandle() : s_play(0), s_rec(0), runnable(false) { }
8222 RtApiPulse::~RtApiPulse()
8224 if ( stream_.state != STREAM_CLOSED )
8228 unsigned int RtApiPulse::getDeviceCount( void )
8233 RtAudio::DeviceInfo RtApiPulse::getDeviceInfo( unsigned int /*device*/ )
8235 RtAudio::DeviceInfo info;
8237 info.name = "PulseAudio";
8238 info.outputChannels = 2;
8239 info.inputChannels = 2;
8240 info.duplexChannels = 2;
8241 info.isDefaultOutput = true;
8242 info.isDefaultInput = true;
8244 for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr )
8245 info.sampleRates.push_back( *sr );
8247 info.preferredSampleRate = 48000;
8248 info.nativeFormats = RTAUDIO_SINT16 | RTAUDIO_SINT32 | RTAUDIO_FLOAT32;
8253 static void *pulseaudio_callback( void * user )
8255 CallbackInfo *cbi = static_cast<CallbackInfo *>( user );
8256 RtApiPulse *context = static_cast<RtApiPulse *>( cbi->object );
8257 volatile bool *isRunning = &cbi->isRunning;
8259 while ( *isRunning ) {
8260 pthread_testcancel();
8261 context->callbackEvent();
8264 pthread_exit( NULL );
8267 void RtApiPulse::closeStream( void )
8269 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8271 stream_.callbackInfo.isRunning = false;
8273 MUTEX_LOCK( &stream_.mutex );
8274 if ( stream_.state == STREAM_STOPPED ) {
8275 pah->runnable = true;
8276 pthread_cond_signal( &pah->runnable_cv );
8278 MUTEX_UNLOCK( &stream_.mutex );
8280 pthread_join( pah->thread, 0 );
8281 if ( pah->s_play ) {
8282 pa_simple_flush( pah->s_play, NULL );
8283 pa_simple_free( pah->s_play );
8286 pa_simple_free( pah->s_rec );
8288 pthread_cond_destroy( &pah->runnable_cv );
8290 stream_.apiHandle = 0;
8293 if ( stream_.userBuffer[0] ) {
8294 free( stream_.userBuffer[0] );
8295 stream_.userBuffer[0] = 0;
8297 if ( stream_.userBuffer[1] ) {
8298 free( stream_.userBuffer[1] );
8299 stream_.userBuffer[1] = 0;
8302 stream_.state = STREAM_CLOSED;
8303 stream_.mode = UNINITIALIZED;
8306 void RtApiPulse::callbackEvent( void )
8308 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8310 if ( stream_.state == STREAM_STOPPED ) {
8311 MUTEX_LOCK( &stream_.mutex );
8312 while ( !pah->runnable )
8313 pthread_cond_wait( &pah->runnable_cv, &stream_.mutex );
8315 if ( stream_.state != STREAM_RUNNING ) {
8316 MUTEX_UNLOCK( &stream_.mutex );
8319 MUTEX_UNLOCK( &stream_.mutex );
8322 if ( stream_.state == STREAM_CLOSED ) {
8323 errorText_ = "RtApiPulse::callbackEvent(): the stream is closed ... "
8324 "this shouldn't happen!";
8325 error( RtAudioError::WARNING );
8329 RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
8330 double streamTime = getStreamTime();
8331 RtAudioStreamStatus status = 0;
8332 int doStopStream = callback( stream_.userBuffer[OUTPUT], stream_.userBuffer[INPUT],
8333 stream_.bufferSize, streamTime, status,
8334 stream_.callbackInfo.userData );
8336 if ( doStopStream == 2 ) {
8341 MUTEX_LOCK( &stream_.mutex );
8342 void *pulse_in = stream_.doConvertBuffer[INPUT] ? stream_.deviceBuffer : stream_.userBuffer[INPUT];
8343 void *pulse_out = stream_.doConvertBuffer[OUTPUT] ? stream_.deviceBuffer : stream_.userBuffer[OUTPUT];
8345 if ( stream_.state != STREAM_RUNNING )
8350 if (stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
8351 if ( stream_.doConvertBuffer[OUTPUT] ) {
8352 convertBuffer( stream_.deviceBuffer,
8353 stream_.userBuffer[OUTPUT],
8354 stream_.convertInfo[OUTPUT] );
8355 bytes = stream_.nDeviceChannels[OUTPUT] * stream_.bufferSize *
8356 formatBytes( stream_.deviceFormat[OUTPUT] );
8358 bytes = stream_.nUserChannels[OUTPUT] * stream_.bufferSize *
8359 formatBytes( stream_.userFormat );
8361 if ( pa_simple_write( pah->s_play, pulse_out, bytes, &pa_error ) < 0 ) {
8362 errorStream_ << "RtApiPulse::callbackEvent: audio write error, " <<
8363 pa_strerror( pa_error ) << ".";
8364 errorText_ = errorStream_.str();
8365 error( RtAudioError::WARNING );
8369 if ( stream_.mode == INPUT || stream_.mode == DUPLEX) {
8370 if ( stream_.doConvertBuffer[INPUT] )
8371 bytes = stream_.nDeviceChannels[INPUT] * stream_.bufferSize *
8372 formatBytes( stream_.deviceFormat[INPUT] );
8374 bytes = stream_.nUserChannels[INPUT] * stream_.bufferSize *
8375 formatBytes( stream_.userFormat );
8377 if ( pa_simple_read( pah->s_rec, pulse_in, bytes, &pa_error ) < 0 ) {
8378 errorStream_ << "RtApiPulse::callbackEvent: audio read error, " <<
8379 pa_strerror( pa_error ) << ".";
8380 errorText_ = errorStream_.str();
8381 error( RtAudioError::WARNING );
8383 if ( stream_.doConvertBuffer[INPUT] ) {
8384 convertBuffer( stream_.userBuffer[INPUT],
8385 stream_.deviceBuffer,
8386 stream_.convertInfo[INPUT] );
8391 MUTEX_UNLOCK( &stream_.mutex );
8392 RtApi::tickStreamTime();
8394 if ( doStopStream == 1 )
8398 void RtApiPulse::startStream( void )
8400 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8402 if ( stream_.state == STREAM_CLOSED ) {
8403 errorText_ = "RtApiPulse::startStream(): the stream is not open!";
8404 error( RtAudioError::INVALID_USE );
8407 if ( stream_.state == STREAM_RUNNING ) {
8408 errorText_ = "RtApiPulse::startStream(): the stream is already running!";
8409 error( RtAudioError::WARNING );
8413 MUTEX_LOCK( &stream_.mutex );
8415 stream_.state = STREAM_RUNNING;
8417 pah->runnable = true;
8418 pthread_cond_signal( &pah->runnable_cv );
8419 MUTEX_UNLOCK( &stream_.mutex );
8422 void RtApiPulse::stopStream( void )
8424 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8426 if ( stream_.state == STREAM_CLOSED ) {
8427 errorText_ = "RtApiPulse::stopStream(): the stream is not open!";
8428 error( RtAudioError::INVALID_USE );
8431 if ( stream_.state == STREAM_STOPPED ) {
8432 errorText_ = "RtApiPulse::stopStream(): the stream is already stopped!";
8433 error( RtAudioError::WARNING );
8437 stream_.state = STREAM_STOPPED;
8438 MUTEX_LOCK( &stream_.mutex );
8440 if ( pah && pah->s_play ) {
8442 if ( pa_simple_drain( pah->s_play, &pa_error ) < 0 ) {
8443 errorStream_ << "RtApiPulse::stopStream: error draining output device, " <<
8444 pa_strerror( pa_error ) << ".";
8445 errorText_ = errorStream_.str();
8446 MUTEX_UNLOCK( &stream_.mutex );
8447 error( RtAudioError::SYSTEM_ERROR );
8452 stream_.state = STREAM_STOPPED;
8453 MUTEX_UNLOCK( &stream_.mutex );
8456 void RtApiPulse::abortStream( void )
8458 PulseAudioHandle *pah = static_cast<PulseAudioHandle*>( stream_.apiHandle );
8460 if ( stream_.state == STREAM_CLOSED ) {
8461 errorText_ = "RtApiPulse::abortStream(): the stream is not open!";
8462 error( RtAudioError::INVALID_USE );
8465 if ( stream_.state == STREAM_STOPPED ) {
8466 errorText_ = "RtApiPulse::abortStream(): the stream is already stopped!";
8467 error( RtAudioError::WARNING );
8471 stream_.state = STREAM_STOPPED;
8472 MUTEX_LOCK( &stream_.mutex );
8474 if ( pah && pah->s_play ) {
8476 if ( pa_simple_flush( pah->s_play, &pa_error ) < 0 ) {
8477 errorStream_ << "RtApiPulse::abortStream: error flushing output device, " <<
8478 pa_strerror( pa_error ) << ".";
8479 errorText_ = errorStream_.str();
8480 MUTEX_UNLOCK( &stream_.mutex );
8481 error( RtAudioError::SYSTEM_ERROR );
8486 stream_.state = STREAM_STOPPED;
8487 MUTEX_UNLOCK( &stream_.mutex );
8490 bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode,
8491 unsigned int channels, unsigned int firstChannel,
8492 unsigned int sampleRate, RtAudioFormat format,
8493 unsigned int *bufferSize, RtAudio::StreamOptions *options )
8495 PulseAudioHandle *pah = 0;
8496 unsigned long bufferBytes = 0;
8499 if ( device != 0 ) return false;
8500 if ( mode != INPUT && mode != OUTPUT ) return false;
8501 if ( channels != 1 && channels != 2 ) {
8502 errorText_ = "RtApiPulse::probeDeviceOpen: unsupported number of channels.";
8505 ss.channels = channels;
8507 if ( firstChannel != 0 ) return false;
8509 bool sr_found = false;
8510 for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr ) {
8511 if ( sampleRate == *sr ) {
8513 stream_.sampleRate = sampleRate;
8514 ss.rate = sampleRate;
8519 errorText_ = "RtApiPulse::probeDeviceOpen: unsupported sample rate.";
8524 for ( const rtaudio_pa_format_mapping_t *sf = supported_sampleformats;
8525 sf->rtaudio_format && sf->pa_format != PA_SAMPLE_INVALID; ++sf ) {
8526 if ( format == sf->rtaudio_format ) {
8528 stream_.userFormat = sf->rtaudio_format;
8529 stream_.deviceFormat[mode] = stream_.userFormat;
8530 ss.format = sf->pa_format;
8534 if ( !sf_found ) { // Use internal data format conversion.
8535 stream_.userFormat = format;
8536 stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
8537 ss.format = PA_SAMPLE_FLOAT32LE;
8540 // Set other stream parameters.
8541 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
8542 else stream_.userInterleaved = true;
8543 stream_.deviceInterleaved[mode] = true;
8544 stream_.nBuffers = 1;
8545 stream_.doByteSwap[mode] = false;
8546 stream_.nUserChannels[mode] = channels;
8547 stream_.nDeviceChannels[mode] = channels + firstChannel;
8548 stream_.channelOffset[mode] = 0;
8549 std::string streamName = "RtAudio";
8551 // Set flags for buffer conversion.
8552 stream_.doConvertBuffer[mode] = false;
8553 if ( stream_.userFormat != stream_.deviceFormat[mode] )
8554 stream_.doConvertBuffer[mode] = true;
8555 if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
8556 stream_.doConvertBuffer[mode] = true;
8558 // Allocate necessary internal buffers.
8559 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
8560 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
8561 if ( stream_.userBuffer[mode] == NULL ) {
8562 errorText_ = "RtApiPulse::probeDeviceOpen: error allocating user buffer memory.";
8565 stream_.bufferSize = *bufferSize;
8567 if ( stream_.doConvertBuffer[mode] ) {
8569 bool makeBuffer = true;
8570 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
8571 if ( mode == INPUT ) {
8572 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
8573 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
8574 if ( bufferBytes <= bytesOut ) makeBuffer = false;
8579 bufferBytes *= *bufferSize;
8580 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
8581 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
8582 if ( stream_.deviceBuffer == NULL ) {
8583 errorText_ = "RtApiPulse::probeDeviceOpen: error allocating device buffer memory.";
8589 stream_.device[mode] = device;
8591 // Setup the buffer conversion information structure.
8592 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
8594 if ( !stream_.apiHandle ) {
8595 PulseAudioHandle *pah = new PulseAudioHandle;
8597 errorText_ = "RtApiPulse::probeDeviceOpen: error allocating memory for handle.";
8601 stream_.apiHandle = pah;
8602 if ( pthread_cond_init( &pah->runnable_cv, NULL ) != 0 ) {
8603 errorText_ = "RtApiPulse::probeDeviceOpen: error creating condition variable.";
8607 pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8610 if ( options && !options->streamName.empty() ) streamName = options->streamName;
8613 pa_buffer_attr buffer_attr;
8614 buffer_attr.fragsize = bufferBytes;
8615 buffer_attr.maxlength = -1;
8617 pah->s_rec = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_RECORD, NULL, "Record", &ss, NULL, &buffer_attr, &error );
8618 if ( !pah->s_rec ) {
8619 errorText_ = "RtApiPulse::probeDeviceOpen: error connecting input to PulseAudio server.";
8624 pah->s_play = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_PLAYBACK, NULL, "Playback", &ss, NULL, NULL, &error );
8625 if ( !pah->s_play ) {
8626 errorText_ = "RtApiPulse::probeDeviceOpen: error connecting output to PulseAudio server.";
8634 if ( stream_.mode == UNINITIALIZED )
8635 stream_.mode = mode;
8636 else if ( stream_.mode == mode )
8639 stream_.mode = DUPLEX;
8641 if ( !stream_.callbackInfo.isRunning ) {
8642 stream_.callbackInfo.object = this;
8643 stream_.callbackInfo.isRunning = true;
8644 if ( pthread_create( &pah->thread, NULL, pulseaudio_callback, (void *)&stream_.callbackInfo) != 0 ) {
8645 errorText_ = "RtApiPulse::probeDeviceOpen: error creating thread.";
8650 stream_.state = STREAM_STOPPED;
8654 if ( pah && stream_.callbackInfo.isRunning ) {
8655 pthread_cond_destroy( &pah->runnable_cv );
8657 stream_.apiHandle = 0;
8660 for ( int i=0; i<2; i++ ) {
8661 if ( stream_.userBuffer[i] ) {
8662 free( stream_.userBuffer[i] );
8663 stream_.userBuffer[i] = 0;
8667 if ( stream_.deviceBuffer ) {
8668 free( stream_.deviceBuffer );
8669 stream_.deviceBuffer = 0;
8675 //******************** End of __LINUX_PULSE__ *********************//
8678 #if defined(__LINUX_OSS__)
8681 #include <sys/ioctl.h>
8684 #include <sys/soundcard.h>
8688 static void *ossCallbackHandler(void * ptr);
8690 // A structure to hold various information related to the OSS API
8693 int id[2]; // device ids
8696 pthread_cond_t runnable;
8699 :triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
8702 RtApiOss :: RtApiOss()
8704 // Nothing to do here.
8707 RtApiOss :: ~RtApiOss()
8709 if ( stream_.state != STREAM_CLOSED ) closeStream();
8712 unsigned int RtApiOss :: getDeviceCount( void )
8714 int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
8715 if ( mixerfd == -1 ) {
8716 errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'.";
8717 error( RtAudioError::WARNING );
8721 oss_sysinfo sysinfo;
8722 if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) {
8724 errorText_ = "RtApiOss::getDeviceCount: error getting sysinfo, OSS version >= 4.0 is required.";
8725 error( RtAudioError::WARNING );
8730 return sysinfo.numaudios;
8733 RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device )
8735 RtAudio::DeviceInfo info;
8736 info.probed = false;
8738 int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
8739 if ( mixerfd == -1 ) {
8740 errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'.";
8741 error( RtAudioError::WARNING );
8745 oss_sysinfo sysinfo;
8746 int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
8747 if ( result == -1 ) {
8749 errorText_ = "RtApiOss::getDeviceInfo: error getting sysinfo, OSS version >= 4.0 is required.";
8750 error( RtAudioError::WARNING );
8754 unsigned nDevices = sysinfo.numaudios;
8755 if ( nDevices == 0 ) {
8757 errorText_ = "RtApiOss::getDeviceInfo: no devices found!";
8758 error( RtAudioError::INVALID_USE );
8762 if ( device >= nDevices ) {
8764 errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!";
8765 error( RtAudioError::INVALID_USE );
8769 oss_audioinfo ainfo;
8771 result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
8773 if ( result == -1 ) {
8774 errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
8775 errorText_ = errorStream_.str();
8776 error( RtAudioError::WARNING );
8781 if ( ainfo.caps & PCM_CAP_OUTPUT ) info.outputChannels = ainfo.max_channels;
8782 if ( ainfo.caps & PCM_CAP_INPUT ) info.inputChannels = ainfo.max_channels;
8783 if ( ainfo.caps & PCM_CAP_DUPLEX ) {
8784 if ( info.outputChannels > 0 && info.inputChannels > 0 && ainfo.caps & PCM_CAP_DUPLEX )
8785 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
8788 // Probe data formats ... do for input
8789 unsigned long mask = ainfo.iformats;
8790 if ( mask & AFMT_S16_LE || mask & AFMT_S16_BE )
8791 info.nativeFormats |= RTAUDIO_SINT16;
8792 if ( mask & AFMT_S8 )
8793 info.nativeFormats |= RTAUDIO_SINT8;
8794 if ( mask & AFMT_S32_LE || mask & AFMT_S32_BE )
8795 info.nativeFormats |= RTAUDIO_SINT32;
8797 if ( mask & AFMT_FLOAT )
8798 info.nativeFormats |= RTAUDIO_FLOAT32;
8800 if ( mask & AFMT_S24_LE || mask & AFMT_S24_BE )
8801 info.nativeFormats |= RTAUDIO_SINT24;
8803 // Check that we have at least one supported format
8804 if ( info.nativeFormats == 0 ) {
8805 errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio.";
8806 errorText_ = errorStream_.str();
8807 error( RtAudioError::WARNING );
8811 // Probe the supported sample rates.
8812 info.sampleRates.clear();
8813 if ( ainfo.nrates ) {
8814 for ( unsigned int i=0; i<ainfo.nrates; i++ ) {
8815 for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
8816 if ( ainfo.rates[i] == SAMPLE_RATES[k] ) {
8817 info.sampleRates.push_back( SAMPLE_RATES[k] );
8819 if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
8820 info.preferredSampleRate = SAMPLE_RATES[k];
8828 // Check min and max rate values;
8829 for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
8830 if ( ainfo.min_rate <= (int) SAMPLE_RATES[k] && ainfo.max_rate >= (int) SAMPLE_RATES[k] ) {
8831 info.sampleRates.push_back( SAMPLE_RATES[k] );
8833 if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
8834 info.preferredSampleRate = SAMPLE_RATES[k];
8839 if ( info.sampleRates.size() == 0 ) {
8840 errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ").";
8841 errorText_ = errorStream_.str();
8842 error( RtAudioError::WARNING );
8846 info.name = ainfo.name;
8853 bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
8854 unsigned int firstChannel, unsigned int sampleRate,
8855 RtAudioFormat format, unsigned int *bufferSize,
8856 RtAudio::StreamOptions *options )
8858 int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
8859 if ( mixerfd == -1 ) {
8860 errorText_ = "RtApiOss::probeDeviceOpen: error opening '/dev/mixer'.";
8864 oss_sysinfo sysinfo;
8865 int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
8866 if ( result == -1 ) {
8868 errorText_ = "RtApiOss::probeDeviceOpen: error getting sysinfo, OSS version >= 4.0 is required.";
8872 unsigned nDevices = sysinfo.numaudios;
8873 if ( nDevices == 0 ) {
8874 // This should not happen because a check is made before this function is called.
8876 errorText_ = "RtApiOss::probeDeviceOpen: no devices found!";
8880 if ( device >= nDevices ) {
8881 // This should not happen because a check is made before this function is called.
8883 errorText_ = "RtApiOss::probeDeviceOpen: device ID is invalid!";
8887 oss_audioinfo ainfo;
8889 result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
8891 if ( result == -1 ) {
8892 errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
8893 errorText_ = errorStream_.str();
8897 // Check if device supports input or output
8898 if ( ( mode == OUTPUT && !( ainfo.caps & PCM_CAP_OUTPUT ) ) ||
8899 ( mode == INPUT && !( ainfo.caps & PCM_CAP_INPUT ) ) ) {
8900 if ( mode == OUTPUT )
8901 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support output.";
8903 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support input.";
8904 errorText_ = errorStream_.str();
8909 OssHandle *handle = (OssHandle *) stream_.apiHandle;
8910 if ( mode == OUTPUT )
8912 else { // mode == INPUT
8913 if (stream_.mode == OUTPUT && stream_.device[0] == device) {
8914 // We just set the same device for playback ... close and reopen for duplex (OSS only).
8915 close( handle->id[0] );
8917 if ( !( ainfo.caps & PCM_CAP_DUPLEX ) ) {
8918 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support duplex mode.";
8919 errorText_ = errorStream_.str();
8922 // Check that the number previously set channels is the same.
8923 if ( stream_.nUserChannels[0] != channels ) {
8924 errorStream_ << "RtApiOss::probeDeviceOpen: input/output channels must be equal for OSS duplex device (" << ainfo.name << ").";
8925 errorText_ = errorStream_.str();
8934 // Set exclusive access if specified.
8935 if ( options && options->flags & RTAUDIO_HOG_DEVICE ) flags |= O_EXCL;
8937 // Try to open the device.
8939 fd = open( ainfo.devnode, flags, 0 );
8941 if ( errno == EBUSY )
8942 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") is busy.";
8944 errorStream_ << "RtApiOss::probeDeviceOpen: error opening device (" << ainfo.name << ").";
8945 errorText_ = errorStream_.str();
8949 // For duplex operation, specifically set this mode (this doesn't seem to work).
8951 if ( flags | O_RDWR ) {
8952 result = ioctl( fd, SNDCTL_DSP_SETDUPLEX, NULL );
8953 if ( result == -1) {
8954 errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ").";
8955 errorText_ = errorStream_.str();
8961 // Check the device channel support.
8962 stream_.nUserChannels[mode] = channels;
8963 if ( ainfo.max_channels < (int)(channels + firstChannel) ) {
8965 errorStream_ << "RtApiOss::probeDeviceOpen: the device (" << ainfo.name << ") does not support requested channel parameters.";
8966 errorText_ = errorStream_.str();
8970 // Set the number of channels.
8971 int deviceChannels = channels + firstChannel;
8972 result = ioctl( fd, SNDCTL_DSP_CHANNELS, &deviceChannels );
8973 if ( result == -1 || deviceChannels < (int)(channels + firstChannel) ) {
8975 errorStream_ << "RtApiOss::probeDeviceOpen: error setting channel parameters on device (" << ainfo.name << ").";
8976 errorText_ = errorStream_.str();
8979 stream_.nDeviceChannels[mode] = deviceChannels;
8981 // Get the data format mask
8983 result = ioctl( fd, SNDCTL_DSP_GETFMTS, &mask );
8984 if ( result == -1 ) {
8986 errorStream_ << "RtApiOss::probeDeviceOpen: error getting device (" << ainfo.name << ") data formats.";
8987 errorText_ = errorStream_.str();
8991 // Determine how to set the device format.
8992 stream_.userFormat = format;
8993 int deviceFormat = -1;
8994 stream_.doByteSwap[mode] = false;
8995 if ( format == RTAUDIO_SINT8 ) {
8996 if ( mask & AFMT_S8 ) {
8997 deviceFormat = AFMT_S8;
8998 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
9001 else if ( format == RTAUDIO_SINT16 ) {
9002 if ( mask & AFMT_S16_NE ) {
9003 deviceFormat = AFMT_S16_NE;
9004 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
9006 else if ( mask & AFMT_S16_OE ) {
9007 deviceFormat = AFMT_S16_OE;
9008 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
9009 stream_.doByteSwap[mode] = true;
9012 else if ( format == RTAUDIO_SINT24 ) {
9013 if ( mask & AFMT_S24_NE ) {
9014 deviceFormat = AFMT_S24_NE;
9015 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
9017 else if ( mask & AFMT_S24_OE ) {
9018 deviceFormat = AFMT_S24_OE;
9019 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
9020 stream_.doByteSwap[mode] = true;
9023 else if ( format == RTAUDIO_SINT32 ) {
9024 if ( mask & AFMT_S32_NE ) {
9025 deviceFormat = AFMT_S32_NE;
9026 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
9028 else if ( mask & AFMT_S32_OE ) {
9029 deviceFormat = AFMT_S32_OE;
9030 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
9031 stream_.doByteSwap[mode] = true;
9035 if ( deviceFormat == -1 ) {
9036 // The user requested format is not natively supported by the device.
9037 if ( mask & AFMT_S16_NE ) {
9038 deviceFormat = AFMT_S16_NE;
9039 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
9041 else if ( mask & AFMT_S32_NE ) {
9042 deviceFormat = AFMT_S32_NE;
9043 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
9045 else if ( mask & AFMT_S24_NE ) {
9046 deviceFormat = AFMT_S24_NE;
9047 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
9049 else if ( mask & AFMT_S16_OE ) {
9050 deviceFormat = AFMT_S16_OE;
9051 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
9052 stream_.doByteSwap[mode] = true;
9054 else if ( mask & AFMT_S32_OE ) {
9055 deviceFormat = AFMT_S32_OE;
9056 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
9057 stream_.doByteSwap[mode] = true;
9059 else if ( mask & AFMT_S24_OE ) {
9060 deviceFormat = AFMT_S24_OE;
9061 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
9062 stream_.doByteSwap[mode] = true;
9064 else if ( mask & AFMT_S8) {
9065 deviceFormat = AFMT_S8;
9066 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
9070 if ( stream_.deviceFormat[mode] == 0 ) {
9071 // This really shouldn't happen ...
9073 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") data format not supported by RtAudio.";
9074 errorText_ = errorStream_.str();
9078 // Set the data format.
9079 int temp = deviceFormat;
9080 result = ioctl( fd, SNDCTL_DSP_SETFMT, &deviceFormat );
9081 if ( result == -1 || deviceFormat != temp ) {
9083 errorStream_ << "RtApiOss::probeDeviceOpen: error setting data format on device (" << ainfo.name << ").";
9084 errorText_ = errorStream_.str();
9088 // Attempt to set the buffer size. According to OSS, the minimum
9089 // number of buffers is two. The supposed minimum buffer size is 16
9090 // bytes, so that will be our lower bound. The argument to this
9091 // call is in the form 0xMMMMSSSS (hex), where the buffer size (in
9092 // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM.
9093 // We'll check the actual value used near the end of the setup
9095 int ossBufferBytes = *bufferSize * formatBytes( stream_.deviceFormat[mode] ) * deviceChannels;
9096 if ( ossBufferBytes < 16 ) ossBufferBytes = 16;
9098 if ( options ) buffers = options->numberOfBuffers;
9099 if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) buffers = 2;
9100 if ( buffers < 2 ) buffers = 3;
9101 temp = ((int) buffers << 16) + (int)( log10( (double)ossBufferBytes ) / log10( 2.0 ) );
9102 result = ioctl( fd, SNDCTL_DSP_SETFRAGMENT, &temp );
9103 if ( result == -1 ) {
9105 errorStream_ << "RtApiOss::probeDeviceOpen: error setting buffer size on device (" << ainfo.name << ").";
9106 errorText_ = errorStream_.str();
9109 stream_.nBuffers = buffers;
9111 // Save buffer size (in sample frames).
9112 *bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels );
9113 stream_.bufferSize = *bufferSize;
9115 // Set the sample rate.
9116 int srate = sampleRate;
9117 result = ioctl( fd, SNDCTL_DSP_SPEED, &srate );
9118 if ( result == -1 ) {
9120 errorStream_ << "RtApiOss::probeDeviceOpen: error setting sample rate (" << sampleRate << ") on device (" << ainfo.name << ").";
9121 errorText_ = errorStream_.str();
9125 // Verify the sample rate setup worked.
9126 if ( abs( srate - (int)sampleRate ) > 100 ) {
9128 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ").";
9129 errorText_ = errorStream_.str();
9132 stream_.sampleRate = sampleRate;
9134 if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device) {
9135 // We're doing duplex setup here.
9136 stream_.deviceFormat[0] = stream_.deviceFormat[1];
9137 stream_.nDeviceChannels[0] = deviceChannels;
9140 // Set interleaving parameters.
9141 stream_.userInterleaved = true;
9142 stream_.deviceInterleaved[mode] = true;
9143 if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
9144 stream_.userInterleaved = false;
9146 // Set flags for buffer conversion
9147 stream_.doConvertBuffer[mode] = false;
9148 if ( stream_.userFormat != stream_.deviceFormat[mode] )
9149 stream_.doConvertBuffer[mode] = true;
9150 if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
9151 stream_.doConvertBuffer[mode] = true;
9152 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
9153 stream_.nUserChannels[mode] > 1 )
9154 stream_.doConvertBuffer[mode] = true;
9156 // Allocate the stream handles if necessary and then save.
9157 if ( stream_.apiHandle == 0 ) {
9159 handle = new OssHandle;
9161 catch ( std::bad_alloc& ) {
9162 errorText_ = "RtApiOss::probeDeviceOpen: error allocating OssHandle memory.";
9166 if ( pthread_cond_init( &handle->runnable, NULL ) ) {
9167 errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable.";
9171 stream_.apiHandle = (void *) handle;
9174 handle = (OssHandle *) stream_.apiHandle;
9176 handle->id[mode] = fd;
9178 // Allocate necessary internal buffers.
9179 unsigned long bufferBytes;
9180 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
9181 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
9182 if ( stream_.userBuffer[mode] == NULL ) {
9183 errorText_ = "RtApiOss::probeDeviceOpen: error allocating user buffer memory.";
9187 if ( stream_.doConvertBuffer[mode] ) {
9189 bool makeBuffer = true;
9190 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
9191 if ( mode == INPUT ) {
9192 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
9193 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
9194 if ( bufferBytes <= bytesOut ) makeBuffer = false;
9199 bufferBytes *= *bufferSize;
9200 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
9201 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
9202 if ( stream_.deviceBuffer == NULL ) {
9203 errorText_ = "RtApiOss::probeDeviceOpen: error allocating device buffer memory.";
9209 stream_.device[mode] = device;
9210 stream_.state = STREAM_STOPPED;
9212 // Setup the buffer conversion information structure.
9213 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
9215 // Setup thread if necessary.
9216 if ( stream_.mode == OUTPUT && mode == INPUT ) {
9217 // We had already set up an output stream.
9218 stream_.mode = DUPLEX;
9219 if ( stream_.device[0] == device ) handle->id[0] = fd;
9222 stream_.mode = mode;
9224 // Setup callback thread.
9225 stream_.callbackInfo.object = (void *) this;
9227 // Set the thread attributes for joinable and realtime scheduling
9228 // priority. The higher priority will only take affect if the
9229 // program is run as root or suid.
9230 pthread_attr_t attr;
9231 pthread_attr_init( &attr );
9232 pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
9233 #ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
9234 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
9235 struct sched_param param;
9236 int priority = options->priority;
9237 int min = sched_get_priority_min( SCHED_RR );
9238 int max = sched_get_priority_max( SCHED_RR );
9239 if ( priority < min ) priority = min;
9240 else if ( priority > max ) priority = max;
9241 param.sched_priority = priority;
9242 pthread_attr_setschedparam( &attr, ¶m );
9243 pthread_attr_setschedpolicy( &attr, SCHED_RR );
9246 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
9248 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
9251 stream_.callbackInfo.isRunning = true;
9252 result = pthread_create( &stream_.callbackInfo.thread, &attr, ossCallbackHandler, &stream_.callbackInfo );
9253 pthread_attr_destroy( &attr );
9255 stream_.callbackInfo.isRunning = false;
9256 errorText_ = "RtApiOss::error creating callback thread!";
9265 pthread_cond_destroy( &handle->runnable );
9266 if ( handle->id[0] ) close( handle->id[0] );
9267 if ( handle->id[1] ) close( handle->id[1] );
9269 stream_.apiHandle = 0;
9272 for ( int i=0; i<2; i++ ) {
9273 if ( stream_.userBuffer[i] ) {
9274 free( stream_.userBuffer[i] );
9275 stream_.userBuffer[i] = 0;
9279 if ( stream_.deviceBuffer ) {
9280 free( stream_.deviceBuffer );
9281 stream_.deviceBuffer = 0;
9287 void RtApiOss :: closeStream()
9289 if ( stream_.state == STREAM_CLOSED ) {
9290 errorText_ = "RtApiOss::closeStream(): no open stream to close!";
9291 error( RtAudioError::WARNING );
9295 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9296 stream_.callbackInfo.isRunning = false;
9297 MUTEX_LOCK( &stream_.mutex );
9298 if ( stream_.state == STREAM_STOPPED )
9299 pthread_cond_signal( &handle->runnable );
9300 MUTEX_UNLOCK( &stream_.mutex );
9301 pthread_join( stream_.callbackInfo.thread, NULL );
9303 if ( stream_.state == STREAM_RUNNING ) {
9304 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
9305 ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
9307 ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
9308 stream_.state = STREAM_STOPPED;
9312 pthread_cond_destroy( &handle->runnable );
9313 if ( handle->id[0] ) close( handle->id[0] );
9314 if ( handle->id[1] ) close( handle->id[1] );
9316 stream_.apiHandle = 0;
9319 for ( int i=0; i<2; i++ ) {
9320 if ( stream_.userBuffer[i] ) {
9321 free( stream_.userBuffer[i] );
9322 stream_.userBuffer[i] = 0;
9326 if ( stream_.deviceBuffer ) {
9327 free( stream_.deviceBuffer );
9328 stream_.deviceBuffer = 0;
9331 stream_.mode = UNINITIALIZED;
9332 stream_.state = STREAM_CLOSED;
9335 void RtApiOss :: startStream()
9338 if ( stream_.state == STREAM_RUNNING ) {
9339 errorText_ = "RtApiOss::startStream(): the stream is already running!";
9340 error( RtAudioError::WARNING );
9344 MUTEX_LOCK( &stream_.mutex );
9346 stream_.state = STREAM_RUNNING;
9348 // No need to do anything else here ... OSS automatically starts
9349 // when fed samples.
9351 MUTEX_UNLOCK( &stream_.mutex );
9353 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9354 pthread_cond_signal( &handle->runnable );
9357 void RtApiOss :: stopStream()
9360 if ( stream_.state == STREAM_STOPPED ) {
9361 errorText_ = "RtApiOss::stopStream(): the stream is already stopped!";
9362 error( RtAudioError::WARNING );
9366 MUTEX_LOCK( &stream_.mutex );
9368 // The state might change while waiting on a mutex.
9369 if ( stream_.state == STREAM_STOPPED ) {
9370 MUTEX_UNLOCK( &stream_.mutex );
9375 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9376 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
9378 // Flush the output with zeros a few times.
9381 RtAudioFormat format;
9383 if ( stream_.doConvertBuffer[0] ) {
9384 buffer = stream_.deviceBuffer;
9385 samples = stream_.bufferSize * stream_.nDeviceChannels[0];
9386 format = stream_.deviceFormat[0];
9389 buffer = stream_.userBuffer[0];
9390 samples = stream_.bufferSize * stream_.nUserChannels[0];
9391 format = stream_.userFormat;
9394 memset( buffer, 0, samples * formatBytes(format) );
9395 for ( unsigned int i=0; i<stream_.nBuffers+1; i++ ) {
9396 result = write( handle->id[0], buffer, samples * formatBytes(format) );
9397 if ( result == -1 ) {
9398 errorText_ = "RtApiOss::stopStream: audio write error.";
9399 error( RtAudioError::WARNING );
9403 result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
9404 if ( result == -1 ) {
9405 errorStream_ << "RtApiOss::stopStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
9406 errorText_ = errorStream_.str();
9409 handle->triggered = false;
9412 if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
9413 result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
9414 if ( result == -1 ) {
9415 errorStream_ << "RtApiOss::stopStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
9416 errorText_ = errorStream_.str();
9422 stream_.state = STREAM_STOPPED;
9423 MUTEX_UNLOCK( &stream_.mutex );
9425 if ( result != -1 ) return;
9426 error( RtAudioError::SYSTEM_ERROR );
9429 void RtApiOss :: abortStream()
9432 if ( stream_.state == STREAM_STOPPED ) {
9433 errorText_ = "RtApiOss::abortStream(): the stream is already stopped!";
9434 error( RtAudioError::WARNING );
9438 MUTEX_LOCK( &stream_.mutex );
9440 // The state might change while waiting on a mutex.
9441 if ( stream_.state == STREAM_STOPPED ) {
9442 MUTEX_UNLOCK( &stream_.mutex );
9447 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9448 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
9449 result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
9450 if ( result == -1 ) {
9451 errorStream_ << "RtApiOss::abortStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
9452 errorText_ = errorStream_.str();
9455 handle->triggered = false;
9458 if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
9459 result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
9460 if ( result == -1 ) {
9461 errorStream_ << "RtApiOss::abortStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
9462 errorText_ = errorStream_.str();
9468 stream_.state = STREAM_STOPPED;
9469 MUTEX_UNLOCK( &stream_.mutex );
9471 if ( result != -1 ) return;
9472 error( RtAudioError::SYSTEM_ERROR );
9475 void RtApiOss :: callbackEvent()
9477 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9478 if ( stream_.state == STREAM_STOPPED ) {
9479 MUTEX_LOCK( &stream_.mutex );
9480 pthread_cond_wait( &handle->runnable, &stream_.mutex );
9481 if ( stream_.state != STREAM_RUNNING ) {
9482 MUTEX_UNLOCK( &stream_.mutex );
9485 MUTEX_UNLOCK( &stream_.mutex );
9488 if ( stream_.state == STREAM_CLOSED ) {
9489 errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!";
9490 error( RtAudioError::WARNING );
9494 // Invoke user callback to get fresh output data.
9495 int doStopStream = 0;
9496 RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
9497 double streamTime = getStreamTime();
9498 RtAudioStreamStatus status = 0;
9499 if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
9500 status |= RTAUDIO_OUTPUT_UNDERFLOW;
9501 handle->xrun[0] = false;
9503 if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
9504 status |= RTAUDIO_INPUT_OVERFLOW;
9505 handle->xrun[1] = false;
9507 doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
9508 stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
9509 if ( doStopStream == 2 ) {
9510 this->abortStream();
9514 MUTEX_LOCK( &stream_.mutex );
9516 // The state might change while waiting on a mutex.
9517 if ( stream_.state == STREAM_STOPPED ) goto unlock;
9522 RtAudioFormat format;
9524 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
9526 // Setup parameters and do buffer conversion if necessary.
9527 if ( stream_.doConvertBuffer[0] ) {
9528 buffer = stream_.deviceBuffer;
9529 convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
9530 samples = stream_.bufferSize * stream_.nDeviceChannels[0];
9531 format = stream_.deviceFormat[0];
9534 buffer = stream_.userBuffer[0];
9535 samples = stream_.bufferSize * stream_.nUserChannels[0];
9536 format = stream_.userFormat;
9539 // Do byte swapping if necessary.
9540 if ( stream_.doByteSwap[0] )
9541 byteSwapBuffer( buffer, samples, format );
9543 if ( stream_.mode == DUPLEX && handle->triggered == false ) {
9545 ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
9546 result = write( handle->id[0], buffer, samples * formatBytes(format) );
9547 trig = PCM_ENABLE_INPUT|PCM_ENABLE_OUTPUT;
9548 ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
9549 handle->triggered = true;
9552 // Write samples to device.
9553 result = write( handle->id[0], buffer, samples * formatBytes(format) );
9555 if ( result == -1 ) {
9556 // We'll assume this is an underrun, though there isn't a
9557 // specific means for determining that.
9558 handle->xrun[0] = true;
9559 errorText_ = "RtApiOss::callbackEvent: audio write error.";
9560 error( RtAudioError::WARNING );
9561 // Continue on to input section.
9565 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
9567 // Setup parameters.
9568 if ( stream_.doConvertBuffer[1] ) {
9569 buffer = stream_.deviceBuffer;
9570 samples = stream_.bufferSize * stream_.nDeviceChannels[1];
9571 format = stream_.deviceFormat[1];
9574 buffer = stream_.userBuffer[1];
9575 samples = stream_.bufferSize * stream_.nUserChannels[1];
9576 format = stream_.userFormat;
9579 // Read samples from device.
9580 result = read( handle->id[1], buffer, samples * formatBytes(format) );
9582 if ( result == -1 ) {
9583 // We'll assume this is an overrun, though there isn't a
9584 // specific means for determining that.
9585 handle->xrun[1] = true;
9586 errorText_ = "RtApiOss::callbackEvent: audio read error.";
9587 error( RtAudioError::WARNING );
9591 // Do byte swapping if necessary.
9592 if ( stream_.doByteSwap[1] )
9593 byteSwapBuffer( buffer, samples, format );
9595 // Do buffer conversion if necessary.
9596 if ( stream_.doConvertBuffer[1] )
9597 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
9601 MUTEX_UNLOCK( &stream_.mutex );
9603 RtApi::tickStreamTime();
9604 if ( doStopStream == 1 ) this->stopStream();
9607 static void *ossCallbackHandler( void *ptr )
9609 CallbackInfo *info = (CallbackInfo *) ptr;
9610 RtApiOss *object = (RtApiOss *) info->object;
9611 bool *isRunning = &info->isRunning;
9613 while ( *isRunning == true ) {
9614 pthread_testcancel();
9615 object->callbackEvent();
9618 pthread_exit( NULL );
9621 //******************** End of __LINUX_OSS__ *********************//
9625 // *************************************************** //
9627 // Protected common (OS-independent) RtAudio methods.
9629 // *************************************************** //
9631 // This method can be modified to control the behavior of error
9632 // message printing.
9633 void RtApi :: error( RtAudioError::Type type )
9635 errorStream_.str(""); // clear the ostringstream
9637 RtAudioErrorCallback errorCallback = (RtAudioErrorCallback) stream_.callbackInfo.errorCallback;
9638 if ( errorCallback ) {
9639 // abortStream() can generate new error messages. Ignore them. Just keep original one.
9641 if ( firstErrorOccurred_ )
9644 firstErrorOccurred_ = true;
9645 const std::string errorMessage = errorText_;
9647 if ( type != RtAudioError::WARNING && stream_.state != STREAM_STOPPED) {
9648 stream_.callbackInfo.isRunning = false; // exit from the thread
9652 errorCallback( type, errorMessage );
9653 firstErrorOccurred_ = false;
9657 if ( type == RtAudioError::WARNING && showWarnings_ == true )
9658 std::cerr << '\n' << errorText_ << "\n\n";
9659 else if ( type != RtAudioError::WARNING )
9660 throw( RtAudioError( errorText_, type ) );
9663 void RtApi :: verifyStream()
9665 if ( stream_.state == STREAM_CLOSED ) {
9666 errorText_ = "RtApi:: a stream is not open!";
9667 error( RtAudioError::INVALID_USE );
9671 void RtApi :: clearStreamInfo()
9673 stream_.mode = UNINITIALIZED;
9674 stream_.state = STREAM_CLOSED;
9675 stream_.sampleRate = 0;
9676 stream_.bufferSize = 0;
9677 stream_.nBuffers = 0;
9678 stream_.userFormat = 0;
9679 stream_.userInterleaved = true;
9680 stream_.streamTime = 0.0;
9681 stream_.apiHandle = 0;
9682 stream_.deviceBuffer = 0;
9683 stream_.callbackInfo.callback = 0;
9684 stream_.callbackInfo.userData = 0;
9685 stream_.callbackInfo.isRunning = false;
9686 stream_.callbackInfo.errorCallback = 0;
9687 for ( int i=0; i<2; i++ ) {
9688 stream_.device[i] = 11111;
9689 stream_.doConvertBuffer[i] = false;
9690 stream_.deviceInterleaved[i] = true;
9691 stream_.doByteSwap[i] = false;
9692 stream_.nUserChannels[i] = 0;
9693 stream_.nDeviceChannels[i] = 0;
9694 stream_.channelOffset[i] = 0;
9695 stream_.deviceFormat[i] = 0;
9696 stream_.latency[i] = 0;
9697 stream_.userBuffer[i] = 0;
9698 stream_.convertInfo[i].channels = 0;
9699 stream_.convertInfo[i].inJump = 0;
9700 stream_.convertInfo[i].outJump = 0;
9701 stream_.convertInfo[i].inFormat = 0;
9702 stream_.convertInfo[i].outFormat = 0;
9703 stream_.convertInfo[i].inOffset.clear();
9704 stream_.convertInfo[i].outOffset.clear();
9708 unsigned int RtApi :: formatBytes( RtAudioFormat format )
9710 if ( format == RTAUDIO_SINT16 )
9712 else if ( format == RTAUDIO_SINT32 || format == RTAUDIO_FLOAT32 )
9714 else if ( format == RTAUDIO_FLOAT64 )
9716 else if ( format == RTAUDIO_SINT24 )
9718 else if ( format == RTAUDIO_SINT8 )
9721 errorText_ = "RtApi::formatBytes: undefined format.";
9722 error( RtAudioError::WARNING );
9727 void RtApi :: setConvertInfo( StreamMode mode, unsigned int firstChannel )
9729 if ( mode == INPUT ) { // convert device to user buffer
9730 stream_.convertInfo[mode].inJump = stream_.nDeviceChannels[1];
9731 stream_.convertInfo[mode].outJump = stream_.nUserChannels[1];
9732 stream_.convertInfo[mode].inFormat = stream_.deviceFormat[1];
9733 stream_.convertInfo[mode].outFormat = stream_.userFormat;
9735 else { // convert user to device buffer
9736 stream_.convertInfo[mode].inJump = stream_.nUserChannels[0];
9737 stream_.convertInfo[mode].outJump = stream_.nDeviceChannels[0];
9738 stream_.convertInfo[mode].inFormat = stream_.userFormat;
9739 stream_.convertInfo[mode].outFormat = stream_.deviceFormat[0];
9742 if ( stream_.convertInfo[mode].inJump < stream_.convertInfo[mode].outJump )
9743 stream_.convertInfo[mode].channels = stream_.convertInfo[mode].inJump;
9745 stream_.convertInfo[mode].channels = stream_.convertInfo[mode].outJump;
9747 // Set up the interleave/deinterleave offsets.
9748 if ( stream_.deviceInterleaved[mode] != stream_.userInterleaved ) {
9749 if ( ( mode == OUTPUT && stream_.deviceInterleaved[mode] ) ||
9750 ( mode == INPUT && stream_.userInterleaved ) ) {
9751 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
9752 stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
9753 stream_.convertInfo[mode].outOffset.push_back( k );
9754 stream_.convertInfo[mode].inJump = 1;
9758 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
9759 stream_.convertInfo[mode].inOffset.push_back( k );
9760 stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
9761 stream_.convertInfo[mode].outJump = 1;
9765 else { // no (de)interleaving
9766 if ( stream_.userInterleaved ) {
9767 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
9768 stream_.convertInfo[mode].inOffset.push_back( k );
9769 stream_.convertInfo[mode].outOffset.push_back( k );
9773 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
9774 stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
9775 stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
9776 stream_.convertInfo[mode].inJump = 1;
9777 stream_.convertInfo[mode].outJump = 1;
9782 // Add channel offset.
9783 if ( firstChannel > 0 ) {
9784 if ( stream_.deviceInterleaved[mode] ) {
9785 if ( mode == OUTPUT ) {
9786 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
9787 stream_.convertInfo[mode].outOffset[k] += firstChannel;
9790 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
9791 stream_.convertInfo[mode].inOffset[k] += firstChannel;
9795 if ( mode == OUTPUT ) {
9796 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
9797 stream_.convertInfo[mode].outOffset[k] += ( firstChannel * stream_.bufferSize );
9800 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
9801 stream_.convertInfo[mode].inOffset[k] += ( firstChannel * stream_.bufferSize );
9807 void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info )
9809 // This function does format conversion, input/output channel compensation, and
9810 // data interleaving/deinterleaving. 24-bit integers are assumed to occupy
9811 // the lower three bytes of a 32-bit integer.
9813 // Clear our device buffer when in/out duplex device channels are different
9814 if ( outBuffer == stream_.deviceBuffer && stream_.mode == DUPLEX &&
9815 ( stream_.nDeviceChannels[0] < stream_.nDeviceChannels[1] ) )
9816 memset( outBuffer, 0, stream_.bufferSize * info.outJump * formatBytes( info.outFormat ) );
9819 if (info.outFormat == RTAUDIO_FLOAT64) {
9821 Float64 *out = (Float64 *)outBuffer;
9823 if (info.inFormat == RTAUDIO_SINT8) {
9824 signed char *in = (signed char *)inBuffer;
9825 scale = 1.0 / 127.5;
9826 for (unsigned int i=0; i<stream_.bufferSize; i++) {
9827 for (j=0; j<info.channels; j++) {
9828 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
9829 out[info.outOffset[j]] += 0.5;
9830 out[info.outOffset[j]] *= scale;
9833 out += info.outJump;
9836 else if (info.inFormat == RTAUDIO_SINT16) {
9837 Int16 *in = (Int16 *)inBuffer;
9838 scale = 1.0 / 32767.5;
9839 for (unsigned int i=0; i<stream_.bufferSize; i++) {
9840 for (j=0; j<info.channels; j++) {
9841 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
9842 out[info.outOffset[j]] += 0.5;
9843 out[info.outOffset[j]] *= scale;
9846 out += info.outJump;
9849 else if (info.inFormat == RTAUDIO_SINT24) {
9850 Int24 *in = (Int24 *)inBuffer;
9851 scale = 1.0 / 8388607.5;
9852 for (unsigned int i=0; i<stream_.bufferSize; i++) {
9853 for (j=0; j<info.channels; j++) {
9854 out[info.outOffset[j]] = (Float64) (in[info.inOffset[j]].asInt());
9855 out[info.outOffset[j]] += 0.5;
9856 out[info.outOffset[j]] *= scale;
9859 out += info.outJump;
9862 else if (info.inFormat == RTAUDIO_SINT32) {
9863 Int32 *in = (Int32 *)inBuffer;
9864 scale = 1.0 / 2147483647.5;
9865 for (unsigned int i=0; i<stream_.bufferSize; i++) {
9866 for (j=0; j<info.channels; j++) {
9867 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
9868 out[info.outOffset[j]] += 0.5;
9869 out[info.outOffset[j]] *= scale;
9872 out += info.outJump;
9875 else if (info.inFormat == RTAUDIO_FLOAT32) {
9876 Float32 *in = (Float32 *)inBuffer;
9877 for (unsigned int i=0; i<stream_.bufferSize; i++) {
9878 for (j=0; j<info.channels; j++) {
9879 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
9882 out += info.outJump;
9885 else if (info.inFormat == RTAUDIO_FLOAT64) {
9886 // Channel compensation and/or (de)interleaving only.
9887 Float64 *in = (Float64 *)inBuffer;
9888 for (unsigned int i=0; i<stream_.bufferSize; i++) {
9889 for (j=0; j<info.channels; j++) {
9890 out[info.outOffset[j]] = in[info.inOffset[j]];
9893 out += info.outJump;
9897 else if (info.outFormat == RTAUDIO_FLOAT32) {
9899 Float32 *out = (Float32 *)outBuffer;
9901 if (info.inFormat == RTAUDIO_SINT8) {
9902 signed char *in = (signed char *)inBuffer;
9903 scale = (Float32) ( 1.0 / 127.5 );
9904 for (unsigned int i=0; i<stream_.bufferSize; i++) {
9905 for (j=0; j<info.channels; j++) {
9906 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
9907 out[info.outOffset[j]] += 0.5;
9908 out[info.outOffset[j]] *= scale;
9911 out += info.outJump;
9914 else if (info.inFormat == RTAUDIO_SINT16) {
9915 Int16 *in = (Int16 *)inBuffer;
9916 scale = (Float32) ( 1.0 / 32767.5 );
9917 for (unsigned int i=0; i<stream_.bufferSize; i++) {
9918 for (j=0; j<info.channels; j++) {
9919 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
9920 out[info.outOffset[j]] += 0.5;
9921 out[info.outOffset[j]] *= scale;
9924 out += info.outJump;
9927 else if (info.inFormat == RTAUDIO_SINT24) {
9928 Int24 *in = (Int24 *)inBuffer;
9929 scale = (Float32) ( 1.0 / 8388607.5 );
9930 for (unsigned int i=0; i<stream_.bufferSize; i++) {
9931 for (j=0; j<info.channels; j++) {
9932 out[info.outOffset[j]] = (Float32) (in[info.inOffset[j]].asInt());
9933 out[info.outOffset[j]] += 0.5;
9934 out[info.outOffset[j]] *= scale;
9937 out += info.outJump;
9940 else if (info.inFormat == RTAUDIO_SINT32) {
9941 Int32 *in = (Int32 *)inBuffer;
9942 scale = (Float32) ( 1.0 / 2147483647.5 );
9943 for (unsigned int i=0; i<stream_.bufferSize; i++) {
9944 for (j=0; j<info.channels; j++) {
9945 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
9946 out[info.outOffset[j]] += 0.5;
9947 out[info.outOffset[j]] *= scale;
9950 out += info.outJump;
9953 else if (info.inFormat == RTAUDIO_FLOAT32) {
9954 // Channel compensation and/or (de)interleaving only.
9955 Float32 *in = (Float32 *)inBuffer;
9956 for (unsigned int i=0; i<stream_.bufferSize; i++) {
9957 for (j=0; j<info.channels; j++) {
9958 out[info.outOffset[j]] = in[info.inOffset[j]];
9961 out += info.outJump;
9964 else if (info.inFormat == RTAUDIO_FLOAT64) {
9965 Float64 *in = (Float64 *)inBuffer;
9966 for (unsigned int i=0; i<stream_.bufferSize; i++) {
9967 for (j=0; j<info.channels; j++) {
9968 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
9971 out += info.outJump;
9975 else if (info.outFormat == RTAUDIO_SINT32) {
9976 Int32 *out = (Int32 *)outBuffer;
9977 if (info.inFormat == RTAUDIO_SINT8) {
9978 signed char *in = (signed char *)inBuffer;
9979 for (unsigned int i=0; i<stream_.bufferSize; i++) {
9980 for (j=0; j<info.channels; j++) {
9981 out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
9982 out[info.outOffset[j]] <<= 24;
9985 out += info.outJump;
9988 else if (info.inFormat == RTAUDIO_SINT16) {
9989 Int16 *in = (Int16 *)inBuffer;
9990 for (unsigned int i=0; i<stream_.bufferSize; i++) {
9991 for (j=0; j<info.channels; j++) {
9992 out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
9993 out[info.outOffset[j]] <<= 16;
9996 out += info.outJump;
9999 else if (info.inFormat == RTAUDIO_SINT24) {
10000 Int24 *in = (Int24 *)inBuffer;
10001 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10002 for (j=0; j<info.channels; j++) {
10003 out[info.outOffset[j]] = (Int32) in[info.inOffset[j]].asInt();
10004 out[info.outOffset[j]] <<= 8;
10007 out += info.outJump;
10010 else if (info.inFormat == RTAUDIO_SINT32) {
10011 // Channel compensation and/or (de)interleaving only.
10012 Int32 *in = (Int32 *)inBuffer;
10013 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10014 for (j=0; j<info.channels; j++) {
10015 out[info.outOffset[j]] = in[info.inOffset[j]];
10018 out += info.outJump;
10021 else if (info.inFormat == RTAUDIO_FLOAT32) {
10022 Float32 *in = (Float32 *)inBuffer;
10023 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10024 for (j=0; j<info.channels; j++) {
10025 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
10028 out += info.outJump;
10031 else if (info.inFormat == RTAUDIO_FLOAT64) {
10032 Float64 *in = (Float64 *)inBuffer;
10033 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10034 for (j=0; j<info.channels; j++) {
10035 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
10038 out += info.outJump;
10042 else if (info.outFormat == RTAUDIO_SINT24) {
10043 Int24 *out = (Int24 *)outBuffer;
10044 if (info.inFormat == RTAUDIO_SINT8) {
10045 signed char *in = (signed char *)inBuffer;
10046 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10047 for (j=0; j<info.channels; j++) {
10048 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] << 16);
10049 //out[info.outOffset[j]] <<= 16;
10052 out += info.outJump;
10055 else if (info.inFormat == RTAUDIO_SINT16) {
10056 Int16 *in = (Int16 *)inBuffer;
10057 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10058 for (j=0; j<info.channels; j++) {
10059 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] << 8);
10060 //out[info.outOffset[j]] <<= 8;
10063 out += info.outJump;
10066 else if (info.inFormat == RTAUDIO_SINT24) {
10067 // Channel compensation and/or (de)interleaving only.
10068 Int24 *in = (Int24 *)inBuffer;
10069 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10070 for (j=0; j<info.channels; j++) {
10071 out[info.outOffset[j]] = in[info.inOffset[j]];
10074 out += info.outJump;
10077 else if (info.inFormat == RTAUDIO_SINT32) {
10078 Int32 *in = (Int32 *)inBuffer;
10079 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10080 for (j=0; j<info.channels; j++) {
10081 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] >> 8);
10082 //out[info.outOffset[j]] >>= 8;
10085 out += info.outJump;
10088 else if (info.inFormat == RTAUDIO_FLOAT32) {
10089 Float32 *in = (Float32 *)inBuffer;
10090 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10091 for (j=0; j<info.channels; j++) {
10092 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
10095 out += info.outJump;
10098 else if (info.inFormat == RTAUDIO_FLOAT64) {
10099 Float64 *in = (Float64 *)inBuffer;
10100 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10101 for (j=0; j<info.channels; j++) {
10102 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
10105 out += info.outJump;
10109 else if (info.outFormat == RTAUDIO_SINT16) {
10110 Int16 *out = (Int16 *)outBuffer;
10111 if (info.inFormat == RTAUDIO_SINT8) {
10112 signed char *in = (signed char *)inBuffer;
10113 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10114 for (j=0; j<info.channels; j++) {
10115 out[info.outOffset[j]] = (Int16) in[info.inOffset[j]];
10116 out[info.outOffset[j]] <<= 8;
10119 out += info.outJump;
10122 else if (info.inFormat == RTAUDIO_SINT16) {
10123 // Channel compensation and/or (de)interleaving only.
10124 Int16 *in = (Int16 *)inBuffer;
10125 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10126 for (j=0; j<info.channels; j++) {
10127 out[info.outOffset[j]] = in[info.inOffset[j]];
10130 out += info.outJump;
10133 else if (info.inFormat == RTAUDIO_SINT24) {
10134 Int24 *in = (Int24 *)inBuffer;
10135 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10136 for (j=0; j<info.channels; j++) {
10137 out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]].asInt() >> 8);
10140 out += info.outJump;
10143 else if (info.inFormat == RTAUDIO_SINT32) {
10144 Int32 *in = (Int32 *)inBuffer;
10145 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10146 for (j=0; j<info.channels; j++) {
10147 out[info.outOffset[j]] = (Int16) ((in[info.inOffset[j]] >> 16) & 0x0000ffff);
10150 out += info.outJump;
10153 else if (info.inFormat == RTAUDIO_FLOAT32) {
10154 Float32 *in = (Float32 *)inBuffer;
10155 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10156 for (j=0; j<info.channels; j++) {
10157 out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
10160 out += info.outJump;
10163 else if (info.inFormat == RTAUDIO_FLOAT64) {
10164 Float64 *in = (Float64 *)inBuffer;
10165 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10166 for (j=0; j<info.channels; j++) {
10167 out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
10170 out += info.outJump;
10174 else if (info.outFormat == RTAUDIO_SINT8) {
10175 signed char *out = (signed char *)outBuffer;
10176 if (info.inFormat == RTAUDIO_SINT8) {
10177 // Channel compensation and/or (de)interleaving only.
10178 signed char *in = (signed char *)inBuffer;
10179 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10180 for (j=0; j<info.channels; j++) {
10181 out[info.outOffset[j]] = in[info.inOffset[j]];
10184 out += info.outJump;
10187 if (info.inFormat == RTAUDIO_SINT16) {
10188 Int16 *in = (Int16 *)inBuffer;
10189 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10190 for (j=0; j<info.channels; j++) {
10191 out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 8) & 0x00ff);
10194 out += info.outJump;
10197 else if (info.inFormat == RTAUDIO_SINT24) {
10198 Int24 *in = (Int24 *)inBuffer;
10199 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10200 for (j=0; j<info.channels; j++) {
10201 out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]].asInt() >> 16);
10204 out += info.outJump;
10207 else if (info.inFormat == RTAUDIO_SINT32) {
10208 Int32 *in = (Int32 *)inBuffer;
10209 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10210 for (j=0; j<info.channels; j++) {
10211 out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 24) & 0x000000ff);
10214 out += info.outJump;
10217 else if (info.inFormat == RTAUDIO_FLOAT32) {
10218 Float32 *in = (Float32 *)inBuffer;
10219 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10220 for (j=0; j<info.channels; j++) {
10221 out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
10224 out += info.outJump;
10227 else if (info.inFormat == RTAUDIO_FLOAT64) {
10228 Float64 *in = (Float64 *)inBuffer;
10229 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10230 for (j=0; j<info.channels; j++) {
10231 out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
10234 out += info.outJump;
10240 //static inline uint16_t bswap_16(uint16_t x) { return (x>>8) | (x<<8); }
10241 //static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); }
10242 //static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); }
10244 void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format )
10250 if ( format == RTAUDIO_SINT16 ) {
10251 for ( unsigned int i=0; i<samples; i++ ) {
10252 // Swap 1st and 2nd bytes.
10257 // Increment 2 bytes.
10261 else if ( format == RTAUDIO_SINT32 ||
10262 format == RTAUDIO_FLOAT32 ) {
10263 for ( unsigned int i=0; i<samples; i++ ) {
10264 // Swap 1st and 4th bytes.
10269 // Swap 2nd and 3rd bytes.
10275 // Increment 3 more bytes.
10279 else if ( format == RTAUDIO_SINT24 ) {
10280 for ( unsigned int i=0; i<samples; i++ ) {
10281 // Swap 1st and 3rd bytes.
10286 // Increment 2 more bytes.
10290 else if ( format == RTAUDIO_FLOAT64 ) {
10291 for ( unsigned int i=0; i<samples; i++ ) {
10292 // Swap 1st and 8th bytes
10297 // Swap 2nd and 7th bytes
10303 // Swap 3rd and 6th bytes
10309 // Swap 4th and 5th bytes
10315 // Increment 5 more bytes.
10321 // Indentation settings for Vim and Emacs
10323 // Local Variables:
10324 // c-basic-offset: 2
10325 // indent-tabs-mode: nil
10328 // vim: et sts=2 sw=2