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 // for cmath functions
3887 using namespace std;
3889 outSampleCount = ( unsigned int ) roundf( inSampleCount * sampleRatio );
3891 // if inSampleRate is a multiple of outSampleRate (or vice versa) there's no need to interpolate
3892 if ( floor( sampleRatio ) == sampleRatio || floor( sampleRatioInv ) == sampleRatioInv )
3894 // frame-by-frame, copy each relative input sample into it's corresponding output sample
3895 for ( unsigned int outSample = 0; outSample < outSampleCount; outSample++ )
3897 unsigned int inSample = ( unsigned int ) inSampleFraction;
3902 memcpy( &( ( char* ) outBuffer )[ outSample * channelCount ], &( ( char* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( char ) );
3904 case RTAUDIO_SINT16:
3905 memcpy( &( ( short* ) outBuffer )[ outSample * channelCount ], &( ( short* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( short ) );
3907 case RTAUDIO_SINT24:
3908 memcpy( &( ( S24* ) outBuffer )[ outSample * channelCount ], &( ( S24* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( S24 ) );
3910 case RTAUDIO_SINT32:
3911 memcpy( &( ( int* ) outBuffer )[ outSample * channelCount ], &( ( int* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( int ) );
3913 case RTAUDIO_FLOAT32:
3914 memcpy( &( ( float* ) outBuffer )[ outSample * channelCount ], &( ( float* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( float ) );
3916 case RTAUDIO_FLOAT64:
3917 memcpy( &( ( double* ) outBuffer )[ outSample * channelCount ], &( ( double* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( double ) );
3921 // jump to next in sample
3922 inSampleFraction += sampleStep;
3925 else // else interpolate
3927 // frame-by-frame, copy each relative input sample into it's corresponding output sample
3928 for ( unsigned int outSample = 0; outSample < outSampleCount; outSample++ )
3930 unsigned int inSample = ( unsigned int ) inSampleFraction;
3931 float inSampleDec = inSampleFraction - inSample;
3932 unsigned int frameInSample = inSample * channelCount;
3933 unsigned int frameOutSample = outSample * channelCount;
3939 for ( unsigned int channel = 0; channel < channelCount; channel++ )
3941 char fromSample = ( ( char* ) inBuffer )[ frameInSample + channel ];
3942 char toSample = ( ( char* ) inBuffer )[ frameInSample + channelCount + channel ];
3943 char sampleDiff = ( char ) ( ( toSample - fromSample ) * inSampleDec );
3944 ( ( char* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;
3948 case RTAUDIO_SINT16:
3950 for ( unsigned int channel = 0; channel < channelCount; channel++ )
3952 short fromSample = ( ( short* ) inBuffer )[ frameInSample + channel ];
3953 short toSample = ( ( short* ) inBuffer )[ frameInSample + channelCount + channel ];
3954 short sampleDiff = ( short ) ( ( toSample - fromSample ) * inSampleDec );
3955 ( ( short* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;
3959 case RTAUDIO_SINT24:
3961 for ( unsigned int channel = 0; channel < channelCount; channel++ )
3963 int fromSample = ( ( S24* ) inBuffer )[ frameInSample + channel ].asInt();
3964 int toSample = ( ( S24* ) inBuffer )[ frameInSample + channelCount + channel ].asInt();
3965 int sampleDiff = ( int ) ( ( toSample - fromSample ) * inSampleDec );
3966 ( ( S24* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;
3970 case RTAUDIO_SINT32:
3972 for ( unsigned int channel = 0; channel < channelCount; channel++ )
3974 int fromSample = ( ( int* ) inBuffer )[ frameInSample + channel ];
3975 int toSample = ( ( int* ) inBuffer )[ frameInSample + channelCount + channel ];
3976 int sampleDiff = ( int ) ( ( toSample - fromSample ) * inSampleDec );
3977 ( ( int* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;
3981 case RTAUDIO_FLOAT32:
3983 for ( unsigned int channel = 0; channel < channelCount; channel++ )
3985 float fromSample = ( ( float* ) inBuffer )[ frameInSample + channel ];
3986 float toSample = ( ( float* ) inBuffer )[ frameInSample + channelCount + channel ];
3987 float sampleDiff = ( toSample - fromSample ) * inSampleDec;
3988 ( ( float* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;
3992 case RTAUDIO_FLOAT64:
3994 for ( unsigned int channel = 0; channel < channelCount; channel++ )
3996 double fromSample = ( ( double* ) inBuffer )[ frameInSample + channel ];
3997 double toSample = ( ( double* ) inBuffer )[ frameInSample + channelCount + channel ];
3998 double sampleDiff = ( toSample - fromSample ) * inSampleDec;
3999 ( ( double* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;
4005 // jump to next in sample
4006 inSampleFraction += sampleStep;
4011 //-----------------------------------------------------------------------------
4013 // A structure to hold various information related to the WASAPI implementation.
4016 IAudioClient* captureAudioClient;
4017 IAudioClient* renderAudioClient;
4018 IAudioCaptureClient* captureClient;
4019 IAudioRenderClient* renderClient;
4020 HANDLE captureEvent;
4024 : captureAudioClient( NULL ),
4025 renderAudioClient( NULL ),
4026 captureClient( NULL ),
4027 renderClient( NULL ),
4028 captureEvent( NULL ),
4029 renderEvent( NULL ) {}
4032 //=============================================================================
4034 RtApiWasapi::RtApiWasapi()
4035 : coInitialized_( false ), deviceEnumerator_( NULL )
4037 // WASAPI can run either apartment or multi-threaded
4038 HRESULT hr = CoInitialize( NULL );
4039 if ( !FAILED( hr ) )
4040 coInitialized_ = true;
4042 // Instantiate device enumerator
4043 hr = CoCreateInstance( __uuidof( MMDeviceEnumerator ), NULL,
4044 CLSCTX_ALL, __uuidof( IMMDeviceEnumerator ),
4045 ( void** ) &deviceEnumerator_ );
4047 if ( FAILED( hr ) ) {
4048 errorText_ = "RtApiWasapi::RtApiWasapi: Unable to instantiate device enumerator";
4049 error( RtAudioError::DRIVER_ERROR );
4053 //-----------------------------------------------------------------------------
4055 RtApiWasapi::~RtApiWasapi()
4057 if ( stream_.state != STREAM_CLOSED )
4060 SAFE_RELEASE( deviceEnumerator_ );
4062 // If this object previously called CoInitialize()
4063 if ( coInitialized_ )
4067 //=============================================================================
4069 unsigned int RtApiWasapi::getDeviceCount( void )
4071 unsigned int captureDeviceCount = 0;
4072 unsigned int renderDeviceCount = 0;
4074 IMMDeviceCollection* captureDevices = NULL;
4075 IMMDeviceCollection* renderDevices = NULL;
4077 // Count capture devices
4079 HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
4080 if ( FAILED( hr ) ) {
4081 errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device collection.";
4085 hr = captureDevices->GetCount( &captureDeviceCount );
4086 if ( FAILED( hr ) ) {
4087 errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device count.";
4091 // Count render devices
4092 hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
4093 if ( FAILED( hr ) ) {
4094 errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device collection.";
4098 hr = renderDevices->GetCount( &renderDeviceCount );
4099 if ( FAILED( hr ) ) {
4100 errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device count.";
4105 // release all references
4106 SAFE_RELEASE( captureDevices );
4107 SAFE_RELEASE( renderDevices );
4109 if ( errorText_.empty() )
4110 return captureDeviceCount + renderDeviceCount;
4112 error( RtAudioError::DRIVER_ERROR );
4116 //-----------------------------------------------------------------------------
4118 RtAudio::DeviceInfo RtApiWasapi::getDeviceInfo( unsigned int device )
4120 RtAudio::DeviceInfo info;
4121 unsigned int captureDeviceCount = 0;
4122 unsigned int renderDeviceCount = 0;
4123 std::string defaultDeviceName;
4124 bool isCaptureDevice = false;
4126 PROPVARIANT deviceNameProp;
4127 PROPVARIANT defaultDeviceNameProp;
4129 IMMDeviceCollection* captureDevices = NULL;
4130 IMMDeviceCollection* renderDevices = NULL;
4131 IMMDevice* devicePtr = NULL;
4132 IMMDevice* defaultDevicePtr = NULL;
4133 IAudioClient* audioClient = NULL;
4134 IPropertyStore* devicePropStore = NULL;
4135 IPropertyStore* defaultDevicePropStore = NULL;
4137 WAVEFORMATEX* deviceFormat = NULL;
4138 WAVEFORMATEX* closestMatchFormat = NULL;
4141 info.probed = false;
4143 // Count capture devices
4145 RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
4146 HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
4147 if ( FAILED( hr ) ) {
4148 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device collection.";
4152 hr = captureDevices->GetCount( &captureDeviceCount );
4153 if ( FAILED( hr ) ) {
4154 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device count.";
4158 // Count render devices
4159 hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
4160 if ( FAILED( hr ) ) {
4161 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device collection.";
4165 hr = renderDevices->GetCount( &renderDeviceCount );
4166 if ( FAILED( hr ) ) {
4167 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device count.";
4171 // validate device index
4172 if ( device >= captureDeviceCount + renderDeviceCount ) {
4173 errorText_ = "RtApiWasapi::getDeviceInfo: Invalid device index.";
4174 errorType = RtAudioError::INVALID_USE;
4178 // determine whether index falls within capture or render devices
4179 if ( device >= renderDeviceCount ) {
4180 hr = captureDevices->Item( device - renderDeviceCount, &devicePtr );
4181 if ( FAILED( hr ) ) {
4182 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device handle.";
4185 isCaptureDevice = true;
4188 hr = renderDevices->Item( device, &devicePtr );
4189 if ( FAILED( hr ) ) {
4190 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device handle.";
4193 isCaptureDevice = false;
4196 // get default device name
4197 if ( isCaptureDevice ) {
4198 hr = deviceEnumerator_->GetDefaultAudioEndpoint( eCapture, eConsole, &defaultDevicePtr );
4199 if ( FAILED( hr ) ) {
4200 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default capture device handle.";
4205 hr = deviceEnumerator_->GetDefaultAudioEndpoint( eRender, eConsole, &defaultDevicePtr );
4206 if ( FAILED( hr ) ) {
4207 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default render device handle.";
4212 hr = defaultDevicePtr->OpenPropertyStore( STGM_READ, &defaultDevicePropStore );
4213 if ( FAILED( hr ) ) {
4214 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open default device property store.";
4217 PropVariantInit( &defaultDeviceNameProp );
4219 hr = defaultDevicePropStore->GetValue( PKEY_Device_FriendlyName, &defaultDeviceNameProp );
4220 if ( FAILED( hr ) ) {
4221 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default device property: PKEY_Device_FriendlyName.";
4225 defaultDeviceName = convertCharPointerToStdString(defaultDeviceNameProp.pwszVal);
4228 hr = devicePtr->OpenPropertyStore( STGM_READ, &devicePropStore );
4229 if ( FAILED( hr ) ) {
4230 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open device property store.";
4234 PropVariantInit( &deviceNameProp );
4236 hr = devicePropStore->GetValue( PKEY_Device_FriendlyName, &deviceNameProp );
4237 if ( FAILED( hr ) ) {
4238 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device property: PKEY_Device_FriendlyName.";
4242 info.name =convertCharPointerToStdString(deviceNameProp.pwszVal);
4245 if ( isCaptureDevice ) {
4246 info.isDefaultInput = info.name == defaultDeviceName;
4247 info.isDefaultOutput = false;
4250 info.isDefaultInput = false;
4251 info.isDefaultOutput = info.name == defaultDeviceName;
4255 hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, NULL, ( void** ) &audioClient );
4256 if ( FAILED( hr ) ) {
4257 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device audio client.";
4261 hr = audioClient->GetMixFormat( &deviceFormat );
4262 if ( FAILED( hr ) ) {
4263 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device mix format.";
4267 if ( isCaptureDevice ) {
4268 info.inputChannels = deviceFormat->nChannels;
4269 info.outputChannels = 0;
4270 info.duplexChannels = 0;
4273 info.inputChannels = 0;
4274 info.outputChannels = deviceFormat->nChannels;
4275 info.duplexChannels = 0;
4279 info.sampleRates.clear();
4281 // allow support for all sample rates as we have a built-in sample rate converter
4282 for ( unsigned int i = 0; i < MAX_SAMPLE_RATES; i++ ) {
4283 info.sampleRates.push_back( SAMPLE_RATES[i] );
4285 info.preferredSampleRate = deviceFormat->nSamplesPerSec;
4288 info.nativeFormats = 0;
4290 if ( deviceFormat->wFormatTag == WAVE_FORMAT_IEEE_FLOAT ||
4291 ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
4292 ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_IEEE_FLOAT ) )
4294 if ( deviceFormat->wBitsPerSample == 32 ) {
4295 info.nativeFormats |= RTAUDIO_FLOAT32;
4297 else if ( deviceFormat->wBitsPerSample == 64 ) {
4298 info.nativeFormats |= RTAUDIO_FLOAT64;
4301 else if ( deviceFormat->wFormatTag == WAVE_FORMAT_PCM ||
4302 ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
4303 ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_PCM ) )
4305 if ( deviceFormat->wBitsPerSample == 8 ) {
4306 info.nativeFormats |= RTAUDIO_SINT8;
4308 else if ( deviceFormat->wBitsPerSample == 16 ) {
4309 info.nativeFormats |= RTAUDIO_SINT16;
4311 else if ( deviceFormat->wBitsPerSample == 24 ) {
4312 info.nativeFormats |= RTAUDIO_SINT24;
4314 else if ( deviceFormat->wBitsPerSample == 32 ) {
4315 info.nativeFormats |= RTAUDIO_SINT32;
4323 // release all references
4324 PropVariantClear( &deviceNameProp );
4325 PropVariantClear( &defaultDeviceNameProp );
4327 SAFE_RELEASE( captureDevices );
4328 SAFE_RELEASE( renderDevices );
4329 SAFE_RELEASE( devicePtr );
4330 SAFE_RELEASE( defaultDevicePtr );
4331 SAFE_RELEASE( audioClient );
4332 SAFE_RELEASE( devicePropStore );
4333 SAFE_RELEASE( defaultDevicePropStore );
4335 CoTaskMemFree( deviceFormat );
4336 CoTaskMemFree( closestMatchFormat );
4338 if ( !errorText_.empty() )
4343 //-----------------------------------------------------------------------------
4345 unsigned int RtApiWasapi::getDefaultOutputDevice( void )
4347 for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {
4348 if ( getDeviceInfo( i ).isDefaultOutput ) {
4356 //-----------------------------------------------------------------------------
4358 unsigned int RtApiWasapi::getDefaultInputDevice( void )
4360 for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {
4361 if ( getDeviceInfo( i ).isDefaultInput ) {
4369 //-----------------------------------------------------------------------------
4371 void RtApiWasapi::closeStream( void )
4373 if ( stream_.state == STREAM_CLOSED ) {
4374 errorText_ = "RtApiWasapi::closeStream: No open stream to close.";
4375 error( RtAudioError::WARNING );
4379 if ( stream_.state != STREAM_STOPPED )
4382 // clean up stream memory
4383 SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient )
4384 SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient )
4386 SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureClient )
4387 SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderClient )
4389 if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent )
4390 CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent );
4392 if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent )
4393 CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent );
4395 delete ( WasapiHandle* ) stream_.apiHandle;
4396 stream_.apiHandle = NULL;
4398 for ( int i = 0; i < 2; i++ ) {
4399 if ( stream_.userBuffer[i] ) {
4400 free( stream_.userBuffer[i] );
4401 stream_.userBuffer[i] = 0;
4405 if ( stream_.deviceBuffer ) {
4406 free( stream_.deviceBuffer );
4407 stream_.deviceBuffer = 0;
4410 // update stream state
4411 stream_.state = STREAM_CLOSED;
4414 //-----------------------------------------------------------------------------
4416 void RtApiWasapi::startStream( void )
4420 if ( stream_.state == STREAM_RUNNING ) {
4421 errorText_ = "RtApiWasapi::startStream: The stream is already running.";
4422 error( RtAudioError::WARNING );
4426 // update stream state
4427 stream_.state = STREAM_RUNNING;
4429 // create WASAPI stream thread
4430 stream_.callbackInfo.thread = ( ThreadHandle ) CreateThread( NULL, 0, runWasapiThread, this, CREATE_SUSPENDED, NULL );
4432 if ( !stream_.callbackInfo.thread ) {
4433 errorText_ = "RtApiWasapi::startStream: Unable to instantiate callback thread.";
4434 error( RtAudioError::THREAD_ERROR );
4437 SetThreadPriority( ( void* ) stream_.callbackInfo.thread, stream_.callbackInfo.priority );
4438 ResumeThread( ( void* ) stream_.callbackInfo.thread );
4442 //-----------------------------------------------------------------------------
4444 void RtApiWasapi::stopStream( void )
4448 if ( stream_.state == STREAM_STOPPED ) {
4449 errorText_ = "RtApiWasapi::stopStream: The stream is already stopped.";
4450 error( RtAudioError::WARNING );
4454 // inform stream thread by setting stream state to STREAM_STOPPING
4455 stream_.state = STREAM_STOPPING;
4457 // wait until stream thread is stopped
4458 while( stream_.state != STREAM_STOPPED ) {
4462 // Wait for the last buffer to play before stopping.
4463 Sleep( 1000 * stream_.bufferSize / stream_.sampleRate );
4465 // stop capture client if applicable
4466 if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) {
4467 HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop();
4468 if ( FAILED( hr ) ) {
4469 errorText_ = "RtApiWasapi::stopStream: Unable to stop capture stream.";
4470 error( RtAudioError::DRIVER_ERROR );
4475 // stop render client if applicable
4476 if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) {
4477 HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop();
4478 if ( FAILED( hr ) ) {
4479 errorText_ = "RtApiWasapi::stopStream: Unable to stop render stream.";
4480 error( RtAudioError::DRIVER_ERROR );
4485 // close thread handle
4486 if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
4487 errorText_ = "RtApiWasapi::stopStream: Unable to close callback thread.";
4488 error( RtAudioError::THREAD_ERROR );
4492 stream_.callbackInfo.thread = (ThreadHandle) NULL;
4495 //-----------------------------------------------------------------------------
4497 void RtApiWasapi::abortStream( void )
4501 if ( stream_.state == STREAM_STOPPED ) {
4502 errorText_ = "RtApiWasapi::abortStream: The stream is already stopped.";
4503 error( RtAudioError::WARNING );
4507 // inform stream thread by setting stream state to STREAM_STOPPING
4508 stream_.state = STREAM_STOPPING;
4510 // wait until stream thread is stopped
4511 while ( stream_.state != STREAM_STOPPED ) {
4515 // stop capture client if applicable
4516 if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) {
4517 HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop();
4518 if ( FAILED( hr ) ) {
4519 errorText_ = "RtApiWasapi::abortStream: Unable to stop capture stream.";
4520 error( RtAudioError::DRIVER_ERROR );
4525 // stop render client if applicable
4526 if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) {
4527 HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop();
4528 if ( FAILED( hr ) ) {
4529 errorText_ = "RtApiWasapi::abortStream: Unable to stop render stream.";
4530 error( RtAudioError::DRIVER_ERROR );
4535 // close thread handle
4536 if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
4537 errorText_ = "RtApiWasapi::abortStream: Unable to close callback thread.";
4538 error( RtAudioError::THREAD_ERROR );
4542 stream_.callbackInfo.thread = (ThreadHandle) NULL;
4545 //-----------------------------------------------------------------------------
4547 bool RtApiWasapi::probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
4548 unsigned int firstChannel, unsigned int sampleRate,
4549 RtAudioFormat format, unsigned int* bufferSize,
4550 RtAudio::StreamOptions* options )
4552 bool methodResult = FAILURE;
4553 unsigned int captureDeviceCount = 0;
4554 unsigned int renderDeviceCount = 0;
4556 IMMDeviceCollection* captureDevices = NULL;
4557 IMMDeviceCollection* renderDevices = NULL;
4558 IMMDevice* devicePtr = NULL;
4559 WAVEFORMATEX* deviceFormat = NULL;
4560 unsigned int bufferBytes;
4561 stream_.state = STREAM_STOPPED;
4563 // create API Handle if not already created
4564 if ( !stream_.apiHandle )
4565 stream_.apiHandle = ( void* ) new WasapiHandle();
4567 // Count capture devices
4569 RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
4570 HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
4571 if ( FAILED( hr ) ) {
4572 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device collection.";
4576 hr = captureDevices->GetCount( &captureDeviceCount );
4577 if ( FAILED( hr ) ) {
4578 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device count.";
4582 // Count render devices
4583 hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
4584 if ( FAILED( hr ) ) {
4585 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device collection.";
4589 hr = renderDevices->GetCount( &renderDeviceCount );
4590 if ( FAILED( hr ) ) {
4591 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device count.";
4595 // validate device index
4596 if ( device >= captureDeviceCount + renderDeviceCount ) {
4597 errorType = RtAudioError::INVALID_USE;
4598 errorText_ = "RtApiWasapi::probeDeviceOpen: Invalid device index.";
4602 // determine whether index falls within capture or render devices
4603 if ( device >= renderDeviceCount ) {
4604 if ( mode != INPUT ) {
4605 errorType = RtAudioError::INVALID_USE;
4606 errorText_ = "RtApiWasapi::probeDeviceOpen: Capture device selected as output device.";
4610 // retrieve captureAudioClient from devicePtr
4611 IAudioClient*& captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
4613 hr = captureDevices->Item( device - renderDeviceCount, &devicePtr );
4614 if ( FAILED( hr ) ) {
4615 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device handle.";
4619 hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
4620 NULL, ( void** ) &captureAudioClient );
4621 if ( FAILED( hr ) ) {
4622 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client.";
4626 hr = captureAudioClient->GetMixFormat( &deviceFormat );
4627 if ( FAILED( hr ) ) {
4628 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format.";
4632 stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
4633 captureAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
4636 if ( mode != OUTPUT ) {
4637 errorType = RtAudioError::INVALID_USE;
4638 errorText_ = "RtApiWasapi::probeDeviceOpen: Render device selected as input device.";
4642 // retrieve renderAudioClient from devicePtr
4643 IAudioClient*& renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
4645 hr = renderDevices->Item( device, &devicePtr );
4646 if ( FAILED( hr ) ) {
4647 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device handle.";
4651 hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
4652 NULL, ( void** ) &renderAudioClient );
4653 if ( FAILED( hr ) ) {
4654 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client.";
4658 hr = renderAudioClient->GetMixFormat( &deviceFormat );
4659 if ( FAILED( hr ) ) {
4660 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format.";
4664 stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
4665 renderAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
4669 if ( ( stream_.mode == OUTPUT && mode == INPUT ) ||
4670 ( stream_.mode == INPUT && mode == OUTPUT ) ) {
4671 stream_.mode = DUPLEX;
4674 stream_.mode = mode;
4677 stream_.device[mode] = device;
4678 stream_.doByteSwap[mode] = false;
4679 stream_.sampleRate = sampleRate;
4680 stream_.bufferSize = *bufferSize;
4681 stream_.nBuffers = 1;
4682 stream_.nUserChannels[mode] = channels;
4683 stream_.channelOffset[mode] = firstChannel;
4684 stream_.userFormat = format;
4685 stream_.deviceFormat[mode] = getDeviceInfo( device ).nativeFormats;
4687 if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
4688 stream_.userInterleaved = false;
4690 stream_.userInterleaved = true;
4691 stream_.deviceInterleaved[mode] = true;
4693 // Set flags for buffer conversion.
4694 stream_.doConvertBuffer[mode] = false;
4695 if ( stream_.userFormat != stream_.deviceFormat[mode] ||
4696 stream_.nUserChannels != stream_.nDeviceChannels )
4697 stream_.doConvertBuffer[mode] = true;
4698 else if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
4699 stream_.nUserChannels[mode] > 1 )
4700 stream_.doConvertBuffer[mode] = true;
4702 if ( stream_.doConvertBuffer[mode] )
4703 setConvertInfo( mode, 0 );
4705 // Allocate necessary internal buffers
4706 bufferBytes = stream_.nUserChannels[mode] * stream_.bufferSize * formatBytes( stream_.userFormat );
4708 stream_.userBuffer[mode] = ( char* ) calloc( bufferBytes, 1 );
4709 if ( !stream_.userBuffer[mode] ) {
4710 errorType = RtAudioError::MEMORY_ERROR;
4711 errorText_ = "RtApiWasapi::probeDeviceOpen: Error allocating user buffer memory.";
4715 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME )
4716 stream_.callbackInfo.priority = 15;
4718 stream_.callbackInfo.priority = 0;
4720 ///! TODO: RTAUDIO_MINIMIZE_LATENCY // Provide stream buffers directly to callback
4721 ///! TODO: RTAUDIO_HOG_DEVICE // Exclusive mode
4723 methodResult = SUCCESS;
4727 SAFE_RELEASE( captureDevices );
4728 SAFE_RELEASE( renderDevices );
4729 SAFE_RELEASE( devicePtr );
4730 CoTaskMemFree( deviceFormat );
4732 // if method failed, close the stream
4733 if ( methodResult == FAILURE )
4736 if ( !errorText_.empty() )
4738 return methodResult;
4741 //=============================================================================
4743 DWORD WINAPI RtApiWasapi::runWasapiThread( void* wasapiPtr )
4746 ( ( RtApiWasapi* ) wasapiPtr )->wasapiThread();
4751 DWORD WINAPI RtApiWasapi::stopWasapiThread( void* wasapiPtr )
4754 ( ( RtApiWasapi* ) wasapiPtr )->stopStream();
4759 DWORD WINAPI RtApiWasapi::abortWasapiThread( void* wasapiPtr )
4762 ( ( RtApiWasapi* ) wasapiPtr )->abortStream();
4767 //-----------------------------------------------------------------------------
4769 void RtApiWasapi::wasapiThread()
4771 // as this is a new thread, we must CoInitialize it
4772 CoInitialize( NULL );
4776 IAudioClient* captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
4777 IAudioClient* renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
4778 IAudioCaptureClient* captureClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureClient;
4779 IAudioRenderClient* renderClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderClient;
4780 HANDLE captureEvent = ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent;
4781 HANDLE renderEvent = ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent;
4783 WAVEFORMATEX* captureFormat = NULL;
4784 WAVEFORMATEX* renderFormat = NULL;
4785 float captureSrRatio = 0.0f;
4786 float renderSrRatio = 0.0f;
4787 WasapiBuffer captureBuffer;
4788 WasapiBuffer renderBuffer;
4790 // declare local stream variables
4791 RtAudioCallback callback = ( RtAudioCallback ) stream_.callbackInfo.callback;
4792 BYTE* streamBuffer = NULL;
4793 unsigned long captureFlags = 0;
4794 unsigned int bufferFrameCount = 0;
4795 unsigned int numFramesPadding = 0;
4796 unsigned int convBufferSize = 0;
4797 bool callbackPushed = false;
4798 bool callbackPulled = false;
4799 bool callbackStopped = false;
4800 int callbackResult = 0;
4802 // convBuffer is used to store converted buffers between WASAPI and the user
4803 char* convBuffer = NULL;
4804 unsigned int convBuffSize = 0;
4805 unsigned int deviceBuffSize = 0;
4808 RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
4810 // Attempt to assign "Pro Audio" characteristic to thread
4811 HMODULE AvrtDll = LoadLibrary( (LPCTSTR) "AVRT.dll" );
4813 DWORD taskIndex = 0;
4814 TAvSetMmThreadCharacteristicsPtr AvSetMmThreadCharacteristicsPtr = ( TAvSetMmThreadCharacteristicsPtr ) GetProcAddress( AvrtDll, "AvSetMmThreadCharacteristicsW" );
4815 AvSetMmThreadCharacteristicsPtr( L"Pro Audio", &taskIndex );
4816 FreeLibrary( AvrtDll );
4819 // start capture stream if applicable
4820 if ( captureAudioClient ) {
4821 hr = captureAudioClient->GetMixFormat( &captureFormat );
4822 if ( FAILED( hr ) ) {
4823 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
4827 captureSrRatio = ( ( float ) captureFormat->nSamplesPerSec / stream_.sampleRate );
4829 // initialize capture stream according to desire buffer size
4830 float desiredBufferSize = stream_.bufferSize * captureSrRatio;
4831 REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / captureFormat->nSamplesPerSec );
4833 if ( !captureClient ) {
4834 hr = captureAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
4835 AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
4836 desiredBufferPeriod,
4837 desiredBufferPeriod,
4840 if ( FAILED( hr ) ) {
4841 errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize capture audio client.";
4845 hr = captureAudioClient->GetService( __uuidof( IAudioCaptureClient ),
4846 ( void** ) &captureClient );
4847 if ( FAILED( hr ) ) {
4848 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture client handle.";
4852 // configure captureEvent to trigger on every available capture buffer
4853 captureEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
4854 if ( !captureEvent ) {
4855 errorType = RtAudioError::SYSTEM_ERROR;
4856 errorText_ = "RtApiWasapi::wasapiThread: Unable to create capture event.";
4860 hr = captureAudioClient->SetEventHandle( captureEvent );
4861 if ( FAILED( hr ) ) {
4862 errorText_ = "RtApiWasapi::wasapiThread: Unable to set capture event handle.";
4866 ( ( WasapiHandle* ) stream_.apiHandle )->captureClient = captureClient;
4867 ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent = captureEvent;
4870 unsigned int inBufferSize = 0;
4871 hr = captureAudioClient->GetBufferSize( &inBufferSize );
4872 if ( FAILED( hr ) ) {
4873 errorText_ = "RtApiWasapi::wasapiThread: Unable to get capture buffer size.";
4877 // scale outBufferSize according to stream->user sample rate ratio
4878 unsigned int outBufferSize = ( unsigned int ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT];
4879 inBufferSize *= stream_.nDeviceChannels[INPUT];
4881 // set captureBuffer size
4882 captureBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[INPUT] ) );
4884 // reset the capture stream
4885 hr = captureAudioClient->Reset();
4886 if ( FAILED( hr ) ) {
4887 errorText_ = "RtApiWasapi::wasapiThread: Unable to reset capture stream.";
4891 // start the capture stream
4892 hr = captureAudioClient->Start();
4893 if ( FAILED( hr ) ) {
4894 errorText_ = "RtApiWasapi::wasapiThread: Unable to start capture stream.";
4899 // start render stream if applicable
4900 if ( renderAudioClient ) {
4901 hr = renderAudioClient->GetMixFormat( &renderFormat );
4902 if ( FAILED( hr ) ) {
4903 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
4907 renderSrRatio = ( ( float ) renderFormat->nSamplesPerSec / stream_.sampleRate );
4909 // initialize render stream according to desire buffer size
4910 float desiredBufferSize = stream_.bufferSize * renderSrRatio;
4911 REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / renderFormat->nSamplesPerSec );
4913 if ( !renderClient ) {
4914 hr = renderAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
4915 AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
4916 desiredBufferPeriod,
4917 desiredBufferPeriod,
4920 if ( FAILED( hr ) ) {
4921 errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize render audio client.";
4925 hr = renderAudioClient->GetService( __uuidof( IAudioRenderClient ),
4926 ( void** ) &renderClient );
4927 if ( FAILED( hr ) ) {
4928 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render client handle.";
4932 // configure renderEvent to trigger on every available render buffer
4933 renderEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
4934 if ( !renderEvent ) {
4935 errorType = RtAudioError::SYSTEM_ERROR;
4936 errorText_ = "RtApiWasapi::wasapiThread: Unable to create render event.";
4940 hr = renderAudioClient->SetEventHandle( renderEvent );
4941 if ( FAILED( hr ) ) {
4942 errorText_ = "RtApiWasapi::wasapiThread: Unable to set render event handle.";
4946 ( ( WasapiHandle* ) stream_.apiHandle )->renderClient = renderClient;
4947 ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent = renderEvent;
4950 unsigned int outBufferSize = 0;
4951 hr = renderAudioClient->GetBufferSize( &outBufferSize );
4952 if ( FAILED( hr ) ) {
4953 errorText_ = "RtApiWasapi::wasapiThread: Unable to get render buffer size.";
4957 // scale inBufferSize according to user->stream sample rate ratio
4958 unsigned int inBufferSize = ( unsigned int ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT];
4959 outBufferSize *= stream_.nDeviceChannels[OUTPUT];
4961 // set renderBuffer size
4962 renderBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[OUTPUT] ) );
4964 // reset the render stream
4965 hr = renderAudioClient->Reset();
4966 if ( FAILED( hr ) ) {
4967 errorText_ = "RtApiWasapi::wasapiThread: Unable to reset render stream.";
4971 // start the render stream
4972 hr = renderAudioClient->Start();
4973 if ( FAILED( hr ) ) {
4974 errorText_ = "RtApiWasapi::wasapiThread: Unable to start render stream.";
4979 if ( stream_.mode == INPUT ) {
4980 using namespace std; // for roundf
4981 convBuffSize = ( size_t ) roundf( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
4982 deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
4984 else if ( stream_.mode == OUTPUT ) {
4985 convBuffSize = ( size_t ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
4986 deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
4988 else if ( stream_.mode == DUPLEX ) {
4989 convBuffSize = std::max( ( size_t ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
4990 ( size_t ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
4991 deviceBuffSize = std::max( stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
4992 stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
4995 convBuffer = ( char* ) malloc( convBuffSize );
4996 stream_.deviceBuffer = ( char* ) malloc( deviceBuffSize );
4997 if ( !convBuffer || !stream_.deviceBuffer ) {
4998 errorType = RtAudioError::MEMORY_ERROR;
4999 errorText_ = "RtApiWasapi::wasapiThread: Error allocating device buffer memory.";
5003 // stream process loop
5004 while ( stream_.state != STREAM_STOPPING ) {
5005 if ( !callbackPulled ) {
5008 // 1. Pull callback buffer from inputBuffer
5009 // 2. If 1. was successful: Convert callback buffer to user sample rate and channel count
5010 // Convert callback buffer to user format
5012 if ( captureAudioClient ) {
5013 // Pull callback buffer from inputBuffer
5014 callbackPulled = captureBuffer.pullBuffer( convBuffer,
5015 ( unsigned int ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT],
5016 stream_.deviceFormat[INPUT] );
5018 if ( callbackPulled ) {
5019 // Convert callback buffer to user sample rate
5020 convertBufferWasapi( stream_.deviceBuffer,
5022 stream_.nDeviceChannels[INPUT],
5023 captureFormat->nSamplesPerSec,
5025 ( unsigned int ) ( stream_.bufferSize * captureSrRatio ),
5027 stream_.deviceFormat[INPUT] );
5029 if ( stream_.doConvertBuffer[INPUT] ) {
5030 // Convert callback buffer to user format
5031 convertBuffer( stream_.userBuffer[INPUT],
5032 stream_.deviceBuffer,
5033 stream_.convertInfo[INPUT] );
5036 // no further conversion, simple copy deviceBuffer to userBuffer
5037 memcpy( stream_.userBuffer[INPUT],
5038 stream_.deviceBuffer,
5039 stream_.bufferSize * stream_.nUserChannels[INPUT] * formatBytes( stream_.userFormat ) );
5044 // if there is no capture stream, set callbackPulled flag
5045 callbackPulled = true;
5050 // 1. Execute user callback method
5051 // 2. Handle return value from callback
5053 // if callback has not requested the stream to stop
5054 if ( callbackPulled && !callbackStopped ) {
5055 // Execute user callback method
5056 callbackResult = callback( stream_.userBuffer[OUTPUT],
5057 stream_.userBuffer[INPUT],
5060 captureFlags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY ? RTAUDIO_INPUT_OVERFLOW : 0,
5061 stream_.callbackInfo.userData );
5063 // Handle return value from callback
5064 if ( callbackResult == 1 ) {
5065 // instantiate a thread to stop this thread
5066 HANDLE threadHandle = CreateThread( NULL, 0, stopWasapiThread, this, 0, NULL );
5067 if ( !threadHandle ) {
5068 errorType = RtAudioError::THREAD_ERROR;
5069 errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream stop thread.";
5072 else if ( !CloseHandle( threadHandle ) ) {
5073 errorType = RtAudioError::THREAD_ERROR;
5074 errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream stop thread handle.";
5078 callbackStopped = true;
5080 else if ( callbackResult == 2 ) {
5081 // instantiate a thread to stop this thread
5082 HANDLE threadHandle = CreateThread( NULL, 0, abortWasapiThread, this, 0, NULL );
5083 if ( !threadHandle ) {
5084 errorType = RtAudioError::THREAD_ERROR;
5085 errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream abort thread.";
5088 else if ( !CloseHandle( threadHandle ) ) {
5089 errorType = RtAudioError::THREAD_ERROR;
5090 errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream abort thread handle.";
5094 callbackStopped = true;
5101 // 1. Convert callback buffer to stream format
5102 // 2. Convert callback buffer to stream sample rate and channel count
5103 // 3. Push callback buffer into outputBuffer
5105 if ( renderAudioClient && callbackPulled ) {
5106 if ( stream_.doConvertBuffer[OUTPUT] ) {
5107 // Convert callback buffer to stream format
5108 convertBuffer( stream_.deviceBuffer,
5109 stream_.userBuffer[OUTPUT],
5110 stream_.convertInfo[OUTPUT] );
5114 // Convert callback buffer to stream sample rate
5115 convertBufferWasapi( convBuffer,
5116 stream_.deviceBuffer,
5117 stream_.nDeviceChannels[OUTPUT],
5119 renderFormat->nSamplesPerSec,
5122 stream_.deviceFormat[OUTPUT] );
5124 // Push callback buffer into outputBuffer
5125 callbackPushed = renderBuffer.pushBuffer( convBuffer,
5126 convBufferSize * stream_.nDeviceChannels[OUTPUT],
5127 stream_.deviceFormat[OUTPUT] );
5130 // if there is no render stream, set callbackPushed flag
5131 callbackPushed = true;
5136 // 1. Get capture buffer from stream
5137 // 2. Push capture buffer into inputBuffer
5138 // 3. If 2. was successful: Release capture buffer
5140 if ( captureAudioClient ) {
5141 // if the callback input buffer was not pulled from captureBuffer, wait for next capture event
5142 if ( !callbackPulled ) {
5143 WaitForSingleObject( captureEvent, INFINITE );
5146 // Get capture buffer from stream
5147 hr = captureClient->GetBuffer( &streamBuffer,
5149 &captureFlags, NULL, NULL );
5150 if ( FAILED( hr ) ) {
5151 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture buffer.";
5155 if ( bufferFrameCount != 0 ) {
5156 // Push capture buffer into inputBuffer
5157 if ( captureBuffer.pushBuffer( ( char* ) streamBuffer,
5158 bufferFrameCount * stream_.nDeviceChannels[INPUT],
5159 stream_.deviceFormat[INPUT] ) )
5161 // Release capture buffer
5162 hr = captureClient->ReleaseBuffer( bufferFrameCount );
5163 if ( FAILED( hr ) ) {
5164 errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
5170 // Inform WASAPI that capture was unsuccessful
5171 hr = captureClient->ReleaseBuffer( 0 );
5172 if ( FAILED( hr ) ) {
5173 errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
5180 // Inform WASAPI that capture was unsuccessful
5181 hr = captureClient->ReleaseBuffer( 0 );
5182 if ( FAILED( hr ) ) {
5183 errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
5191 // 1. Get render buffer from stream
5192 // 2. Pull next buffer from outputBuffer
5193 // 3. If 2. was successful: Fill render buffer with next buffer
5194 // Release render buffer
5196 if ( renderAudioClient ) {
5197 // if the callback output buffer was not pushed to renderBuffer, wait for next render event
5198 if ( callbackPulled && !callbackPushed ) {
5199 WaitForSingleObject( renderEvent, INFINITE );
5202 // Get render buffer from stream
5203 hr = renderAudioClient->GetBufferSize( &bufferFrameCount );
5204 if ( FAILED( hr ) ) {
5205 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer size.";
5209 hr = renderAudioClient->GetCurrentPadding( &numFramesPadding );
5210 if ( FAILED( hr ) ) {
5211 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer padding.";
5215 bufferFrameCount -= numFramesPadding;
5217 if ( bufferFrameCount != 0 ) {
5218 hr = renderClient->GetBuffer( bufferFrameCount, &streamBuffer );
5219 if ( FAILED( hr ) ) {
5220 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer.";
5224 // Pull next buffer from outputBuffer
5225 // Fill render buffer with next buffer
5226 if ( renderBuffer.pullBuffer( ( char* ) streamBuffer,
5227 bufferFrameCount * stream_.nDeviceChannels[OUTPUT],
5228 stream_.deviceFormat[OUTPUT] ) )
5230 // Release render buffer
5231 hr = renderClient->ReleaseBuffer( bufferFrameCount, 0 );
5232 if ( FAILED( hr ) ) {
5233 errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
5239 // Inform WASAPI that render was unsuccessful
5240 hr = renderClient->ReleaseBuffer( 0, 0 );
5241 if ( FAILED( hr ) ) {
5242 errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
5249 // Inform WASAPI that render was unsuccessful
5250 hr = renderClient->ReleaseBuffer( 0, 0 );
5251 if ( FAILED( hr ) ) {
5252 errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
5258 // if the callback buffer was pushed renderBuffer reset callbackPulled flag
5259 if ( callbackPushed ) {
5260 callbackPulled = false;
5262 RtApi::tickStreamTime();
5269 CoTaskMemFree( captureFormat );
5270 CoTaskMemFree( renderFormat );
5272 free ( convBuffer );
5276 // update stream state
5277 stream_.state = STREAM_STOPPED;
5279 if ( errorText_.empty() )
5285 //******************** End of __WINDOWS_WASAPI__ *********************//
5289 #if defined(__WINDOWS_DS__) // Windows DirectSound API
5291 // Modified by Robin Davies, October 2005
5292 // - Improvements to DirectX pointer chasing.
5293 // - Bug fix for non-power-of-two Asio granularity used by Edirol PCR-A30.
5294 // - Auto-call CoInitialize for DSOUND and ASIO platforms.
5295 // Various revisions for RtAudio 4.0 by Gary Scavone, April 2007
5296 // Changed device query structure for RtAudio 4.0.7, January 2010
5298 #include <mmsystem.h>
5302 #include <algorithm>
5304 #if defined(__MINGW32__)
5305 // missing from latest mingw winapi
5306 #define WAVE_FORMAT_96M08 0x00010000 /* 96 kHz, Mono, 8-bit */
5307 #define WAVE_FORMAT_96S08 0x00020000 /* 96 kHz, Stereo, 8-bit */
5308 #define WAVE_FORMAT_96M16 0x00040000 /* 96 kHz, Mono, 16-bit */
5309 #define WAVE_FORMAT_96S16 0x00080000 /* 96 kHz, Stereo, 16-bit */
5312 #define MINIMUM_DEVICE_BUFFER_SIZE 32768
5314 #ifdef _MSC_VER // if Microsoft Visual C++
5315 #pragma comment( lib, "winmm.lib" ) // then, auto-link winmm.lib. Otherwise, it has to be added manually.
5318 static inline DWORD dsPointerBetween( DWORD pointer, DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize )
5320 if ( pointer > bufferSize ) pointer -= bufferSize;
5321 if ( laterPointer < earlierPointer ) laterPointer += bufferSize;
5322 if ( pointer < earlierPointer ) pointer += bufferSize;
5323 return pointer >= earlierPointer && pointer < laterPointer;
5326 // A structure to hold various information related to the DirectSound
5327 // API implementation.
5329 unsigned int drainCounter; // Tracks callback counts when draining
5330 bool internalDrain; // Indicates if stop is initiated from callback or not.
5334 UINT bufferPointer[2];
5335 DWORD dsBufferSize[2];
5336 DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by.
5340 :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; }
5343 // Declarations for utility functions, callbacks, and structures
5344 // specific to the DirectSound implementation.
5345 static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
5346 LPCTSTR description,
5350 static const char* getErrorString( int code );
5352 static unsigned __stdcall callbackHandler( void *ptr );
5361 : found(false) { validId[0] = false; validId[1] = false; }
5364 struct DsProbeData {
5366 std::vector<struct DsDevice>* dsDevices;
5369 RtApiDs :: RtApiDs()
5371 // Dsound will run both-threaded. If CoInitialize fails, then just
5372 // accept whatever the mainline chose for a threading model.
5373 coInitialized_ = false;
5374 HRESULT hr = CoInitialize( NULL );
5375 if ( !FAILED( hr ) ) coInitialized_ = true;
5378 RtApiDs :: ~RtApiDs()
5380 if ( stream_.state != STREAM_CLOSED ) closeStream();
5381 if ( coInitialized_ ) CoUninitialize(); // balanced call.
5384 // The DirectSound default output is always the first device.
5385 unsigned int RtApiDs :: getDefaultOutputDevice( void )
5390 // The DirectSound default input is always the first input device,
5391 // which is the first capture device enumerated.
5392 unsigned int RtApiDs :: getDefaultInputDevice( void )
5397 unsigned int RtApiDs :: getDeviceCount( void )
5399 // Set query flag for previously found devices to false, so that we
5400 // can check for any devices that have disappeared.
5401 for ( unsigned int i=0; i<dsDevices.size(); i++ )
5402 dsDevices[i].found = false;
5404 // Query DirectSound devices.
5405 struct DsProbeData probeInfo;
5406 probeInfo.isInput = false;
5407 probeInfo.dsDevices = &dsDevices;
5408 HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );
5409 if ( FAILED( result ) ) {
5410 errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!";
5411 errorText_ = errorStream_.str();
5412 error( RtAudioError::WARNING );
5415 // Query DirectSoundCapture devices.
5416 probeInfo.isInput = true;
5417 result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );
5418 if ( FAILED( result ) ) {
5419 errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!";
5420 errorText_ = errorStream_.str();
5421 error( RtAudioError::WARNING );
5424 // Clean out any devices that may have disappeared (code update submitted by Eli Zehngut).
5425 for ( unsigned int i=0; i<dsDevices.size(); ) {
5426 if ( dsDevices[i].found == false ) dsDevices.erase( dsDevices.begin() + i );
5430 return static_cast<unsigned int>(dsDevices.size());
5433 RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device )
5435 RtAudio::DeviceInfo info;
5436 info.probed = false;
5438 if ( dsDevices.size() == 0 ) {
5439 // Force a query of all devices
5441 if ( dsDevices.size() == 0 ) {
5442 errorText_ = "RtApiDs::getDeviceInfo: no devices found!";
5443 error( RtAudioError::INVALID_USE );
5448 if ( device >= dsDevices.size() ) {
5449 errorText_ = "RtApiDs::getDeviceInfo: device ID is invalid!";
5450 error( RtAudioError::INVALID_USE );
5455 if ( dsDevices[ device ].validId[0] == false ) goto probeInput;
5457 LPDIRECTSOUND output;
5459 result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );
5460 if ( FAILED( result ) ) {
5461 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!";
5462 errorText_ = errorStream_.str();
5463 error( RtAudioError::WARNING );
5467 outCaps.dwSize = sizeof( outCaps );
5468 result = output->GetCaps( &outCaps );
5469 if ( FAILED( result ) ) {
5471 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting capabilities!";
5472 errorText_ = errorStream_.str();
5473 error( RtAudioError::WARNING );
5477 // Get output channel information.
5478 info.outputChannels = ( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1;
5480 // Get sample rate information.
5481 info.sampleRates.clear();
5482 for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
5483 if ( SAMPLE_RATES[k] >= (unsigned int) outCaps.dwMinSecondarySampleRate &&
5484 SAMPLE_RATES[k] <= (unsigned int) outCaps.dwMaxSecondarySampleRate ) {
5485 info.sampleRates.push_back( SAMPLE_RATES[k] );
5487 if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
5488 info.preferredSampleRate = SAMPLE_RATES[k];
5492 // Get format information.
5493 if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT ) info.nativeFormats |= RTAUDIO_SINT16;
5494 if ( outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) info.nativeFormats |= RTAUDIO_SINT8;
5498 if ( getDefaultOutputDevice() == device )
5499 info.isDefaultOutput = true;
5501 if ( dsDevices[ device ].validId[1] == false ) {
5502 info.name = dsDevices[ device ].name;
5509 LPDIRECTSOUNDCAPTURE input;
5510 result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );
5511 if ( FAILED( result ) ) {
5512 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";
5513 errorText_ = errorStream_.str();
5514 error( RtAudioError::WARNING );
5519 inCaps.dwSize = sizeof( inCaps );
5520 result = input->GetCaps( &inCaps );
5521 if ( FAILED( result ) ) {
5523 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsDevices[ device ].name << ")!";
5524 errorText_ = errorStream_.str();
5525 error( RtAudioError::WARNING );
5529 // Get input channel information.
5530 info.inputChannels = inCaps.dwChannels;
5532 // Get sample rate and format information.
5533 std::vector<unsigned int> rates;
5534 if ( inCaps.dwChannels >= 2 ) {
5535 if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.nativeFormats |= RTAUDIO_SINT16;
5536 if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.nativeFormats |= RTAUDIO_SINT16;
5537 if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.nativeFormats |= RTAUDIO_SINT16;
5538 if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.nativeFormats |= RTAUDIO_SINT16;
5539 if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.nativeFormats |= RTAUDIO_SINT8;
5540 if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.nativeFormats |= RTAUDIO_SINT8;
5541 if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.nativeFormats |= RTAUDIO_SINT8;
5542 if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.nativeFormats |= RTAUDIO_SINT8;
5544 if ( info.nativeFormats & RTAUDIO_SINT16 ) {
5545 if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) rates.push_back( 11025 );
5546 if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) rates.push_back( 22050 );
5547 if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) rates.push_back( 44100 );
5548 if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) rates.push_back( 96000 );
5550 else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
5551 if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) rates.push_back( 11025 );
5552 if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) rates.push_back( 22050 );
5553 if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) rates.push_back( 44100 );
5554 if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) rates.push_back( 96000 );
5557 else if ( inCaps.dwChannels == 1 ) {
5558 if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.nativeFormats |= RTAUDIO_SINT16;
5559 if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.nativeFormats |= RTAUDIO_SINT16;
5560 if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.nativeFormats |= RTAUDIO_SINT16;
5561 if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.nativeFormats |= RTAUDIO_SINT16;
5562 if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.nativeFormats |= RTAUDIO_SINT8;
5563 if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.nativeFormats |= RTAUDIO_SINT8;
5564 if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.nativeFormats |= RTAUDIO_SINT8;
5565 if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.nativeFormats |= RTAUDIO_SINT8;
5567 if ( info.nativeFormats & RTAUDIO_SINT16 ) {
5568 if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) rates.push_back( 11025 );
5569 if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) rates.push_back( 22050 );
5570 if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) rates.push_back( 44100 );
5571 if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) rates.push_back( 96000 );
5573 else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
5574 if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) rates.push_back( 11025 );
5575 if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) rates.push_back( 22050 );
5576 if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) rates.push_back( 44100 );
5577 if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) rates.push_back( 96000 );
5580 else info.inputChannels = 0; // technically, this would be an error
5584 if ( info.inputChannels == 0 ) return info;
5586 // Copy the supported rates to the info structure but avoid duplication.
5588 for ( unsigned int i=0; i<rates.size(); i++ ) {
5590 for ( unsigned int j=0; j<info.sampleRates.size(); j++ ) {
5591 if ( rates[i] == info.sampleRates[j] ) {
5596 if ( found == false ) info.sampleRates.push_back( rates[i] );
5598 std::sort( info.sampleRates.begin(), info.sampleRates.end() );
5600 // If device opens for both playback and capture, we determine the channels.
5601 if ( info.outputChannels > 0 && info.inputChannels > 0 )
5602 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
5604 if ( device == 0 ) info.isDefaultInput = true;
5606 // Copy name and return.
5607 info.name = dsDevices[ device ].name;
5612 bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
5613 unsigned int firstChannel, unsigned int sampleRate,
5614 RtAudioFormat format, unsigned int *bufferSize,
5615 RtAudio::StreamOptions *options )
5617 if ( channels + firstChannel > 2 ) {
5618 errorText_ = "RtApiDs::probeDeviceOpen: DirectSound does not support more than 2 channels per device.";
5622 size_t nDevices = dsDevices.size();
5623 if ( nDevices == 0 ) {
5624 // This should not happen because a check is made before this function is called.
5625 errorText_ = "RtApiDs::probeDeviceOpen: no devices found!";
5629 if ( device >= nDevices ) {
5630 // This should not happen because a check is made before this function is called.
5631 errorText_ = "RtApiDs::probeDeviceOpen: device ID is invalid!";
5635 if ( mode == OUTPUT ) {
5636 if ( dsDevices[ device ].validId[0] == false ) {
5637 errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support output!";
5638 errorText_ = errorStream_.str();
5642 else { // mode == INPUT
5643 if ( dsDevices[ device ].validId[1] == false ) {
5644 errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support input!";
5645 errorText_ = errorStream_.str();
5650 // According to a note in PortAudio, using GetDesktopWindow()
5651 // instead of GetForegroundWindow() is supposed to avoid problems
5652 // that occur when the application's window is not the foreground
5653 // window. Also, if the application window closes before the
5654 // DirectSound buffer, DirectSound can crash. In the past, I had
5655 // problems when using GetDesktopWindow() but it seems fine now
5656 // (January 2010). I'll leave it commented here.
5657 // HWND hWnd = GetForegroundWindow();
5658 HWND hWnd = GetDesktopWindow();
5660 // Check the numberOfBuffers parameter and limit the lowest value to
5661 // two. This is a judgement call and a value of two is probably too
5662 // low for capture, but it should work for playback.
5664 if ( options ) nBuffers = options->numberOfBuffers;
5665 if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) nBuffers = 2;
5666 if ( nBuffers < 2 ) nBuffers = 3;
5668 // Check the lower range of the user-specified buffer size and set
5669 // (arbitrarily) to a lower bound of 32.
5670 if ( *bufferSize < 32 ) *bufferSize = 32;
5672 // Create the wave format structure. The data format setting will
5673 // be determined later.
5674 WAVEFORMATEX waveFormat;
5675 ZeroMemory( &waveFormat, sizeof(WAVEFORMATEX) );
5676 waveFormat.wFormatTag = WAVE_FORMAT_PCM;
5677 waveFormat.nChannels = channels + firstChannel;
5678 waveFormat.nSamplesPerSec = (unsigned long) sampleRate;
5680 // Determine the device buffer size. By default, we'll use the value
5681 // defined above (32K), but we will grow it to make allowances for
5682 // very large software buffer sizes.
5683 DWORD dsBufferSize = MINIMUM_DEVICE_BUFFER_SIZE;
5684 DWORD dsPointerLeadTime = 0;
5686 void *ohandle = 0, *bhandle = 0;
5688 if ( mode == OUTPUT ) {
5690 LPDIRECTSOUND output;
5691 result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );
5692 if ( FAILED( result ) ) {
5693 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!";
5694 errorText_ = errorStream_.str();
5699 outCaps.dwSize = sizeof( outCaps );
5700 result = output->GetCaps( &outCaps );
5701 if ( FAILED( result ) ) {
5703 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting capabilities (" << dsDevices[ device ].name << ")!";
5704 errorText_ = errorStream_.str();
5708 // Check channel information.
5709 if ( channels + firstChannel == 2 && !( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ) {
5710 errorStream_ << "RtApiDs::getDeviceInfo: the output device (" << dsDevices[ device ].name << ") does not support stereo playback.";
5711 errorText_ = errorStream_.str();
5715 // Check format information. Use 16-bit format unless not
5716 // supported or user requests 8-bit.
5717 if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT &&
5718 !( format == RTAUDIO_SINT8 && outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) ) {
5719 waveFormat.wBitsPerSample = 16;
5720 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
5723 waveFormat.wBitsPerSample = 8;
5724 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
5726 stream_.userFormat = format;
5728 // Update wave format structure and buffer information.
5729 waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
5730 waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
5731 dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
5733 // If the user wants an even bigger buffer, increase the device buffer size accordingly.
5734 while ( dsPointerLeadTime * 2U > dsBufferSize )
5737 // Set cooperative level to DSSCL_EXCLUSIVE ... sound stops when window focus changes.
5738 // result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE );
5739 // Set cooperative level to DSSCL_PRIORITY ... sound remains when window focus changes.
5740 result = output->SetCooperativeLevel( hWnd, DSSCL_PRIORITY );
5741 if ( FAILED( result ) ) {
5743 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting cooperative level (" << dsDevices[ device ].name << ")!";
5744 errorText_ = errorStream_.str();
5748 // Even though we will write to the secondary buffer, we need to
5749 // access the primary buffer to set the correct output format
5750 // (since the default is 8-bit, 22 kHz!). Setup the DS primary
5751 // buffer description.
5752 DSBUFFERDESC bufferDescription;
5753 ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
5754 bufferDescription.dwSize = sizeof( DSBUFFERDESC );
5755 bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER;
5757 // Obtain the primary buffer
5758 LPDIRECTSOUNDBUFFER buffer;
5759 result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
5760 if ( FAILED( result ) ) {
5762 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") accessing primary buffer (" << dsDevices[ device ].name << ")!";
5763 errorText_ = errorStream_.str();
5767 // Set the primary DS buffer sound format.
5768 result = buffer->SetFormat( &waveFormat );
5769 if ( FAILED( result ) ) {
5771 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting primary buffer format (" << dsDevices[ device ].name << ")!";
5772 errorText_ = errorStream_.str();
5776 // Setup the secondary DS buffer description.
5777 ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
5778 bufferDescription.dwSize = sizeof( DSBUFFERDESC );
5779 bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
5780 DSBCAPS_GLOBALFOCUS |
5781 DSBCAPS_GETCURRENTPOSITION2 |
5782 DSBCAPS_LOCHARDWARE ); // Force hardware mixing
5783 bufferDescription.dwBufferBytes = dsBufferSize;
5784 bufferDescription.lpwfxFormat = &waveFormat;
5786 // Try to create the secondary DS buffer. If that doesn't work,
5787 // try to use software mixing. Otherwise, there's a problem.
5788 result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
5789 if ( FAILED( result ) ) {
5790 bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
5791 DSBCAPS_GLOBALFOCUS |
5792 DSBCAPS_GETCURRENTPOSITION2 |
5793 DSBCAPS_LOCSOFTWARE ); // Force software mixing
5794 result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
5795 if ( FAILED( result ) ) {
5797 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating secondary buffer (" << dsDevices[ device ].name << ")!";
5798 errorText_ = errorStream_.str();
5803 // Get the buffer size ... might be different from what we specified.
5805 dsbcaps.dwSize = sizeof( DSBCAPS );
5806 result = buffer->GetCaps( &dsbcaps );
5807 if ( FAILED( result ) ) {
5810 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";
5811 errorText_ = errorStream_.str();
5815 dsBufferSize = dsbcaps.dwBufferBytes;
5817 // Lock the DS buffer
5820 result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );
5821 if ( FAILED( result ) ) {
5824 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking buffer (" << dsDevices[ device ].name << ")!";
5825 errorText_ = errorStream_.str();
5829 // Zero the DS buffer
5830 ZeroMemory( audioPtr, dataLen );
5832 // Unlock the DS buffer
5833 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
5834 if ( FAILED( result ) ) {
5837 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking buffer (" << dsDevices[ device ].name << ")!";
5838 errorText_ = errorStream_.str();
5842 ohandle = (void *) output;
5843 bhandle = (void *) buffer;
5846 if ( mode == INPUT ) {
5848 LPDIRECTSOUNDCAPTURE input;
5849 result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );
5850 if ( FAILED( result ) ) {
5851 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";
5852 errorText_ = errorStream_.str();
5857 inCaps.dwSize = sizeof( inCaps );
5858 result = input->GetCaps( &inCaps );
5859 if ( FAILED( result ) ) {
5861 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsDevices[ device ].name << ")!";
5862 errorText_ = errorStream_.str();
5866 // Check channel information.
5867 if ( inCaps.dwChannels < channels + firstChannel ) {
5868 errorText_ = "RtApiDs::getDeviceInfo: the input device does not support requested input channels.";
5872 // Check format information. Use 16-bit format unless user
5874 DWORD deviceFormats;
5875 if ( channels + firstChannel == 2 ) {
5876 deviceFormats = WAVE_FORMAT_1S08 | WAVE_FORMAT_2S08 | WAVE_FORMAT_4S08 | WAVE_FORMAT_96S08;
5877 if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
5878 waveFormat.wBitsPerSample = 8;
5879 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
5881 else { // assume 16-bit is supported
5882 waveFormat.wBitsPerSample = 16;
5883 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
5886 else { // channel == 1
5887 deviceFormats = WAVE_FORMAT_1M08 | WAVE_FORMAT_2M08 | WAVE_FORMAT_4M08 | WAVE_FORMAT_96M08;
5888 if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
5889 waveFormat.wBitsPerSample = 8;
5890 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
5892 else { // assume 16-bit is supported
5893 waveFormat.wBitsPerSample = 16;
5894 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
5897 stream_.userFormat = format;
5899 // Update wave format structure and buffer information.
5900 waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
5901 waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
5902 dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
5904 // If the user wants an even bigger buffer, increase the device buffer size accordingly.
5905 while ( dsPointerLeadTime * 2U > dsBufferSize )
5908 // Setup the secondary DS buffer description.
5909 DSCBUFFERDESC bufferDescription;
5910 ZeroMemory( &bufferDescription, sizeof( DSCBUFFERDESC ) );
5911 bufferDescription.dwSize = sizeof( DSCBUFFERDESC );
5912 bufferDescription.dwFlags = 0;
5913 bufferDescription.dwReserved = 0;
5914 bufferDescription.dwBufferBytes = dsBufferSize;
5915 bufferDescription.lpwfxFormat = &waveFormat;
5917 // Create the capture buffer.
5918 LPDIRECTSOUNDCAPTUREBUFFER buffer;
5919 result = input->CreateCaptureBuffer( &bufferDescription, &buffer, NULL );
5920 if ( FAILED( result ) ) {
5922 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating input buffer (" << dsDevices[ device ].name << ")!";
5923 errorText_ = errorStream_.str();
5927 // Get the buffer size ... might be different from what we specified.
5929 dscbcaps.dwSize = sizeof( DSCBCAPS );
5930 result = buffer->GetCaps( &dscbcaps );
5931 if ( FAILED( result ) ) {
5934 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";
5935 errorText_ = errorStream_.str();
5939 dsBufferSize = dscbcaps.dwBufferBytes;
5941 // NOTE: We could have a problem here if this is a duplex stream
5942 // and the play and capture hardware buffer sizes are different
5943 // (I'm actually not sure if that is a problem or not).
5944 // Currently, we are not verifying that.
5946 // Lock the capture buffer
5949 result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );
5950 if ( FAILED( result ) ) {
5953 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking input buffer (" << dsDevices[ device ].name << ")!";
5954 errorText_ = errorStream_.str();
5959 ZeroMemory( audioPtr, dataLen );
5961 // Unlock the buffer
5962 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
5963 if ( FAILED( result ) ) {
5966 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking input buffer (" << dsDevices[ device ].name << ")!";
5967 errorText_ = errorStream_.str();
5971 ohandle = (void *) input;
5972 bhandle = (void *) buffer;
5975 // Set various stream parameters
5976 DsHandle *handle = 0;
5977 stream_.nDeviceChannels[mode] = channels + firstChannel;
5978 stream_.nUserChannels[mode] = channels;
5979 stream_.bufferSize = *bufferSize;
5980 stream_.channelOffset[mode] = firstChannel;
5981 stream_.deviceInterleaved[mode] = true;
5982 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
5983 else stream_.userInterleaved = true;
5985 // Set flag for buffer conversion
5986 stream_.doConvertBuffer[mode] = false;
5987 if (stream_.nUserChannels[mode] != stream_.nDeviceChannels[mode])
5988 stream_.doConvertBuffer[mode] = true;
5989 if (stream_.userFormat != stream_.deviceFormat[mode])
5990 stream_.doConvertBuffer[mode] = true;
5991 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
5992 stream_.nUserChannels[mode] > 1 )
5993 stream_.doConvertBuffer[mode] = true;
5995 // Allocate necessary internal buffers
5996 long bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
5997 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
5998 if ( stream_.userBuffer[mode] == NULL ) {
5999 errorText_ = "RtApiDs::probeDeviceOpen: error allocating user buffer memory.";
6003 if ( stream_.doConvertBuffer[mode] ) {
6005 bool makeBuffer = true;
6006 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
6007 if ( mode == INPUT ) {
6008 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
6009 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
6010 if ( bufferBytes <= (long) bytesOut ) makeBuffer = false;
6015 bufferBytes *= *bufferSize;
6016 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
6017 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
6018 if ( stream_.deviceBuffer == NULL ) {
6019 errorText_ = "RtApiDs::probeDeviceOpen: error allocating device buffer memory.";
6025 // Allocate our DsHandle structures for the stream.
6026 if ( stream_.apiHandle == 0 ) {
6028 handle = new DsHandle;
6030 catch ( std::bad_alloc& ) {
6031 errorText_ = "RtApiDs::probeDeviceOpen: error allocating AsioHandle memory.";
6035 // Create a manual-reset event.
6036 handle->condition = CreateEvent( NULL, // no security
6037 TRUE, // manual-reset
6038 FALSE, // non-signaled initially
6040 stream_.apiHandle = (void *) handle;
6043 handle = (DsHandle *) stream_.apiHandle;
6044 handle->id[mode] = ohandle;
6045 handle->buffer[mode] = bhandle;
6046 handle->dsBufferSize[mode] = dsBufferSize;
6047 handle->dsPointerLeadTime[mode] = dsPointerLeadTime;
6049 stream_.device[mode] = device;
6050 stream_.state = STREAM_STOPPED;
6051 if ( stream_.mode == OUTPUT && mode == INPUT )
6052 // We had already set up an output stream.
6053 stream_.mode = DUPLEX;
6055 stream_.mode = mode;
6056 stream_.nBuffers = nBuffers;
6057 stream_.sampleRate = sampleRate;
6059 // Setup the buffer conversion information structure.
6060 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
6062 // Setup the callback thread.
6063 if ( stream_.callbackInfo.isRunning == false ) {
6065 stream_.callbackInfo.isRunning = true;
6066 stream_.callbackInfo.object = (void *) this;
6067 stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &callbackHandler,
6068 &stream_.callbackInfo, 0, &threadId );
6069 if ( stream_.callbackInfo.thread == 0 ) {
6070 errorText_ = "RtApiDs::probeDeviceOpen: error creating callback thread!";
6074 // Boost DS thread priority
6075 SetThreadPriority( (HANDLE) stream_.callbackInfo.thread, THREAD_PRIORITY_HIGHEST );
6081 if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
6082 LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
6083 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6084 if ( buffer ) buffer->Release();
6087 if ( handle->buffer[1] ) {
6088 LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
6089 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6090 if ( buffer ) buffer->Release();
6093 CloseHandle( handle->condition );
6095 stream_.apiHandle = 0;
6098 for ( int i=0; i<2; i++ ) {
6099 if ( stream_.userBuffer[i] ) {
6100 free( stream_.userBuffer[i] );
6101 stream_.userBuffer[i] = 0;
6105 if ( stream_.deviceBuffer ) {
6106 free( stream_.deviceBuffer );
6107 stream_.deviceBuffer = 0;
6110 stream_.state = STREAM_CLOSED;
6114 void RtApiDs :: closeStream()
6116 if ( stream_.state == STREAM_CLOSED ) {
6117 errorText_ = "RtApiDs::closeStream(): no open stream to close!";
6118 error( RtAudioError::WARNING );
6122 // Stop the callback thread.
6123 stream_.callbackInfo.isRunning = false;
6124 WaitForSingleObject( (HANDLE) stream_.callbackInfo.thread, INFINITE );
6125 CloseHandle( (HANDLE) stream_.callbackInfo.thread );
6127 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6129 if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
6130 LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
6131 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6138 if ( handle->buffer[1] ) {
6139 LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
6140 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6147 CloseHandle( handle->condition );
6149 stream_.apiHandle = 0;
6152 for ( int i=0; i<2; i++ ) {
6153 if ( stream_.userBuffer[i] ) {
6154 free( stream_.userBuffer[i] );
6155 stream_.userBuffer[i] = 0;
6159 if ( stream_.deviceBuffer ) {
6160 free( stream_.deviceBuffer );
6161 stream_.deviceBuffer = 0;
6164 stream_.mode = UNINITIALIZED;
6165 stream_.state = STREAM_CLOSED;
6168 void RtApiDs :: startStream()
6171 if ( stream_.state == STREAM_RUNNING ) {
6172 errorText_ = "RtApiDs::startStream(): the stream is already running!";
6173 error( RtAudioError::WARNING );
6177 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6179 // Increase scheduler frequency on lesser windows (a side-effect of
6180 // increasing timer accuracy). On greater windows (Win2K or later),
6181 // this is already in effect.
6182 timeBeginPeriod( 1 );
6184 buffersRolling = false;
6185 duplexPrerollBytes = 0;
6187 if ( stream_.mode == DUPLEX ) {
6188 // 0.5 seconds of silence in DUPLEX mode while the devices spin up and synchronize.
6189 duplexPrerollBytes = (int) ( 0.5 * stream_.sampleRate * formatBytes( stream_.deviceFormat[1] ) * stream_.nDeviceChannels[1] );
6193 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
6195 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6196 result = buffer->Play( 0, 0, DSBPLAY_LOOPING );
6197 if ( FAILED( result ) ) {
6198 errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting output buffer!";
6199 errorText_ = errorStream_.str();
6204 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
6206 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6207 result = buffer->Start( DSCBSTART_LOOPING );
6208 if ( FAILED( result ) ) {
6209 errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting input buffer!";
6210 errorText_ = errorStream_.str();
6215 handle->drainCounter = 0;
6216 handle->internalDrain = false;
6217 ResetEvent( handle->condition );
6218 stream_.state = STREAM_RUNNING;
6221 if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );
6224 void RtApiDs :: stopStream()
6227 if ( stream_.state == STREAM_STOPPED ) {
6228 errorText_ = "RtApiDs::stopStream(): the stream is already stopped!";
6229 error( RtAudioError::WARNING );
6236 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6237 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
6238 if ( handle->drainCounter == 0 ) {
6239 handle->drainCounter = 2;
6240 WaitForSingleObject( handle->condition, INFINITE ); // block until signaled
6243 stream_.state = STREAM_STOPPED;
6245 MUTEX_LOCK( &stream_.mutex );
6247 // Stop the buffer and clear memory
6248 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6249 result = buffer->Stop();
6250 if ( FAILED( result ) ) {
6251 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping output buffer!";
6252 errorText_ = errorStream_.str();
6256 // Lock the buffer and clear it so that if we start to play again,
6257 // we won't have old data playing.
6258 result = buffer->Lock( 0, handle->dsBufferSize[0], &audioPtr, &dataLen, NULL, NULL, 0 );
6259 if ( FAILED( result ) ) {
6260 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking output buffer!";
6261 errorText_ = errorStream_.str();
6265 // Zero the DS buffer
6266 ZeroMemory( audioPtr, dataLen );
6268 // Unlock the DS buffer
6269 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
6270 if ( FAILED( result ) ) {
6271 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking output buffer!";
6272 errorText_ = errorStream_.str();
6276 // If we start playing again, we must begin at beginning of buffer.
6277 handle->bufferPointer[0] = 0;
6280 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
6281 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6285 stream_.state = STREAM_STOPPED;
6287 if ( stream_.mode != DUPLEX )
6288 MUTEX_LOCK( &stream_.mutex );
6290 result = buffer->Stop();
6291 if ( FAILED( result ) ) {
6292 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping input buffer!";
6293 errorText_ = errorStream_.str();
6297 // Lock the buffer and clear it so that if we start to play again,
6298 // we won't have old data playing.
6299 result = buffer->Lock( 0, handle->dsBufferSize[1], &audioPtr, &dataLen, NULL, NULL, 0 );
6300 if ( FAILED( result ) ) {
6301 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking input buffer!";
6302 errorText_ = errorStream_.str();
6306 // Zero the DS buffer
6307 ZeroMemory( audioPtr, dataLen );
6309 // Unlock the DS buffer
6310 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
6311 if ( FAILED( result ) ) {
6312 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking input buffer!";
6313 errorText_ = errorStream_.str();
6317 // If we start recording again, we must begin at beginning of buffer.
6318 handle->bufferPointer[1] = 0;
6322 timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows.
6323 MUTEX_UNLOCK( &stream_.mutex );
6325 if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );
6328 void RtApiDs :: abortStream()
6331 if ( stream_.state == STREAM_STOPPED ) {
6332 errorText_ = "RtApiDs::abortStream(): the stream is already stopped!";
6333 error( RtAudioError::WARNING );
6337 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6338 handle->drainCounter = 2;
6343 void RtApiDs :: callbackEvent()
6345 if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) {
6346 Sleep( 50 ); // sleep 50 milliseconds
6350 if ( stream_.state == STREAM_CLOSED ) {
6351 errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!";
6352 error( RtAudioError::WARNING );
6356 CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
6357 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6359 // Check if we were draining the stream and signal is finished.
6360 if ( handle->drainCounter > stream_.nBuffers + 2 ) {
6362 stream_.state = STREAM_STOPPING;
6363 if ( handle->internalDrain == false )
6364 SetEvent( handle->condition );
6370 // Invoke user callback to get fresh output data UNLESS we are
6372 if ( handle->drainCounter == 0 ) {
6373 RtAudioCallback callback = (RtAudioCallback) info->callback;
6374 double streamTime = getStreamTime();
6375 RtAudioStreamStatus status = 0;
6376 if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
6377 status |= RTAUDIO_OUTPUT_UNDERFLOW;
6378 handle->xrun[0] = false;
6380 if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
6381 status |= RTAUDIO_INPUT_OVERFLOW;
6382 handle->xrun[1] = false;
6384 int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
6385 stream_.bufferSize, streamTime, status, info->userData );
6386 if ( cbReturnValue == 2 ) {
6387 stream_.state = STREAM_STOPPING;
6388 handle->drainCounter = 2;
6392 else if ( cbReturnValue == 1 ) {
6393 handle->drainCounter = 1;
6394 handle->internalDrain = true;
6399 DWORD currentWritePointer, safeWritePointer;
6400 DWORD currentReadPointer, safeReadPointer;
6401 UINT nextWritePointer;
6403 LPVOID buffer1 = NULL;
6404 LPVOID buffer2 = NULL;
6405 DWORD bufferSize1 = 0;
6406 DWORD bufferSize2 = 0;
6411 MUTEX_LOCK( &stream_.mutex );
6412 if ( stream_.state == STREAM_STOPPED ) {
6413 MUTEX_UNLOCK( &stream_.mutex );
6417 if ( buffersRolling == false ) {
6418 if ( stream_.mode == DUPLEX ) {
6419 //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
6421 // It takes a while for the devices to get rolling. As a result,
6422 // there's no guarantee that the capture and write device pointers
6423 // will move in lockstep. Wait here for both devices to start
6424 // rolling, and then set our buffer pointers accordingly.
6425 // e.g. Crystal Drivers: the capture buffer starts up 5700 to 9600
6426 // bytes later than the write buffer.
6428 // Stub: a serious risk of having a pre-emptive scheduling round
6429 // take place between the two GetCurrentPosition calls... but I'm
6430 // really not sure how to solve the problem. Temporarily boost to
6431 // Realtime priority, maybe; but I'm not sure what priority the
6432 // DirectSound service threads run at. We *should* be roughly
6433 // within a ms or so of correct.
6435 LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6436 LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6438 DWORD startSafeWritePointer, startSafeReadPointer;
6440 result = dsWriteBuffer->GetCurrentPosition( NULL, &startSafeWritePointer );
6441 if ( FAILED( result ) ) {
6442 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
6443 errorText_ = errorStream_.str();
6444 MUTEX_UNLOCK( &stream_.mutex );
6445 error( RtAudioError::SYSTEM_ERROR );
6448 result = dsCaptureBuffer->GetCurrentPosition( NULL, &startSafeReadPointer );
6449 if ( FAILED( result ) ) {
6450 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
6451 errorText_ = errorStream_.str();
6452 MUTEX_UNLOCK( &stream_.mutex );
6453 error( RtAudioError::SYSTEM_ERROR );
6457 result = dsWriteBuffer->GetCurrentPosition( NULL, &safeWritePointer );
6458 if ( FAILED( result ) ) {
6459 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
6460 errorText_ = errorStream_.str();
6461 MUTEX_UNLOCK( &stream_.mutex );
6462 error( RtAudioError::SYSTEM_ERROR );
6465 result = dsCaptureBuffer->GetCurrentPosition( NULL, &safeReadPointer );
6466 if ( FAILED( result ) ) {
6467 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
6468 errorText_ = errorStream_.str();
6469 MUTEX_UNLOCK( &stream_.mutex );
6470 error( RtAudioError::SYSTEM_ERROR );
6473 if ( safeWritePointer != startSafeWritePointer && safeReadPointer != startSafeReadPointer ) break;
6477 //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
6479 handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];
6480 if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];
6481 handle->bufferPointer[1] = safeReadPointer;
6483 else if ( stream_.mode == OUTPUT ) {
6485 // Set the proper nextWritePosition after initial startup.
6486 LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6487 result = dsWriteBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer );
6488 if ( FAILED( result ) ) {
6489 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
6490 errorText_ = errorStream_.str();
6491 MUTEX_UNLOCK( &stream_.mutex );
6492 error( RtAudioError::SYSTEM_ERROR );
6495 handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];
6496 if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];
6499 buffersRolling = true;
6502 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
6504 LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6506 if ( handle->drainCounter > 1 ) { // write zeros to the output stream
6507 bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
6508 bufferBytes *= formatBytes( stream_.userFormat );
6509 memset( stream_.userBuffer[0], 0, bufferBytes );
6512 // Setup parameters and do buffer conversion if necessary.
6513 if ( stream_.doConvertBuffer[0] ) {
6514 buffer = stream_.deviceBuffer;
6515 convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
6516 bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[0];
6517 bufferBytes *= formatBytes( stream_.deviceFormat[0] );
6520 buffer = stream_.userBuffer[0];
6521 bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
6522 bufferBytes *= formatBytes( stream_.userFormat );
6525 // No byte swapping necessary in DirectSound implementation.
6527 // Ahhh ... windoze. 16-bit data is signed but 8-bit data is
6528 // unsigned. So, we need to convert our signed 8-bit data here to
6530 if ( stream_.deviceFormat[0] == RTAUDIO_SINT8 )
6531 for ( int i=0; i<bufferBytes; i++ ) buffer[i] = (unsigned char) ( buffer[i] + 128 );
6533 DWORD dsBufferSize = handle->dsBufferSize[0];
6534 nextWritePointer = handle->bufferPointer[0];
6536 DWORD endWrite, leadPointer;
6538 // Find out where the read and "safe write" pointers are.
6539 result = dsBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer );
6540 if ( FAILED( result ) ) {
6541 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
6542 errorText_ = errorStream_.str();
6543 MUTEX_UNLOCK( &stream_.mutex );
6544 error( RtAudioError::SYSTEM_ERROR );
6548 // We will copy our output buffer into the region between
6549 // safeWritePointer and leadPointer. If leadPointer is not
6550 // beyond the next endWrite position, wait until it is.
6551 leadPointer = safeWritePointer + handle->dsPointerLeadTime[0];
6552 //std::cout << "safeWritePointer = " << safeWritePointer << ", leadPointer = " << leadPointer << ", nextWritePointer = " << nextWritePointer << std::endl;
6553 if ( leadPointer > dsBufferSize ) leadPointer -= dsBufferSize;
6554 if ( leadPointer < nextWritePointer ) leadPointer += dsBufferSize; // unwrap offset
6555 endWrite = nextWritePointer + bufferBytes;
6557 // Check whether the entire write region is behind the play pointer.
6558 if ( leadPointer >= endWrite ) break;
6560 // If we are here, then we must wait until the leadPointer advances
6561 // beyond the end of our next write region. We use the
6562 // Sleep() function to suspend operation until that happens.
6563 double millis = ( endWrite - leadPointer ) * 1000.0;
6564 millis /= ( formatBytes( stream_.deviceFormat[0]) * stream_.nDeviceChannels[0] * stream_.sampleRate);
6565 if ( millis < 1.0 ) millis = 1.0;
6566 Sleep( (DWORD) millis );
6569 if ( dsPointerBetween( nextWritePointer, safeWritePointer, currentWritePointer, dsBufferSize )
6570 || dsPointerBetween( endWrite, safeWritePointer, currentWritePointer, dsBufferSize ) ) {
6571 // We've strayed into the forbidden zone ... resync the read pointer.
6572 handle->xrun[0] = true;
6573 nextWritePointer = safeWritePointer + handle->dsPointerLeadTime[0] - bufferBytes;
6574 if ( nextWritePointer >= dsBufferSize ) nextWritePointer -= dsBufferSize;
6575 handle->bufferPointer[0] = nextWritePointer;
6576 endWrite = nextWritePointer + bufferBytes;
6579 // Lock free space in the buffer
6580 result = dsBuffer->Lock( nextWritePointer, bufferBytes, &buffer1,
6581 &bufferSize1, &buffer2, &bufferSize2, 0 );
6582 if ( FAILED( result ) ) {
6583 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking buffer during playback!";
6584 errorText_ = errorStream_.str();
6585 MUTEX_UNLOCK( &stream_.mutex );
6586 error( RtAudioError::SYSTEM_ERROR );
6590 // Copy our buffer into the DS buffer
6591 CopyMemory( buffer1, buffer, bufferSize1 );
6592 if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 );
6594 // Update our buffer offset and unlock sound buffer
6595 dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
6596 if ( FAILED( result ) ) {
6597 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!";
6598 errorText_ = errorStream_.str();
6599 MUTEX_UNLOCK( &stream_.mutex );
6600 error( RtAudioError::SYSTEM_ERROR );
6603 nextWritePointer = ( nextWritePointer + bufferSize1 + bufferSize2 ) % dsBufferSize;
6604 handle->bufferPointer[0] = nextWritePointer;
6607 // Don't bother draining input
6608 if ( handle->drainCounter ) {
6609 handle->drainCounter++;
6613 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
6615 // Setup parameters.
6616 if ( stream_.doConvertBuffer[1] ) {
6617 buffer = stream_.deviceBuffer;
6618 bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[1];
6619 bufferBytes *= formatBytes( stream_.deviceFormat[1] );
6622 buffer = stream_.userBuffer[1];
6623 bufferBytes = stream_.bufferSize * stream_.nUserChannels[1];
6624 bufferBytes *= formatBytes( stream_.userFormat );
6627 LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6628 long nextReadPointer = handle->bufferPointer[1];
6629 DWORD dsBufferSize = handle->dsBufferSize[1];
6631 // Find out where the write and "safe read" pointers are.
6632 result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer );
6633 if ( FAILED( result ) ) {
6634 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
6635 errorText_ = errorStream_.str();
6636 MUTEX_UNLOCK( &stream_.mutex );
6637 error( RtAudioError::SYSTEM_ERROR );
6641 if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset
6642 DWORD endRead = nextReadPointer + bufferBytes;
6644 // Handling depends on whether we are INPUT or DUPLEX.
6645 // If we're in INPUT mode then waiting is a good thing. If we're in DUPLEX mode,
6646 // then a wait here will drag the write pointers into the forbidden zone.
6648 // In DUPLEX mode, rather than wait, we will back off the read pointer until
6649 // it's in a safe position. This causes dropouts, but it seems to be the only
6650 // practical way to sync up the read and write pointers reliably, given the
6651 // the very complex relationship between phase and increment of the read and write
6654 // In order to minimize audible dropouts in DUPLEX mode, we will
6655 // provide a pre-roll period of 0.5 seconds in which we return
6656 // zeros from the read buffer while the pointers sync up.
6658 if ( stream_.mode == DUPLEX ) {
6659 if ( safeReadPointer < endRead ) {
6660 if ( duplexPrerollBytes <= 0 ) {
6661 // Pre-roll time over. Be more agressive.
6662 int adjustment = endRead-safeReadPointer;
6664 handle->xrun[1] = true;
6666 // - large adjustments: we've probably run out of CPU cycles, so just resync exactly,
6667 // and perform fine adjustments later.
6668 // - small adjustments: back off by twice as much.
6669 if ( adjustment >= 2*bufferBytes )
6670 nextReadPointer = safeReadPointer-2*bufferBytes;
6672 nextReadPointer = safeReadPointer-bufferBytes-adjustment;
6674 if ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;
6678 // In pre=roll time. Just do it.
6679 nextReadPointer = safeReadPointer - bufferBytes;
6680 while ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;
6682 endRead = nextReadPointer + bufferBytes;
6685 else { // mode == INPUT
6686 while ( safeReadPointer < endRead && stream_.callbackInfo.isRunning ) {
6687 // See comments for playback.
6688 double millis = (endRead - safeReadPointer) * 1000.0;
6689 millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate);
6690 if ( millis < 1.0 ) millis = 1.0;
6691 Sleep( (DWORD) millis );
6693 // Wake up and find out where we are now.
6694 result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer );
6695 if ( FAILED( result ) ) {
6696 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
6697 errorText_ = errorStream_.str();
6698 MUTEX_UNLOCK( &stream_.mutex );
6699 error( RtAudioError::SYSTEM_ERROR );
6703 if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset
6707 // Lock free space in the buffer
6708 result = dsBuffer->Lock( nextReadPointer, bufferBytes, &buffer1,
6709 &bufferSize1, &buffer2, &bufferSize2, 0 );
6710 if ( FAILED( result ) ) {
6711 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking capture buffer!";
6712 errorText_ = errorStream_.str();
6713 MUTEX_UNLOCK( &stream_.mutex );
6714 error( RtAudioError::SYSTEM_ERROR );
6718 if ( duplexPrerollBytes <= 0 ) {
6719 // Copy our buffer into the DS buffer
6720 CopyMemory( buffer, buffer1, bufferSize1 );
6721 if ( buffer2 != NULL ) CopyMemory( buffer+bufferSize1, buffer2, bufferSize2 );
6724 memset( buffer, 0, bufferSize1 );
6725 if ( buffer2 != NULL ) memset( buffer + bufferSize1, 0, bufferSize2 );
6726 duplexPrerollBytes -= bufferSize1 + bufferSize2;
6729 // Update our buffer offset and unlock sound buffer
6730 nextReadPointer = ( nextReadPointer + bufferSize1 + bufferSize2 ) % dsBufferSize;
6731 dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
6732 if ( FAILED( result ) ) {
6733 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking capture buffer!";
6734 errorText_ = errorStream_.str();
6735 MUTEX_UNLOCK( &stream_.mutex );
6736 error( RtAudioError::SYSTEM_ERROR );
6739 handle->bufferPointer[1] = nextReadPointer;
6741 // No byte swapping necessary in DirectSound implementation.
6743 // If necessary, convert 8-bit data from unsigned to signed.
6744 if ( stream_.deviceFormat[1] == RTAUDIO_SINT8 )
6745 for ( int j=0; j<bufferBytes; j++ ) buffer[j] = (signed char) ( buffer[j] - 128 );
6747 // Do buffer conversion if necessary.
6748 if ( stream_.doConvertBuffer[1] )
6749 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
6753 MUTEX_UNLOCK( &stream_.mutex );
6754 RtApi::tickStreamTime();
6757 // Definitions for utility functions and callbacks
6758 // specific to the DirectSound implementation.
6760 static unsigned __stdcall callbackHandler( void *ptr )
6762 CallbackInfo *info = (CallbackInfo *) ptr;
6763 RtApiDs *object = (RtApiDs *) info->object;
6764 bool* isRunning = &info->isRunning;
6766 while ( *isRunning == true ) {
6767 object->callbackEvent();
6774 static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
6775 LPCTSTR description,
6779 struct DsProbeData& probeInfo = *(struct DsProbeData*) lpContext;
6780 std::vector<struct DsDevice>& dsDevices = *probeInfo.dsDevices;
6783 bool validDevice = false;
6784 if ( probeInfo.isInput == true ) {
6786 LPDIRECTSOUNDCAPTURE object;
6788 hr = DirectSoundCaptureCreate( lpguid, &object, NULL );
6789 if ( hr != DS_OK ) return TRUE;
6791 caps.dwSize = sizeof(caps);
6792 hr = object->GetCaps( &caps );
6793 if ( hr == DS_OK ) {
6794 if ( caps.dwChannels > 0 && caps.dwFormats > 0 )
6801 LPDIRECTSOUND object;
6802 hr = DirectSoundCreate( lpguid, &object, NULL );
6803 if ( hr != DS_OK ) return TRUE;
6805 caps.dwSize = sizeof(caps);
6806 hr = object->GetCaps( &caps );
6807 if ( hr == DS_OK ) {
6808 if ( caps.dwFlags & DSCAPS_PRIMARYMONO || caps.dwFlags & DSCAPS_PRIMARYSTEREO )
6814 // If good device, then save its name and guid.
6815 std::string name = convertCharPointerToStdString( description );
6816 //if ( name == "Primary Sound Driver" || name == "Primary Sound Capture Driver" )
6817 if ( lpguid == NULL )
6818 name = "Default Device";
6819 if ( validDevice ) {
6820 for ( unsigned int i=0; i<dsDevices.size(); i++ ) {
6821 if ( dsDevices[i].name == name ) {
6822 dsDevices[i].found = true;
6823 if ( probeInfo.isInput ) {
6824 dsDevices[i].id[1] = lpguid;
6825 dsDevices[i].validId[1] = true;
6828 dsDevices[i].id[0] = lpguid;
6829 dsDevices[i].validId[0] = true;
6837 device.found = true;
6838 if ( probeInfo.isInput ) {
6839 device.id[1] = lpguid;
6840 device.validId[1] = true;
6843 device.id[0] = lpguid;
6844 device.validId[0] = true;
6846 dsDevices.push_back( device );
6852 static const char* getErrorString( int code )
6856 case DSERR_ALLOCATED:
6857 return "Already allocated";
6859 case DSERR_CONTROLUNAVAIL:
6860 return "Control unavailable";
6862 case DSERR_INVALIDPARAM:
6863 return "Invalid parameter";
6865 case DSERR_INVALIDCALL:
6866 return "Invalid call";
6869 return "Generic error";
6871 case DSERR_PRIOLEVELNEEDED:
6872 return "Priority level needed";
6874 case DSERR_OUTOFMEMORY:
6875 return "Out of memory";
6877 case DSERR_BADFORMAT:
6878 return "The sample rate or the channel format is not supported";
6880 case DSERR_UNSUPPORTED:
6881 return "Not supported";
6883 case DSERR_NODRIVER:
6886 case DSERR_ALREADYINITIALIZED:
6887 return "Already initialized";
6889 case DSERR_NOAGGREGATION:
6890 return "No aggregation";
6892 case DSERR_BUFFERLOST:
6893 return "Buffer lost";
6895 case DSERR_OTHERAPPHASPRIO:
6896 return "Another application already has priority";
6898 case DSERR_UNINITIALIZED:
6899 return "Uninitialized";
6902 return "DirectSound unknown error";
6905 //******************** End of __WINDOWS_DS__ *********************//
6909 #if defined(__LINUX_ALSA__)
6911 #include <alsa/asoundlib.h>
6914 // A structure to hold various information related to the ALSA API
6917 snd_pcm_t *handles[2];
6920 pthread_cond_t runnable_cv;
6924 :synchronized(false), runnable(false) { xrun[0] = false; xrun[1] = false; }
6927 static void *alsaCallbackHandler( void * ptr );
6929 RtApiAlsa :: RtApiAlsa()
6931 // Nothing to do here.
6934 RtApiAlsa :: ~RtApiAlsa()
6936 if ( stream_.state != STREAM_CLOSED ) closeStream();
6939 unsigned int RtApiAlsa :: getDeviceCount( void )
6941 unsigned nDevices = 0;
6942 int result, subdevice, card;
6946 // Count cards and devices
6948 snd_card_next( &card );
6949 while ( card >= 0 ) {
6950 sprintf( name, "hw:%d", card );
6951 result = snd_ctl_open( &handle, name, 0 );
6953 errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << ".";
6954 errorText_ = errorStream_.str();
6955 error( RtAudioError::WARNING );
6960 result = snd_ctl_pcm_next_device( handle, &subdevice );
6962 errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
6963 errorText_ = errorStream_.str();
6964 error( RtAudioError::WARNING );
6967 if ( subdevice < 0 )
6972 snd_ctl_close( handle );
6973 snd_card_next( &card );
6976 result = snd_ctl_open( &handle, "default", 0 );
6979 snd_ctl_close( handle );
6985 RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device )
6987 RtAudio::DeviceInfo info;
6988 info.probed = false;
6990 unsigned nDevices = 0;
6991 int result, subdevice, card;
6995 // Count cards and devices
6998 snd_card_next( &card );
6999 while ( card >= 0 ) {
7000 sprintf( name, "hw:%d", card );
7001 result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
7003 errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << ".";
7004 errorText_ = errorStream_.str();
7005 error( RtAudioError::WARNING );
7010 result = snd_ctl_pcm_next_device( chandle, &subdevice );
7012 errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
7013 errorText_ = errorStream_.str();
7014 error( RtAudioError::WARNING );
7017 if ( subdevice < 0 ) break;
7018 if ( nDevices == device ) {
7019 sprintf( name, "hw:%d,%d", card, subdevice );
7025 snd_ctl_close( chandle );
7026 snd_card_next( &card );
7029 result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
7030 if ( result == 0 ) {
7031 if ( nDevices == device ) {
7032 strcpy( name, "default" );
7038 if ( nDevices == 0 ) {
7039 errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!";
7040 error( RtAudioError::INVALID_USE );
7044 if ( device >= nDevices ) {
7045 errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!";
7046 error( RtAudioError::INVALID_USE );
7052 // If a stream is already open, we cannot probe the stream devices.
7053 // Thus, use the saved results.
7054 if ( stream_.state != STREAM_CLOSED &&
7055 ( stream_.device[0] == device || stream_.device[1] == device ) ) {
7056 snd_ctl_close( chandle );
7057 if ( device >= devices_.size() ) {
7058 errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened.";
7059 error( RtAudioError::WARNING );
7062 return devices_[ device ];
7065 int openMode = SND_PCM_ASYNC;
7066 snd_pcm_stream_t stream;
7067 snd_pcm_info_t *pcminfo;
7068 snd_pcm_info_alloca( &pcminfo );
7070 snd_pcm_hw_params_t *params;
7071 snd_pcm_hw_params_alloca( ¶ms );
7073 // First try for playback unless default device (which has subdev -1)
7074 stream = SND_PCM_STREAM_PLAYBACK;
7075 snd_pcm_info_set_stream( pcminfo, stream );
7076 if ( subdevice != -1 ) {
7077 snd_pcm_info_set_device( pcminfo, subdevice );
7078 snd_pcm_info_set_subdevice( pcminfo, 0 );
7080 result = snd_ctl_pcm_info( chandle, pcminfo );
7082 // Device probably doesn't support playback.
7087 result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK );
7089 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
7090 errorText_ = errorStream_.str();
7091 error( RtAudioError::WARNING );
7095 // The device is open ... fill the parameter structure.
7096 result = snd_pcm_hw_params_any( phandle, params );
7098 snd_pcm_close( phandle );
7099 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
7100 errorText_ = errorStream_.str();
7101 error( RtAudioError::WARNING );
7105 // Get output channel information.
7107 result = snd_pcm_hw_params_get_channels_max( params, &value );
7109 snd_pcm_close( phandle );
7110 errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << ".";
7111 errorText_ = errorStream_.str();
7112 error( RtAudioError::WARNING );
7115 info.outputChannels = value;
7116 snd_pcm_close( phandle );
7119 stream = SND_PCM_STREAM_CAPTURE;
7120 snd_pcm_info_set_stream( pcminfo, stream );
7122 // Now try for capture unless default device (with subdev = -1)
7123 if ( subdevice != -1 ) {
7124 result = snd_ctl_pcm_info( chandle, pcminfo );
7125 snd_ctl_close( chandle );
7127 // Device probably doesn't support capture.
7128 if ( info.outputChannels == 0 ) return info;
7129 goto probeParameters;
7133 snd_ctl_close( chandle );
7135 result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
7137 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
7138 errorText_ = errorStream_.str();
7139 error( RtAudioError::WARNING );
7140 if ( info.outputChannels == 0 ) return info;
7141 goto probeParameters;
7144 // The device is open ... fill the parameter structure.
7145 result = snd_pcm_hw_params_any( phandle, params );
7147 snd_pcm_close( phandle );
7148 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
7149 errorText_ = errorStream_.str();
7150 error( RtAudioError::WARNING );
7151 if ( info.outputChannels == 0 ) return info;
7152 goto probeParameters;
7155 result = snd_pcm_hw_params_get_channels_max( params, &value );
7157 snd_pcm_close( phandle );
7158 errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << ".";
7159 errorText_ = errorStream_.str();
7160 error( RtAudioError::WARNING );
7161 if ( info.outputChannels == 0 ) return info;
7162 goto probeParameters;
7164 info.inputChannels = value;
7165 snd_pcm_close( phandle );
7167 // If device opens for both playback and capture, we determine the channels.
7168 if ( info.outputChannels > 0 && info.inputChannels > 0 )
7169 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
7171 // ALSA doesn't provide default devices so we'll use the first available one.
7172 if ( device == 0 && info.outputChannels > 0 )
7173 info.isDefaultOutput = true;
7174 if ( device == 0 && info.inputChannels > 0 )
7175 info.isDefaultInput = true;
7178 // At this point, we just need to figure out the supported data
7179 // formats and sample rates. We'll proceed by opening the device in
7180 // the direction with the maximum number of channels, or playback if
7181 // they are equal. This might limit our sample rate options, but so
7184 if ( info.outputChannels >= info.inputChannels )
7185 stream = SND_PCM_STREAM_PLAYBACK;
7187 stream = SND_PCM_STREAM_CAPTURE;
7188 snd_pcm_info_set_stream( pcminfo, stream );
7190 result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
7192 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
7193 errorText_ = errorStream_.str();
7194 error( RtAudioError::WARNING );
7198 // The device is open ... fill the parameter structure.
7199 result = snd_pcm_hw_params_any( phandle, params );
7201 snd_pcm_close( phandle );
7202 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
7203 errorText_ = errorStream_.str();
7204 error( RtAudioError::WARNING );
7208 // Test our discrete set of sample rate values.
7209 info.sampleRates.clear();
7210 for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {
7211 if ( snd_pcm_hw_params_test_rate( phandle, params, SAMPLE_RATES[i], 0 ) == 0 ) {
7212 info.sampleRates.push_back( SAMPLE_RATES[i] );
7214 if ( !info.preferredSampleRate || ( SAMPLE_RATES[i] <= 48000 && SAMPLE_RATES[i] > info.preferredSampleRate ) )
7215 info.preferredSampleRate = SAMPLE_RATES[i];
7218 if ( info.sampleRates.size() == 0 ) {
7219 snd_pcm_close( phandle );
7220 errorStream_ << "RtApiAlsa::getDeviceInfo: no supported sample rates found for device (" << name << ").";
7221 errorText_ = errorStream_.str();
7222 error( RtAudioError::WARNING );
7226 // Probe the supported data formats ... we don't care about endian-ness just yet
7227 snd_pcm_format_t format;
7228 info.nativeFormats = 0;
7229 format = SND_PCM_FORMAT_S8;
7230 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7231 info.nativeFormats |= RTAUDIO_SINT8;
7232 format = SND_PCM_FORMAT_S16;
7233 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7234 info.nativeFormats |= RTAUDIO_SINT16;
7235 format = SND_PCM_FORMAT_S24;
7236 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7237 info.nativeFormats |= RTAUDIO_SINT24;
7238 format = SND_PCM_FORMAT_S32;
7239 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7240 info.nativeFormats |= RTAUDIO_SINT32;
7241 format = SND_PCM_FORMAT_FLOAT;
7242 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7243 info.nativeFormats |= RTAUDIO_FLOAT32;
7244 format = SND_PCM_FORMAT_FLOAT64;
7245 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7246 info.nativeFormats |= RTAUDIO_FLOAT64;
7248 // Check that we have at least one supported format
7249 if ( info.nativeFormats == 0 ) {
7250 snd_pcm_close( phandle );
7251 errorStream_ << "RtApiAlsa::getDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio.";
7252 errorText_ = errorStream_.str();
7253 error( RtAudioError::WARNING );
7257 // Get the device name
7259 result = snd_card_get_name( card, &cardname );
7260 if ( result >= 0 ) {
7261 sprintf( name, "hw:%s,%d", cardname, subdevice );
7266 // That's all ... close the device and return
7267 snd_pcm_close( phandle );
7272 void RtApiAlsa :: saveDeviceInfo( void )
7276 unsigned int nDevices = getDeviceCount();
7277 devices_.resize( nDevices );
7278 for ( unsigned int i=0; i<nDevices; i++ )
7279 devices_[i] = getDeviceInfo( i );
7282 bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
7283 unsigned int firstChannel, unsigned int sampleRate,
7284 RtAudioFormat format, unsigned int *bufferSize,
7285 RtAudio::StreamOptions *options )
7288 #if defined(__RTAUDIO_DEBUG__)
7290 snd_output_stdio_attach(&out, stderr, 0);
7293 // I'm not using the "plug" interface ... too much inconsistent behavior.
7295 unsigned nDevices = 0;
7296 int result, subdevice, card;
7300 if ( options && options->flags & RTAUDIO_ALSA_USE_DEFAULT )
7301 snprintf(name, sizeof(name), "%s", "default");
7303 // Count cards and devices
7305 snd_card_next( &card );
7306 while ( card >= 0 ) {
7307 sprintf( name, "hw:%d", card );
7308 result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
7310 errorStream_ << "RtApiAlsa::probeDeviceOpen: control open, card = " << card << ", " << snd_strerror( result ) << ".";
7311 errorText_ = errorStream_.str();
7316 result = snd_ctl_pcm_next_device( chandle, &subdevice );
7317 if ( result < 0 ) break;
7318 if ( subdevice < 0 ) break;
7319 if ( nDevices == device ) {
7320 sprintf( name, "hw:%d,%d", card, subdevice );
7321 snd_ctl_close( chandle );
7326 snd_ctl_close( chandle );
7327 snd_card_next( &card );
7330 result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
7331 if ( result == 0 ) {
7332 if ( nDevices == device ) {
7333 strcpy( name, "default" );
7339 if ( nDevices == 0 ) {
7340 // This should not happen because a check is made before this function is called.
7341 errorText_ = "RtApiAlsa::probeDeviceOpen: no devices found!";
7345 if ( device >= nDevices ) {
7346 // This should not happen because a check is made before this function is called.
7347 errorText_ = "RtApiAlsa::probeDeviceOpen: device ID is invalid!";
7354 // The getDeviceInfo() function will not work for a device that is
7355 // already open. Thus, we'll probe the system before opening a
7356 // stream and save the results for use by getDeviceInfo().
7357 if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) // only do once
7358 this->saveDeviceInfo();
7360 snd_pcm_stream_t stream;
7361 if ( mode == OUTPUT )
7362 stream = SND_PCM_STREAM_PLAYBACK;
7364 stream = SND_PCM_STREAM_CAPTURE;
7367 int openMode = SND_PCM_ASYNC;
7368 result = snd_pcm_open( &phandle, name, stream, openMode );
7370 if ( mode == OUTPUT )
7371 errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for output.";
7373 errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for input.";
7374 errorText_ = errorStream_.str();
7378 // Fill the parameter structure.
7379 snd_pcm_hw_params_t *hw_params;
7380 snd_pcm_hw_params_alloca( &hw_params );
7381 result = snd_pcm_hw_params_any( phandle, hw_params );
7383 snd_pcm_close( phandle );
7384 errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") parameters, " << snd_strerror( result ) << ".";
7385 errorText_ = errorStream_.str();
7389 #if defined(__RTAUDIO_DEBUG__)
7390 fprintf( stderr, "\nRtApiAlsa: dump hardware params just after device open:\n\n" );
7391 snd_pcm_hw_params_dump( hw_params, out );
7394 // Set access ... check user preference.
7395 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) {
7396 stream_.userInterleaved = false;
7397 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
7399 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
7400 stream_.deviceInterleaved[mode] = true;
7403 stream_.deviceInterleaved[mode] = false;
7406 stream_.userInterleaved = true;
7407 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
7409 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
7410 stream_.deviceInterleaved[mode] = false;
7413 stream_.deviceInterleaved[mode] = true;
7417 snd_pcm_close( phandle );
7418 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") access, " << snd_strerror( result ) << ".";
7419 errorText_ = errorStream_.str();
7423 // Determine how to set the device format.
7424 stream_.userFormat = format;
7425 snd_pcm_format_t deviceFormat = SND_PCM_FORMAT_UNKNOWN;
7427 if ( format == RTAUDIO_SINT8 )
7428 deviceFormat = SND_PCM_FORMAT_S8;
7429 else if ( format == RTAUDIO_SINT16 )
7430 deviceFormat = SND_PCM_FORMAT_S16;
7431 else if ( format == RTAUDIO_SINT24 )
7432 deviceFormat = SND_PCM_FORMAT_S24;
7433 else if ( format == RTAUDIO_SINT32 )
7434 deviceFormat = SND_PCM_FORMAT_S32;
7435 else if ( format == RTAUDIO_FLOAT32 )
7436 deviceFormat = SND_PCM_FORMAT_FLOAT;
7437 else if ( format == RTAUDIO_FLOAT64 )
7438 deviceFormat = SND_PCM_FORMAT_FLOAT64;
7440 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat) == 0) {
7441 stream_.deviceFormat[mode] = format;
7445 // The user requested format is not natively supported by the device.
7446 deviceFormat = SND_PCM_FORMAT_FLOAT64;
7447 if ( snd_pcm_hw_params_test_format( phandle, hw_params, deviceFormat ) == 0 ) {
7448 stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
7452 deviceFormat = SND_PCM_FORMAT_FLOAT;
7453 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7454 stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
7458 deviceFormat = SND_PCM_FORMAT_S32;
7459 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7460 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
7464 deviceFormat = SND_PCM_FORMAT_S24;
7465 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7466 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
7470 deviceFormat = SND_PCM_FORMAT_S16;
7471 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7472 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
7476 deviceFormat = SND_PCM_FORMAT_S8;
7477 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7478 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
7482 // If we get here, no supported format was found.
7483 snd_pcm_close( phandle );
7484 errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device " << device << " data format not supported by RtAudio.";
7485 errorText_ = errorStream_.str();
7489 result = snd_pcm_hw_params_set_format( phandle, hw_params, deviceFormat );
7491 snd_pcm_close( phandle );
7492 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") data format, " << snd_strerror( result ) << ".";
7493 errorText_ = errorStream_.str();
7497 // Determine whether byte-swaping is necessary.
7498 stream_.doByteSwap[mode] = false;
7499 if ( deviceFormat != SND_PCM_FORMAT_S8 ) {
7500 result = snd_pcm_format_cpu_endian( deviceFormat );
7502 stream_.doByteSwap[mode] = true;
7503 else if (result < 0) {
7504 snd_pcm_close( phandle );
7505 errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") endian-ness, " << snd_strerror( result ) << ".";
7506 errorText_ = errorStream_.str();
7511 // Set the sample rate.
7512 result = snd_pcm_hw_params_set_rate_near( phandle, hw_params, (unsigned int*) &sampleRate, 0 );
7514 snd_pcm_close( phandle );
7515 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting sample rate on device (" << name << "), " << snd_strerror( result ) << ".";
7516 errorText_ = errorStream_.str();
7520 // Determine the number of channels for this device. We support a possible
7521 // minimum device channel number > than the value requested by the user.
7522 stream_.nUserChannels[mode] = channels;
7524 result = snd_pcm_hw_params_get_channels_max( hw_params, &value );
7525 unsigned int deviceChannels = value;
7526 if ( result < 0 || deviceChannels < channels + firstChannel ) {
7527 snd_pcm_close( phandle );
7528 errorStream_ << "RtApiAlsa::probeDeviceOpen: requested channel parameters not supported by device (" << name << "), " << snd_strerror( result ) << ".";
7529 errorText_ = errorStream_.str();
7533 result = snd_pcm_hw_params_get_channels_min( hw_params, &value );
7535 snd_pcm_close( phandle );
7536 errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting minimum channels for device (" << name << "), " << snd_strerror( result ) << ".";
7537 errorText_ = errorStream_.str();
7540 deviceChannels = value;
7541 if ( deviceChannels < channels + firstChannel ) deviceChannels = channels + firstChannel;
7542 stream_.nDeviceChannels[mode] = deviceChannels;
7544 // Set the device channels.
7545 result = snd_pcm_hw_params_set_channels( phandle, hw_params, deviceChannels );
7547 snd_pcm_close( phandle );
7548 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting channels for device (" << name << "), " << snd_strerror( result ) << ".";
7549 errorText_ = errorStream_.str();
7553 // Set the buffer (or period) size.
7555 snd_pcm_uframes_t periodSize = *bufferSize;
7556 result = snd_pcm_hw_params_set_period_size_near( phandle, hw_params, &periodSize, &dir );
7558 snd_pcm_close( phandle );
7559 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting period size for device (" << name << "), " << snd_strerror( result ) << ".";
7560 errorText_ = errorStream_.str();
7563 *bufferSize = periodSize;
7565 // Set the buffer number, which in ALSA is referred to as the "period".
7566 unsigned int periods = 0;
7567 if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2;
7568 if ( options && options->numberOfBuffers > 0 ) periods = options->numberOfBuffers;
7569 if ( periods < 2 ) periods = 4; // a fairly safe default value
7570 result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir );
7572 snd_pcm_close( phandle );
7573 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << ".";
7574 errorText_ = errorStream_.str();
7578 // If attempting to setup a duplex stream, the bufferSize parameter
7579 // MUST be the same in both directions!
7580 if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {
7581 snd_pcm_close( phandle );
7582 errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ").";
7583 errorText_ = errorStream_.str();
7587 stream_.bufferSize = *bufferSize;
7589 // Install the hardware configuration
7590 result = snd_pcm_hw_params( phandle, hw_params );
7592 snd_pcm_close( phandle );
7593 errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing hardware configuration on device (" << name << "), " << snd_strerror( result ) << ".";
7594 errorText_ = errorStream_.str();
7598 #if defined(__RTAUDIO_DEBUG__)
7599 fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n");
7600 snd_pcm_hw_params_dump( hw_params, out );
7603 // Set the software configuration to fill buffers with zeros and prevent device stopping on xruns.
7604 snd_pcm_sw_params_t *sw_params = NULL;
7605 snd_pcm_sw_params_alloca( &sw_params );
7606 snd_pcm_sw_params_current( phandle, sw_params );
7607 snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize );
7608 snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, ULONG_MAX );
7609 snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 );
7611 // The following two settings were suggested by Theo Veenker
7612 //snd_pcm_sw_params_set_avail_min( phandle, sw_params, *bufferSize );
7613 //snd_pcm_sw_params_set_xfer_align( phandle, sw_params, 1 );
7615 // here are two options for a fix
7616 //snd_pcm_sw_params_set_silence_size( phandle, sw_params, ULONG_MAX );
7617 snd_pcm_uframes_t val;
7618 snd_pcm_sw_params_get_boundary( sw_params, &val );
7619 snd_pcm_sw_params_set_silence_size( phandle, sw_params, val );
7621 result = snd_pcm_sw_params( phandle, sw_params );
7623 snd_pcm_close( phandle );
7624 errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing software configuration on device (" << name << "), " << snd_strerror( result ) << ".";
7625 errorText_ = errorStream_.str();
7629 #if defined(__RTAUDIO_DEBUG__)
7630 fprintf(stderr, "\nRtApiAlsa: dump software params after installation:\n\n");
7631 snd_pcm_sw_params_dump( sw_params, out );
7634 // Set flags for buffer conversion
7635 stream_.doConvertBuffer[mode] = false;
7636 if ( stream_.userFormat != stream_.deviceFormat[mode] )
7637 stream_.doConvertBuffer[mode] = true;
7638 if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
7639 stream_.doConvertBuffer[mode] = true;
7640 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
7641 stream_.nUserChannels[mode] > 1 )
7642 stream_.doConvertBuffer[mode] = true;
7644 // Allocate the ApiHandle if necessary and then save.
7645 AlsaHandle *apiInfo = 0;
7646 if ( stream_.apiHandle == 0 ) {
7648 apiInfo = (AlsaHandle *) new AlsaHandle;
7650 catch ( std::bad_alloc& ) {
7651 errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating AlsaHandle memory.";
7655 if ( pthread_cond_init( &apiInfo->runnable_cv, NULL ) ) {
7656 errorText_ = "RtApiAlsa::probeDeviceOpen: error initializing pthread condition variable.";
7660 stream_.apiHandle = (void *) apiInfo;
7661 apiInfo->handles[0] = 0;
7662 apiInfo->handles[1] = 0;
7665 apiInfo = (AlsaHandle *) stream_.apiHandle;
7667 apiInfo->handles[mode] = phandle;
7670 // Allocate necessary internal buffers.
7671 unsigned long bufferBytes;
7672 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
7673 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
7674 if ( stream_.userBuffer[mode] == NULL ) {
7675 errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating user buffer memory.";
7679 if ( stream_.doConvertBuffer[mode] ) {
7681 bool makeBuffer = true;
7682 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
7683 if ( mode == INPUT ) {
7684 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
7685 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
7686 if ( bufferBytes <= bytesOut ) makeBuffer = false;
7691 bufferBytes *= *bufferSize;
7692 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
7693 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
7694 if ( stream_.deviceBuffer == NULL ) {
7695 errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating device buffer memory.";
7701 stream_.sampleRate = sampleRate;
7702 stream_.nBuffers = periods;
7703 stream_.device[mode] = device;
7704 stream_.state = STREAM_STOPPED;
7706 // Setup the buffer conversion information structure.
7707 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
7709 // Setup thread if necessary.
7710 if ( stream_.mode == OUTPUT && mode == INPUT ) {
7711 // We had already set up an output stream.
7712 stream_.mode = DUPLEX;
7713 // Link the streams if possible.
7714 apiInfo->synchronized = false;
7715 if ( snd_pcm_link( apiInfo->handles[0], apiInfo->handles[1] ) == 0 )
7716 apiInfo->synchronized = true;
7718 errorText_ = "RtApiAlsa::probeDeviceOpen: unable to synchronize input and output devices.";
7719 error( RtAudioError::WARNING );
7723 stream_.mode = mode;
7725 // Setup callback thread.
7726 stream_.callbackInfo.object = (void *) this;
7728 // Set the thread attributes for joinable and realtime scheduling
7729 // priority (optional). The higher priority will only take affect
7730 // if the program is run as root or suid. Note, under Linux
7731 // processes with CAP_SYS_NICE privilege, a user can change
7732 // scheduling policy and priority (thus need not be root). See
7733 // POSIX "capabilities".
7734 pthread_attr_t attr;
7735 pthread_attr_init( &attr );
7736 pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
7738 #ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
7739 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
7740 // We previously attempted to increase the audio callback priority
7741 // to SCHED_RR here via the attributes. However, while no errors
7742 // were reported in doing so, it did not work. So, now this is
7743 // done in the alsaCallbackHandler function.
7744 stream_.callbackInfo.doRealtime = true;
7745 int priority = options->priority;
7746 int min = sched_get_priority_min( SCHED_RR );
7747 int max = sched_get_priority_max( SCHED_RR );
7748 if ( priority < min ) priority = min;
7749 else if ( priority > max ) priority = max;
7750 stream_.callbackInfo.priority = priority;
7754 stream_.callbackInfo.isRunning = true;
7755 result = pthread_create( &stream_.callbackInfo.thread, &attr, alsaCallbackHandler, &stream_.callbackInfo );
7756 pthread_attr_destroy( &attr );
7758 stream_.callbackInfo.isRunning = false;
7759 errorText_ = "RtApiAlsa::error creating callback thread!";
7768 pthread_cond_destroy( &apiInfo->runnable_cv );
7769 if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
7770 if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
7772 stream_.apiHandle = 0;
7775 if ( phandle) snd_pcm_close( phandle );
7777 for ( int i=0; i<2; i++ ) {
7778 if ( stream_.userBuffer[i] ) {
7779 free( stream_.userBuffer[i] );
7780 stream_.userBuffer[i] = 0;
7784 if ( stream_.deviceBuffer ) {
7785 free( stream_.deviceBuffer );
7786 stream_.deviceBuffer = 0;
7789 stream_.state = STREAM_CLOSED;
7793 void RtApiAlsa :: closeStream()
7795 if ( stream_.state == STREAM_CLOSED ) {
7796 errorText_ = "RtApiAlsa::closeStream(): no open stream to close!";
7797 error( RtAudioError::WARNING );
7801 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
7802 stream_.callbackInfo.isRunning = false;
7803 MUTEX_LOCK( &stream_.mutex );
7804 if ( stream_.state == STREAM_STOPPED ) {
7805 apiInfo->runnable = true;
7806 pthread_cond_signal( &apiInfo->runnable_cv );
7808 MUTEX_UNLOCK( &stream_.mutex );
7809 pthread_join( stream_.callbackInfo.thread, NULL );
7811 if ( stream_.state == STREAM_RUNNING ) {
7812 stream_.state = STREAM_STOPPED;
7813 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
7814 snd_pcm_drop( apiInfo->handles[0] );
7815 if ( stream_.mode == INPUT || stream_.mode == DUPLEX )
7816 snd_pcm_drop( apiInfo->handles[1] );
7820 pthread_cond_destroy( &apiInfo->runnable_cv );
7821 if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
7822 if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
7824 stream_.apiHandle = 0;
7827 for ( int i=0; i<2; i++ ) {
7828 if ( stream_.userBuffer[i] ) {
7829 free( stream_.userBuffer[i] );
7830 stream_.userBuffer[i] = 0;
7834 if ( stream_.deviceBuffer ) {
7835 free( stream_.deviceBuffer );
7836 stream_.deviceBuffer = 0;
7839 stream_.mode = UNINITIALIZED;
7840 stream_.state = STREAM_CLOSED;
7843 void RtApiAlsa :: startStream()
7845 // This method calls snd_pcm_prepare if the device isn't already in that state.
7848 if ( stream_.state == STREAM_RUNNING ) {
7849 errorText_ = "RtApiAlsa::startStream(): the stream is already running!";
7850 error( RtAudioError::WARNING );
7854 MUTEX_LOCK( &stream_.mutex );
7857 snd_pcm_state_t state;
7858 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
7859 snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
7860 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
7861 state = snd_pcm_state( handle[0] );
7862 if ( state != SND_PCM_STATE_PREPARED ) {
7863 result = snd_pcm_prepare( handle[0] );
7865 errorStream_ << "RtApiAlsa::startStream: error preparing output pcm device, " << snd_strerror( result ) << ".";
7866 errorText_ = errorStream_.str();
7872 if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
7873 result = snd_pcm_drop(handle[1]); // fix to remove stale data received since device has been open
7874 state = snd_pcm_state( handle[1] );
7875 if ( state != SND_PCM_STATE_PREPARED ) {
7876 result = snd_pcm_prepare( handle[1] );
7878 errorStream_ << "RtApiAlsa::startStream: error preparing input pcm device, " << snd_strerror( result ) << ".";
7879 errorText_ = errorStream_.str();
7885 stream_.state = STREAM_RUNNING;
7888 apiInfo->runnable = true;
7889 pthread_cond_signal( &apiInfo->runnable_cv );
7890 MUTEX_UNLOCK( &stream_.mutex );
7892 if ( result >= 0 ) return;
7893 error( RtAudioError::SYSTEM_ERROR );
7896 void RtApiAlsa :: stopStream()
7899 if ( stream_.state == STREAM_STOPPED ) {
7900 errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!";
7901 error( RtAudioError::WARNING );
7905 stream_.state = STREAM_STOPPED;
7906 MUTEX_LOCK( &stream_.mutex );
7909 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
7910 snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
7911 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
7912 if ( apiInfo->synchronized )
7913 result = snd_pcm_drop( handle[0] );
7915 result = snd_pcm_drain( handle[0] );
7917 errorStream_ << "RtApiAlsa::stopStream: error draining output pcm device, " << snd_strerror( result ) << ".";
7918 errorText_ = errorStream_.str();
7923 if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
7924 result = snd_pcm_drop( handle[1] );
7926 errorStream_ << "RtApiAlsa::stopStream: error stopping input pcm device, " << snd_strerror( result ) << ".";
7927 errorText_ = errorStream_.str();
7933 apiInfo->runnable = false; // fixes high CPU usage when stopped
7934 MUTEX_UNLOCK( &stream_.mutex );
7936 if ( result >= 0 ) return;
7937 error( RtAudioError::SYSTEM_ERROR );
7940 void RtApiAlsa :: abortStream()
7943 if ( stream_.state == STREAM_STOPPED ) {
7944 errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!";
7945 error( RtAudioError::WARNING );
7949 stream_.state = STREAM_STOPPED;
7950 MUTEX_LOCK( &stream_.mutex );
7953 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
7954 snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
7955 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
7956 result = snd_pcm_drop( handle[0] );
7958 errorStream_ << "RtApiAlsa::abortStream: error aborting output pcm device, " << snd_strerror( result ) << ".";
7959 errorText_ = errorStream_.str();
7964 if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
7965 result = snd_pcm_drop( handle[1] );
7967 errorStream_ << "RtApiAlsa::abortStream: error aborting input pcm device, " << snd_strerror( result ) << ".";
7968 errorText_ = errorStream_.str();
7974 apiInfo->runnable = false; // fixes high CPU usage when stopped
7975 MUTEX_UNLOCK( &stream_.mutex );
7977 if ( result >= 0 ) return;
7978 error( RtAudioError::SYSTEM_ERROR );
7981 void RtApiAlsa :: callbackEvent()
7983 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
7984 if ( stream_.state == STREAM_STOPPED ) {
7985 MUTEX_LOCK( &stream_.mutex );
7986 while ( !apiInfo->runnable )
7987 pthread_cond_wait( &apiInfo->runnable_cv, &stream_.mutex );
7989 if ( stream_.state != STREAM_RUNNING ) {
7990 MUTEX_UNLOCK( &stream_.mutex );
7993 MUTEX_UNLOCK( &stream_.mutex );
7996 if ( stream_.state == STREAM_CLOSED ) {
7997 errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!";
7998 error( RtAudioError::WARNING );
8002 int doStopStream = 0;
8003 RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
8004 double streamTime = getStreamTime();
8005 RtAudioStreamStatus status = 0;
8006 if ( stream_.mode != INPUT && apiInfo->xrun[0] == true ) {
8007 status |= RTAUDIO_OUTPUT_UNDERFLOW;
8008 apiInfo->xrun[0] = false;
8010 if ( stream_.mode != OUTPUT && apiInfo->xrun[1] == true ) {
8011 status |= RTAUDIO_INPUT_OVERFLOW;
8012 apiInfo->xrun[1] = false;
8014 doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
8015 stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
8017 if ( doStopStream == 2 ) {
8022 MUTEX_LOCK( &stream_.mutex );
8024 // The state might change while waiting on a mutex.
8025 if ( stream_.state == STREAM_STOPPED ) goto unlock;
8031 snd_pcm_sframes_t frames;
8032 RtAudioFormat format;
8033 handle = (snd_pcm_t **) apiInfo->handles;
8035 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
8037 // Setup parameters.
8038 if ( stream_.doConvertBuffer[1] ) {
8039 buffer = stream_.deviceBuffer;
8040 channels = stream_.nDeviceChannels[1];
8041 format = stream_.deviceFormat[1];
8044 buffer = stream_.userBuffer[1];
8045 channels = stream_.nUserChannels[1];
8046 format = stream_.userFormat;
8049 // Read samples from device in interleaved/non-interleaved format.
8050 if ( stream_.deviceInterleaved[1] )
8051 result = snd_pcm_readi( handle[1], buffer, stream_.bufferSize );
8053 void *bufs[channels];
8054 size_t offset = stream_.bufferSize * formatBytes( format );
8055 for ( int i=0; i<channels; i++ )
8056 bufs[i] = (void *) (buffer + (i * offset));
8057 result = snd_pcm_readn( handle[1], bufs, stream_.bufferSize );
8060 if ( result < (int) stream_.bufferSize ) {
8061 // Either an error or overrun occured.
8062 if ( result == -EPIPE ) {
8063 snd_pcm_state_t state = snd_pcm_state( handle[1] );
8064 if ( state == SND_PCM_STATE_XRUN ) {
8065 apiInfo->xrun[1] = true;
8066 result = snd_pcm_prepare( handle[1] );
8068 errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after overrun, " << snd_strerror( result ) << ".";
8069 errorText_ = errorStream_.str();
8073 errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
8074 errorText_ = errorStream_.str();
8078 errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << ".";
8079 errorText_ = errorStream_.str();
8081 error( RtAudioError::WARNING );
8085 // Do byte swapping if necessary.
8086 if ( stream_.doByteSwap[1] )
8087 byteSwapBuffer( buffer, stream_.bufferSize * channels, format );
8089 // Do buffer conversion if necessary.
8090 if ( stream_.doConvertBuffer[1] )
8091 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
8093 // Check stream latency
8094 result = snd_pcm_delay( handle[1], &frames );
8095 if ( result == 0 && frames > 0 ) stream_.latency[1] = frames;
8100 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
8102 // Setup parameters and do buffer conversion if necessary.
8103 if ( stream_.doConvertBuffer[0] ) {
8104 buffer = stream_.deviceBuffer;
8105 convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
8106 channels = stream_.nDeviceChannels[0];
8107 format = stream_.deviceFormat[0];
8110 buffer = stream_.userBuffer[0];
8111 channels = stream_.nUserChannels[0];
8112 format = stream_.userFormat;
8115 // Do byte swapping if necessary.
8116 if ( stream_.doByteSwap[0] )
8117 byteSwapBuffer(buffer, stream_.bufferSize * channels, format);
8119 // Write samples to device in interleaved/non-interleaved format.
8120 if ( stream_.deviceInterleaved[0] )
8121 result = snd_pcm_writei( handle[0], buffer, stream_.bufferSize );
8123 void *bufs[channels];
8124 size_t offset = stream_.bufferSize * formatBytes( format );
8125 for ( int i=0; i<channels; i++ )
8126 bufs[i] = (void *) (buffer + (i * offset));
8127 result = snd_pcm_writen( handle[0], bufs, stream_.bufferSize );
8130 if ( result < (int) stream_.bufferSize ) {
8131 // Either an error or underrun occured.
8132 if ( result == -EPIPE ) {
8133 snd_pcm_state_t state = snd_pcm_state( handle[0] );
8134 if ( state == SND_PCM_STATE_XRUN ) {
8135 apiInfo->xrun[0] = true;
8136 result = snd_pcm_prepare( handle[0] );
8138 errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after underrun, " << snd_strerror( result ) << ".";
8139 errorText_ = errorStream_.str();
8142 errorText_ = "RtApiAlsa::callbackEvent: audio write error, underrun.";
8145 errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
8146 errorText_ = errorStream_.str();
8150 errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << ".";
8151 errorText_ = errorStream_.str();
8153 error( RtAudioError::WARNING );
8157 // Check stream latency
8158 result = snd_pcm_delay( handle[0], &frames );
8159 if ( result == 0 && frames > 0 ) stream_.latency[0] = frames;
8163 MUTEX_UNLOCK( &stream_.mutex );
8165 RtApi::tickStreamTime();
8166 if ( doStopStream == 1 ) this->stopStream();
8169 static void *alsaCallbackHandler( void *ptr )
8171 CallbackInfo *info = (CallbackInfo *) ptr;
8172 RtApiAlsa *object = (RtApiAlsa *) info->object;
8173 bool *isRunning = &info->isRunning;
8175 #ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
8176 if ( info->doRealtime ) {
8177 pthread_t tID = pthread_self(); // ID of this thread
8178 sched_param prio = { info->priority }; // scheduling priority of thread
8179 pthread_setschedparam( tID, SCHED_RR, &prio );
8183 while ( *isRunning == true ) {
8184 pthread_testcancel();
8185 object->callbackEvent();
8188 pthread_exit( NULL );
8191 //******************** End of __LINUX_ALSA__ *********************//
8194 #if defined(__LINUX_PULSE__)
8196 // Code written by Peter Meerwald, pmeerw@pmeerw.net
8197 // and Tristan Matthews.
8199 #include <pulse/error.h>
8200 #include <pulse/simple.h>
8203 static const unsigned int SUPPORTED_SAMPLERATES[] = { 8000, 16000, 22050, 32000,
8204 44100, 48000, 96000, 0};
8206 struct rtaudio_pa_format_mapping_t {
8207 RtAudioFormat rtaudio_format;
8208 pa_sample_format_t pa_format;
8211 static const rtaudio_pa_format_mapping_t supported_sampleformats[] = {
8212 {RTAUDIO_SINT16, PA_SAMPLE_S16LE},
8213 {RTAUDIO_SINT32, PA_SAMPLE_S32LE},
8214 {RTAUDIO_FLOAT32, PA_SAMPLE_FLOAT32LE},
8215 {0, PA_SAMPLE_INVALID}};
8217 struct PulseAudioHandle {
8221 pthread_cond_t runnable_cv;
8223 PulseAudioHandle() : s_play(0), s_rec(0), runnable(false) { }
8226 RtApiPulse::~RtApiPulse()
8228 if ( stream_.state != STREAM_CLOSED )
8232 unsigned int RtApiPulse::getDeviceCount( void )
8237 RtAudio::DeviceInfo RtApiPulse::getDeviceInfo( unsigned int /*device*/ )
8239 RtAudio::DeviceInfo info;
8241 info.name = "PulseAudio";
8242 info.outputChannels = 2;
8243 info.inputChannels = 2;
8244 info.duplexChannels = 2;
8245 info.isDefaultOutput = true;
8246 info.isDefaultInput = true;
8248 for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr )
8249 info.sampleRates.push_back( *sr );
8251 info.preferredSampleRate = 48000;
8252 info.nativeFormats = RTAUDIO_SINT16 | RTAUDIO_SINT32 | RTAUDIO_FLOAT32;
8257 static void *pulseaudio_callback( void * user )
8259 CallbackInfo *cbi = static_cast<CallbackInfo *>( user );
8260 RtApiPulse *context = static_cast<RtApiPulse *>( cbi->object );
8261 volatile bool *isRunning = &cbi->isRunning;
8263 while ( *isRunning ) {
8264 pthread_testcancel();
8265 context->callbackEvent();
8268 pthread_exit( NULL );
8271 void RtApiPulse::closeStream( void )
8273 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8275 stream_.callbackInfo.isRunning = false;
8277 MUTEX_LOCK( &stream_.mutex );
8278 if ( stream_.state == STREAM_STOPPED ) {
8279 pah->runnable = true;
8280 pthread_cond_signal( &pah->runnable_cv );
8282 MUTEX_UNLOCK( &stream_.mutex );
8284 pthread_join( pah->thread, 0 );
8285 if ( pah->s_play ) {
8286 pa_simple_flush( pah->s_play, NULL );
8287 pa_simple_free( pah->s_play );
8290 pa_simple_free( pah->s_rec );
8292 pthread_cond_destroy( &pah->runnable_cv );
8294 stream_.apiHandle = 0;
8297 if ( stream_.userBuffer[0] ) {
8298 free( stream_.userBuffer[0] );
8299 stream_.userBuffer[0] = 0;
8301 if ( stream_.userBuffer[1] ) {
8302 free( stream_.userBuffer[1] );
8303 stream_.userBuffer[1] = 0;
8306 stream_.state = STREAM_CLOSED;
8307 stream_.mode = UNINITIALIZED;
8310 void RtApiPulse::callbackEvent( void )
8312 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8314 if ( stream_.state == STREAM_STOPPED ) {
8315 MUTEX_LOCK( &stream_.mutex );
8316 while ( !pah->runnable )
8317 pthread_cond_wait( &pah->runnable_cv, &stream_.mutex );
8319 if ( stream_.state != STREAM_RUNNING ) {
8320 MUTEX_UNLOCK( &stream_.mutex );
8323 MUTEX_UNLOCK( &stream_.mutex );
8326 if ( stream_.state == STREAM_CLOSED ) {
8327 errorText_ = "RtApiPulse::callbackEvent(): the stream is closed ... "
8328 "this shouldn't happen!";
8329 error( RtAudioError::WARNING );
8333 RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
8334 double streamTime = getStreamTime();
8335 RtAudioStreamStatus status = 0;
8336 int doStopStream = callback( stream_.userBuffer[OUTPUT], stream_.userBuffer[INPUT],
8337 stream_.bufferSize, streamTime, status,
8338 stream_.callbackInfo.userData );
8340 if ( doStopStream == 2 ) {
8345 MUTEX_LOCK( &stream_.mutex );
8346 void *pulse_in = stream_.doConvertBuffer[INPUT] ? stream_.deviceBuffer : stream_.userBuffer[INPUT];
8347 void *pulse_out = stream_.doConvertBuffer[OUTPUT] ? stream_.deviceBuffer : stream_.userBuffer[OUTPUT];
8349 if ( stream_.state != STREAM_RUNNING )
8354 if (stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
8355 if ( stream_.doConvertBuffer[OUTPUT] ) {
8356 convertBuffer( stream_.deviceBuffer,
8357 stream_.userBuffer[OUTPUT],
8358 stream_.convertInfo[OUTPUT] );
8359 bytes = stream_.nDeviceChannels[OUTPUT] * stream_.bufferSize *
8360 formatBytes( stream_.deviceFormat[OUTPUT] );
8362 bytes = stream_.nUserChannels[OUTPUT] * stream_.bufferSize *
8363 formatBytes( stream_.userFormat );
8365 if ( pa_simple_write( pah->s_play, pulse_out, bytes, &pa_error ) < 0 ) {
8366 errorStream_ << "RtApiPulse::callbackEvent: audio write error, " <<
8367 pa_strerror( pa_error ) << ".";
8368 errorText_ = errorStream_.str();
8369 error( RtAudioError::WARNING );
8373 if ( stream_.mode == INPUT || stream_.mode == DUPLEX) {
8374 if ( stream_.doConvertBuffer[INPUT] )
8375 bytes = stream_.nDeviceChannels[INPUT] * stream_.bufferSize *
8376 formatBytes( stream_.deviceFormat[INPUT] );
8378 bytes = stream_.nUserChannels[INPUT] * stream_.bufferSize *
8379 formatBytes( stream_.userFormat );
8381 if ( pa_simple_read( pah->s_rec, pulse_in, bytes, &pa_error ) < 0 ) {
8382 errorStream_ << "RtApiPulse::callbackEvent: audio read error, " <<
8383 pa_strerror( pa_error ) << ".";
8384 errorText_ = errorStream_.str();
8385 error( RtAudioError::WARNING );
8387 if ( stream_.doConvertBuffer[INPUT] ) {
8388 convertBuffer( stream_.userBuffer[INPUT],
8389 stream_.deviceBuffer,
8390 stream_.convertInfo[INPUT] );
8395 MUTEX_UNLOCK( &stream_.mutex );
8396 RtApi::tickStreamTime();
8398 if ( doStopStream == 1 )
8402 void RtApiPulse::startStream( void )
8404 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8406 if ( stream_.state == STREAM_CLOSED ) {
8407 errorText_ = "RtApiPulse::startStream(): the stream is not open!";
8408 error( RtAudioError::INVALID_USE );
8411 if ( stream_.state == STREAM_RUNNING ) {
8412 errorText_ = "RtApiPulse::startStream(): the stream is already running!";
8413 error( RtAudioError::WARNING );
8417 MUTEX_LOCK( &stream_.mutex );
8419 stream_.state = STREAM_RUNNING;
8421 pah->runnable = true;
8422 pthread_cond_signal( &pah->runnable_cv );
8423 MUTEX_UNLOCK( &stream_.mutex );
8426 void RtApiPulse::stopStream( void )
8428 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8430 if ( stream_.state == STREAM_CLOSED ) {
8431 errorText_ = "RtApiPulse::stopStream(): the stream is not open!";
8432 error( RtAudioError::INVALID_USE );
8435 if ( stream_.state == STREAM_STOPPED ) {
8436 errorText_ = "RtApiPulse::stopStream(): the stream is already stopped!";
8437 error( RtAudioError::WARNING );
8441 stream_.state = STREAM_STOPPED;
8442 MUTEX_LOCK( &stream_.mutex );
8444 if ( pah && pah->s_play ) {
8446 if ( pa_simple_drain( pah->s_play, &pa_error ) < 0 ) {
8447 errorStream_ << "RtApiPulse::stopStream: error draining output device, " <<
8448 pa_strerror( pa_error ) << ".";
8449 errorText_ = errorStream_.str();
8450 MUTEX_UNLOCK( &stream_.mutex );
8451 error( RtAudioError::SYSTEM_ERROR );
8456 stream_.state = STREAM_STOPPED;
8457 MUTEX_UNLOCK( &stream_.mutex );
8460 void RtApiPulse::abortStream( void )
8462 PulseAudioHandle *pah = static_cast<PulseAudioHandle*>( stream_.apiHandle );
8464 if ( stream_.state == STREAM_CLOSED ) {
8465 errorText_ = "RtApiPulse::abortStream(): the stream is not open!";
8466 error( RtAudioError::INVALID_USE );
8469 if ( stream_.state == STREAM_STOPPED ) {
8470 errorText_ = "RtApiPulse::abortStream(): the stream is already stopped!";
8471 error( RtAudioError::WARNING );
8475 stream_.state = STREAM_STOPPED;
8476 MUTEX_LOCK( &stream_.mutex );
8478 if ( pah && pah->s_play ) {
8480 if ( pa_simple_flush( pah->s_play, &pa_error ) < 0 ) {
8481 errorStream_ << "RtApiPulse::abortStream: error flushing output device, " <<
8482 pa_strerror( pa_error ) << ".";
8483 errorText_ = errorStream_.str();
8484 MUTEX_UNLOCK( &stream_.mutex );
8485 error( RtAudioError::SYSTEM_ERROR );
8490 stream_.state = STREAM_STOPPED;
8491 MUTEX_UNLOCK( &stream_.mutex );
8494 bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode,
8495 unsigned int channels, unsigned int firstChannel,
8496 unsigned int sampleRate, RtAudioFormat format,
8497 unsigned int *bufferSize, RtAudio::StreamOptions *options )
8499 PulseAudioHandle *pah = 0;
8500 unsigned long bufferBytes = 0;
8503 if ( device != 0 ) return false;
8504 if ( mode != INPUT && mode != OUTPUT ) return false;
8505 if ( channels != 1 && channels != 2 ) {
8506 errorText_ = "RtApiPulse::probeDeviceOpen: unsupported number of channels.";
8509 ss.channels = channels;
8511 if ( firstChannel != 0 ) return false;
8513 bool sr_found = false;
8514 for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr ) {
8515 if ( sampleRate == *sr ) {
8517 stream_.sampleRate = sampleRate;
8518 ss.rate = sampleRate;
8523 errorText_ = "RtApiPulse::probeDeviceOpen: unsupported sample rate.";
8528 for ( const rtaudio_pa_format_mapping_t *sf = supported_sampleformats;
8529 sf->rtaudio_format && sf->pa_format != PA_SAMPLE_INVALID; ++sf ) {
8530 if ( format == sf->rtaudio_format ) {
8532 stream_.userFormat = sf->rtaudio_format;
8533 stream_.deviceFormat[mode] = stream_.userFormat;
8534 ss.format = sf->pa_format;
8538 if ( !sf_found ) { // Use internal data format conversion.
8539 stream_.userFormat = format;
8540 stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
8541 ss.format = PA_SAMPLE_FLOAT32LE;
8544 // Set other stream parameters.
8545 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
8546 else stream_.userInterleaved = true;
8547 stream_.deviceInterleaved[mode] = true;
8548 stream_.nBuffers = 1;
8549 stream_.doByteSwap[mode] = false;
8550 stream_.nUserChannels[mode] = channels;
8551 stream_.nDeviceChannels[mode] = channels + firstChannel;
8552 stream_.channelOffset[mode] = 0;
8553 std::string streamName = "RtAudio";
8555 // Set flags for buffer conversion.
8556 stream_.doConvertBuffer[mode] = false;
8557 if ( stream_.userFormat != stream_.deviceFormat[mode] )
8558 stream_.doConvertBuffer[mode] = true;
8559 if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
8560 stream_.doConvertBuffer[mode] = true;
8562 // Allocate necessary internal buffers.
8563 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
8564 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
8565 if ( stream_.userBuffer[mode] == NULL ) {
8566 errorText_ = "RtApiPulse::probeDeviceOpen: error allocating user buffer memory.";
8569 stream_.bufferSize = *bufferSize;
8571 if ( stream_.doConvertBuffer[mode] ) {
8573 bool makeBuffer = true;
8574 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
8575 if ( mode == INPUT ) {
8576 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
8577 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
8578 if ( bufferBytes <= bytesOut ) makeBuffer = false;
8583 bufferBytes *= *bufferSize;
8584 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
8585 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
8586 if ( stream_.deviceBuffer == NULL ) {
8587 errorText_ = "RtApiPulse::probeDeviceOpen: error allocating device buffer memory.";
8593 stream_.device[mode] = device;
8595 // Setup the buffer conversion information structure.
8596 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
8598 if ( !stream_.apiHandle ) {
8599 PulseAudioHandle *pah = new PulseAudioHandle;
8601 errorText_ = "RtApiPulse::probeDeviceOpen: error allocating memory for handle.";
8605 stream_.apiHandle = pah;
8606 if ( pthread_cond_init( &pah->runnable_cv, NULL ) != 0 ) {
8607 errorText_ = "RtApiPulse::probeDeviceOpen: error creating condition variable.";
8611 pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8614 if ( options && !options->streamName.empty() ) streamName = options->streamName;
8617 pa_buffer_attr buffer_attr;
8618 buffer_attr.fragsize = bufferBytes;
8619 buffer_attr.maxlength = -1;
8621 pah->s_rec = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_RECORD, NULL, "Record", &ss, NULL, &buffer_attr, &error );
8622 if ( !pah->s_rec ) {
8623 errorText_ = "RtApiPulse::probeDeviceOpen: error connecting input to PulseAudio server.";
8628 pah->s_play = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_PLAYBACK, NULL, "Playback", &ss, NULL, NULL, &error );
8629 if ( !pah->s_play ) {
8630 errorText_ = "RtApiPulse::probeDeviceOpen: error connecting output to PulseAudio server.";
8638 if ( stream_.mode == UNINITIALIZED )
8639 stream_.mode = mode;
8640 else if ( stream_.mode == mode )
8643 stream_.mode = DUPLEX;
8645 if ( !stream_.callbackInfo.isRunning ) {
8646 stream_.callbackInfo.object = this;
8647 stream_.callbackInfo.isRunning = true;
8648 if ( pthread_create( &pah->thread, NULL, pulseaudio_callback, (void *)&stream_.callbackInfo) != 0 ) {
8649 errorText_ = "RtApiPulse::probeDeviceOpen: error creating thread.";
8654 stream_.state = STREAM_STOPPED;
8658 if ( pah && stream_.callbackInfo.isRunning ) {
8659 pthread_cond_destroy( &pah->runnable_cv );
8661 stream_.apiHandle = 0;
8664 for ( int i=0; i<2; i++ ) {
8665 if ( stream_.userBuffer[i] ) {
8666 free( stream_.userBuffer[i] );
8667 stream_.userBuffer[i] = 0;
8671 if ( stream_.deviceBuffer ) {
8672 free( stream_.deviceBuffer );
8673 stream_.deviceBuffer = 0;
8679 //******************** End of __LINUX_PULSE__ *********************//
8682 #if defined(__LINUX_OSS__)
8685 #include <sys/ioctl.h>
8688 #include <sys/soundcard.h>
8692 static void *ossCallbackHandler(void * ptr);
8694 // A structure to hold various information related to the OSS API
8697 int id[2]; // device ids
8700 pthread_cond_t runnable;
8703 :triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
8706 RtApiOss :: RtApiOss()
8708 // Nothing to do here.
8711 RtApiOss :: ~RtApiOss()
8713 if ( stream_.state != STREAM_CLOSED ) closeStream();
8716 unsigned int RtApiOss :: getDeviceCount( void )
8718 int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
8719 if ( mixerfd == -1 ) {
8720 errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'.";
8721 error( RtAudioError::WARNING );
8725 oss_sysinfo sysinfo;
8726 if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) {
8728 errorText_ = "RtApiOss::getDeviceCount: error getting sysinfo, OSS version >= 4.0 is required.";
8729 error( RtAudioError::WARNING );
8734 return sysinfo.numaudios;
8737 RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device )
8739 RtAudio::DeviceInfo info;
8740 info.probed = false;
8742 int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
8743 if ( mixerfd == -1 ) {
8744 errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'.";
8745 error( RtAudioError::WARNING );
8749 oss_sysinfo sysinfo;
8750 int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
8751 if ( result == -1 ) {
8753 errorText_ = "RtApiOss::getDeviceInfo: error getting sysinfo, OSS version >= 4.0 is required.";
8754 error( RtAudioError::WARNING );
8758 unsigned nDevices = sysinfo.numaudios;
8759 if ( nDevices == 0 ) {
8761 errorText_ = "RtApiOss::getDeviceInfo: no devices found!";
8762 error( RtAudioError::INVALID_USE );
8766 if ( device >= nDevices ) {
8768 errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!";
8769 error( RtAudioError::INVALID_USE );
8773 oss_audioinfo ainfo;
8775 result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
8777 if ( result == -1 ) {
8778 errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
8779 errorText_ = errorStream_.str();
8780 error( RtAudioError::WARNING );
8785 if ( ainfo.caps & PCM_CAP_OUTPUT ) info.outputChannels = ainfo.max_channels;
8786 if ( ainfo.caps & PCM_CAP_INPUT ) info.inputChannels = ainfo.max_channels;
8787 if ( ainfo.caps & PCM_CAP_DUPLEX ) {
8788 if ( info.outputChannels > 0 && info.inputChannels > 0 && ainfo.caps & PCM_CAP_DUPLEX )
8789 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
8792 // Probe data formats ... do for input
8793 unsigned long mask = ainfo.iformats;
8794 if ( mask & AFMT_S16_LE || mask & AFMT_S16_BE )
8795 info.nativeFormats |= RTAUDIO_SINT16;
8796 if ( mask & AFMT_S8 )
8797 info.nativeFormats |= RTAUDIO_SINT8;
8798 if ( mask & AFMT_S32_LE || mask & AFMT_S32_BE )
8799 info.nativeFormats |= RTAUDIO_SINT32;
8801 if ( mask & AFMT_FLOAT )
8802 info.nativeFormats |= RTAUDIO_FLOAT32;
8804 if ( mask & AFMT_S24_LE || mask & AFMT_S24_BE )
8805 info.nativeFormats |= RTAUDIO_SINT24;
8807 // Check that we have at least one supported format
8808 if ( info.nativeFormats == 0 ) {
8809 errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio.";
8810 errorText_ = errorStream_.str();
8811 error( RtAudioError::WARNING );
8815 // Probe the supported sample rates.
8816 info.sampleRates.clear();
8817 if ( ainfo.nrates ) {
8818 for ( unsigned int i=0; i<ainfo.nrates; i++ ) {
8819 for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
8820 if ( ainfo.rates[i] == SAMPLE_RATES[k] ) {
8821 info.sampleRates.push_back( SAMPLE_RATES[k] );
8823 if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
8824 info.preferredSampleRate = SAMPLE_RATES[k];
8832 // Check min and max rate values;
8833 for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
8834 if ( ainfo.min_rate <= (int) SAMPLE_RATES[k] && ainfo.max_rate >= (int) SAMPLE_RATES[k] ) {
8835 info.sampleRates.push_back( SAMPLE_RATES[k] );
8837 if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
8838 info.preferredSampleRate = SAMPLE_RATES[k];
8843 if ( info.sampleRates.size() == 0 ) {
8844 errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ").";
8845 errorText_ = errorStream_.str();
8846 error( RtAudioError::WARNING );
8850 info.name = ainfo.name;
8857 bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
8858 unsigned int firstChannel, unsigned int sampleRate,
8859 RtAudioFormat format, unsigned int *bufferSize,
8860 RtAudio::StreamOptions *options )
8862 int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
8863 if ( mixerfd == -1 ) {
8864 errorText_ = "RtApiOss::probeDeviceOpen: error opening '/dev/mixer'.";
8868 oss_sysinfo sysinfo;
8869 int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
8870 if ( result == -1 ) {
8872 errorText_ = "RtApiOss::probeDeviceOpen: error getting sysinfo, OSS version >= 4.0 is required.";
8876 unsigned nDevices = sysinfo.numaudios;
8877 if ( nDevices == 0 ) {
8878 // This should not happen because a check is made before this function is called.
8880 errorText_ = "RtApiOss::probeDeviceOpen: no devices found!";
8884 if ( device >= nDevices ) {
8885 // This should not happen because a check is made before this function is called.
8887 errorText_ = "RtApiOss::probeDeviceOpen: device ID is invalid!";
8891 oss_audioinfo ainfo;
8893 result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
8895 if ( result == -1 ) {
8896 errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
8897 errorText_ = errorStream_.str();
8901 // Check if device supports input or output
8902 if ( ( mode == OUTPUT && !( ainfo.caps & PCM_CAP_OUTPUT ) ) ||
8903 ( mode == INPUT && !( ainfo.caps & PCM_CAP_INPUT ) ) ) {
8904 if ( mode == OUTPUT )
8905 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support output.";
8907 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support input.";
8908 errorText_ = errorStream_.str();
8913 OssHandle *handle = (OssHandle *) stream_.apiHandle;
8914 if ( mode == OUTPUT )
8916 else { // mode == INPUT
8917 if (stream_.mode == OUTPUT && stream_.device[0] == device) {
8918 // We just set the same device for playback ... close and reopen for duplex (OSS only).
8919 close( handle->id[0] );
8921 if ( !( ainfo.caps & PCM_CAP_DUPLEX ) ) {
8922 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support duplex mode.";
8923 errorText_ = errorStream_.str();
8926 // Check that the number previously set channels is the same.
8927 if ( stream_.nUserChannels[0] != channels ) {
8928 errorStream_ << "RtApiOss::probeDeviceOpen: input/output channels must be equal for OSS duplex device (" << ainfo.name << ").";
8929 errorText_ = errorStream_.str();
8938 // Set exclusive access if specified.
8939 if ( options && options->flags & RTAUDIO_HOG_DEVICE ) flags |= O_EXCL;
8941 // Try to open the device.
8943 fd = open( ainfo.devnode, flags, 0 );
8945 if ( errno == EBUSY )
8946 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") is busy.";
8948 errorStream_ << "RtApiOss::probeDeviceOpen: error opening device (" << ainfo.name << ").";
8949 errorText_ = errorStream_.str();
8953 // For duplex operation, specifically set this mode (this doesn't seem to work).
8955 if ( flags | O_RDWR ) {
8956 result = ioctl( fd, SNDCTL_DSP_SETDUPLEX, NULL );
8957 if ( result == -1) {
8958 errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ").";
8959 errorText_ = errorStream_.str();
8965 // Check the device channel support.
8966 stream_.nUserChannels[mode] = channels;
8967 if ( ainfo.max_channels < (int)(channels + firstChannel) ) {
8969 errorStream_ << "RtApiOss::probeDeviceOpen: the device (" << ainfo.name << ") does not support requested channel parameters.";
8970 errorText_ = errorStream_.str();
8974 // Set the number of channels.
8975 int deviceChannels = channels + firstChannel;
8976 result = ioctl( fd, SNDCTL_DSP_CHANNELS, &deviceChannels );
8977 if ( result == -1 || deviceChannels < (int)(channels + firstChannel) ) {
8979 errorStream_ << "RtApiOss::probeDeviceOpen: error setting channel parameters on device (" << ainfo.name << ").";
8980 errorText_ = errorStream_.str();
8983 stream_.nDeviceChannels[mode] = deviceChannels;
8985 // Get the data format mask
8987 result = ioctl( fd, SNDCTL_DSP_GETFMTS, &mask );
8988 if ( result == -1 ) {
8990 errorStream_ << "RtApiOss::probeDeviceOpen: error getting device (" << ainfo.name << ") data formats.";
8991 errorText_ = errorStream_.str();
8995 // Determine how to set the device format.
8996 stream_.userFormat = format;
8997 int deviceFormat = -1;
8998 stream_.doByteSwap[mode] = false;
8999 if ( format == RTAUDIO_SINT8 ) {
9000 if ( mask & AFMT_S8 ) {
9001 deviceFormat = AFMT_S8;
9002 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
9005 else if ( format == RTAUDIO_SINT16 ) {
9006 if ( mask & AFMT_S16_NE ) {
9007 deviceFormat = AFMT_S16_NE;
9008 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
9010 else if ( mask & AFMT_S16_OE ) {
9011 deviceFormat = AFMT_S16_OE;
9012 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
9013 stream_.doByteSwap[mode] = true;
9016 else if ( format == RTAUDIO_SINT24 ) {
9017 if ( mask & AFMT_S24_NE ) {
9018 deviceFormat = AFMT_S24_NE;
9019 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
9021 else if ( mask & AFMT_S24_OE ) {
9022 deviceFormat = AFMT_S24_OE;
9023 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
9024 stream_.doByteSwap[mode] = true;
9027 else if ( format == RTAUDIO_SINT32 ) {
9028 if ( mask & AFMT_S32_NE ) {
9029 deviceFormat = AFMT_S32_NE;
9030 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
9032 else if ( mask & AFMT_S32_OE ) {
9033 deviceFormat = AFMT_S32_OE;
9034 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
9035 stream_.doByteSwap[mode] = true;
9039 if ( deviceFormat == -1 ) {
9040 // The user requested format is not natively supported by the device.
9041 if ( mask & AFMT_S16_NE ) {
9042 deviceFormat = AFMT_S16_NE;
9043 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
9045 else if ( mask & AFMT_S32_NE ) {
9046 deviceFormat = AFMT_S32_NE;
9047 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
9049 else if ( mask & AFMT_S24_NE ) {
9050 deviceFormat = AFMT_S24_NE;
9051 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
9053 else if ( mask & AFMT_S16_OE ) {
9054 deviceFormat = AFMT_S16_OE;
9055 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
9056 stream_.doByteSwap[mode] = true;
9058 else if ( mask & AFMT_S32_OE ) {
9059 deviceFormat = AFMT_S32_OE;
9060 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
9061 stream_.doByteSwap[mode] = true;
9063 else if ( mask & AFMT_S24_OE ) {
9064 deviceFormat = AFMT_S24_OE;
9065 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
9066 stream_.doByteSwap[mode] = true;
9068 else if ( mask & AFMT_S8) {
9069 deviceFormat = AFMT_S8;
9070 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
9074 if ( stream_.deviceFormat[mode] == 0 ) {
9075 // This really shouldn't happen ...
9077 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") data format not supported by RtAudio.";
9078 errorText_ = errorStream_.str();
9082 // Set the data format.
9083 int temp = deviceFormat;
9084 result = ioctl( fd, SNDCTL_DSP_SETFMT, &deviceFormat );
9085 if ( result == -1 || deviceFormat != temp ) {
9087 errorStream_ << "RtApiOss::probeDeviceOpen: error setting data format on device (" << ainfo.name << ").";
9088 errorText_ = errorStream_.str();
9092 // Attempt to set the buffer size. According to OSS, the minimum
9093 // number of buffers is two. The supposed minimum buffer size is 16
9094 // bytes, so that will be our lower bound. The argument to this
9095 // call is in the form 0xMMMMSSSS (hex), where the buffer size (in
9096 // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM.
9097 // We'll check the actual value used near the end of the setup
9099 int ossBufferBytes = *bufferSize * formatBytes( stream_.deviceFormat[mode] ) * deviceChannels;
9100 if ( ossBufferBytes < 16 ) ossBufferBytes = 16;
9102 if ( options ) buffers = options->numberOfBuffers;
9103 if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) buffers = 2;
9104 if ( buffers < 2 ) buffers = 3;
9105 temp = ((int) buffers << 16) + (int)( log10( (double)ossBufferBytes ) / log10( 2.0 ) );
9106 result = ioctl( fd, SNDCTL_DSP_SETFRAGMENT, &temp );
9107 if ( result == -1 ) {
9109 errorStream_ << "RtApiOss::probeDeviceOpen: error setting buffer size on device (" << ainfo.name << ").";
9110 errorText_ = errorStream_.str();
9113 stream_.nBuffers = buffers;
9115 // Save buffer size (in sample frames).
9116 *bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels );
9117 stream_.bufferSize = *bufferSize;
9119 // Set the sample rate.
9120 int srate = sampleRate;
9121 result = ioctl( fd, SNDCTL_DSP_SPEED, &srate );
9122 if ( result == -1 ) {
9124 errorStream_ << "RtApiOss::probeDeviceOpen: error setting sample rate (" << sampleRate << ") on device (" << ainfo.name << ").";
9125 errorText_ = errorStream_.str();
9129 // Verify the sample rate setup worked.
9130 if ( abs( srate - (int)sampleRate ) > 100 ) {
9132 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ").";
9133 errorText_ = errorStream_.str();
9136 stream_.sampleRate = sampleRate;
9138 if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device) {
9139 // We're doing duplex setup here.
9140 stream_.deviceFormat[0] = stream_.deviceFormat[1];
9141 stream_.nDeviceChannels[0] = deviceChannels;
9144 // Set interleaving parameters.
9145 stream_.userInterleaved = true;
9146 stream_.deviceInterleaved[mode] = true;
9147 if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
9148 stream_.userInterleaved = false;
9150 // Set flags for buffer conversion
9151 stream_.doConvertBuffer[mode] = false;
9152 if ( stream_.userFormat != stream_.deviceFormat[mode] )
9153 stream_.doConvertBuffer[mode] = true;
9154 if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
9155 stream_.doConvertBuffer[mode] = true;
9156 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
9157 stream_.nUserChannels[mode] > 1 )
9158 stream_.doConvertBuffer[mode] = true;
9160 // Allocate the stream handles if necessary and then save.
9161 if ( stream_.apiHandle == 0 ) {
9163 handle = new OssHandle;
9165 catch ( std::bad_alloc& ) {
9166 errorText_ = "RtApiOss::probeDeviceOpen: error allocating OssHandle memory.";
9170 if ( pthread_cond_init( &handle->runnable, NULL ) ) {
9171 errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable.";
9175 stream_.apiHandle = (void *) handle;
9178 handle = (OssHandle *) stream_.apiHandle;
9180 handle->id[mode] = fd;
9182 // Allocate necessary internal buffers.
9183 unsigned long bufferBytes;
9184 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
9185 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
9186 if ( stream_.userBuffer[mode] == NULL ) {
9187 errorText_ = "RtApiOss::probeDeviceOpen: error allocating user buffer memory.";
9191 if ( stream_.doConvertBuffer[mode] ) {
9193 bool makeBuffer = true;
9194 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
9195 if ( mode == INPUT ) {
9196 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
9197 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
9198 if ( bufferBytes <= bytesOut ) makeBuffer = false;
9203 bufferBytes *= *bufferSize;
9204 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
9205 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
9206 if ( stream_.deviceBuffer == NULL ) {
9207 errorText_ = "RtApiOss::probeDeviceOpen: error allocating device buffer memory.";
9213 stream_.device[mode] = device;
9214 stream_.state = STREAM_STOPPED;
9216 // Setup the buffer conversion information structure.
9217 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
9219 // Setup thread if necessary.
9220 if ( stream_.mode == OUTPUT && mode == INPUT ) {
9221 // We had already set up an output stream.
9222 stream_.mode = DUPLEX;
9223 if ( stream_.device[0] == device ) handle->id[0] = fd;
9226 stream_.mode = mode;
9228 // Setup callback thread.
9229 stream_.callbackInfo.object = (void *) this;
9231 // Set the thread attributes for joinable and realtime scheduling
9232 // priority. The higher priority will only take affect if the
9233 // program is run as root or suid.
9234 pthread_attr_t attr;
9235 pthread_attr_init( &attr );
9236 pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
9237 #ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
9238 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
9239 struct sched_param param;
9240 int priority = options->priority;
9241 int min = sched_get_priority_min( SCHED_RR );
9242 int max = sched_get_priority_max( SCHED_RR );
9243 if ( priority < min ) priority = min;
9244 else if ( priority > max ) priority = max;
9245 param.sched_priority = priority;
9246 pthread_attr_setschedparam( &attr, ¶m );
9247 pthread_attr_setschedpolicy( &attr, SCHED_RR );
9250 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
9252 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
9255 stream_.callbackInfo.isRunning = true;
9256 result = pthread_create( &stream_.callbackInfo.thread, &attr, ossCallbackHandler, &stream_.callbackInfo );
9257 pthread_attr_destroy( &attr );
9259 stream_.callbackInfo.isRunning = false;
9260 errorText_ = "RtApiOss::error creating callback thread!";
9269 pthread_cond_destroy( &handle->runnable );
9270 if ( handle->id[0] ) close( handle->id[0] );
9271 if ( handle->id[1] ) close( handle->id[1] );
9273 stream_.apiHandle = 0;
9276 for ( int i=0; i<2; i++ ) {
9277 if ( stream_.userBuffer[i] ) {
9278 free( stream_.userBuffer[i] );
9279 stream_.userBuffer[i] = 0;
9283 if ( stream_.deviceBuffer ) {
9284 free( stream_.deviceBuffer );
9285 stream_.deviceBuffer = 0;
9291 void RtApiOss :: closeStream()
9293 if ( stream_.state == STREAM_CLOSED ) {
9294 errorText_ = "RtApiOss::closeStream(): no open stream to close!";
9295 error( RtAudioError::WARNING );
9299 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9300 stream_.callbackInfo.isRunning = false;
9301 MUTEX_LOCK( &stream_.mutex );
9302 if ( stream_.state == STREAM_STOPPED )
9303 pthread_cond_signal( &handle->runnable );
9304 MUTEX_UNLOCK( &stream_.mutex );
9305 pthread_join( stream_.callbackInfo.thread, NULL );
9307 if ( stream_.state == STREAM_RUNNING ) {
9308 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
9309 ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
9311 ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
9312 stream_.state = STREAM_STOPPED;
9316 pthread_cond_destroy( &handle->runnable );
9317 if ( handle->id[0] ) close( handle->id[0] );
9318 if ( handle->id[1] ) close( handle->id[1] );
9320 stream_.apiHandle = 0;
9323 for ( int i=0; i<2; i++ ) {
9324 if ( stream_.userBuffer[i] ) {
9325 free( stream_.userBuffer[i] );
9326 stream_.userBuffer[i] = 0;
9330 if ( stream_.deviceBuffer ) {
9331 free( stream_.deviceBuffer );
9332 stream_.deviceBuffer = 0;
9335 stream_.mode = UNINITIALIZED;
9336 stream_.state = STREAM_CLOSED;
9339 void RtApiOss :: startStream()
9342 if ( stream_.state == STREAM_RUNNING ) {
9343 errorText_ = "RtApiOss::startStream(): the stream is already running!";
9344 error( RtAudioError::WARNING );
9348 MUTEX_LOCK( &stream_.mutex );
9350 stream_.state = STREAM_RUNNING;
9352 // No need to do anything else here ... OSS automatically starts
9353 // when fed samples.
9355 MUTEX_UNLOCK( &stream_.mutex );
9357 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9358 pthread_cond_signal( &handle->runnable );
9361 void RtApiOss :: stopStream()
9364 if ( stream_.state == STREAM_STOPPED ) {
9365 errorText_ = "RtApiOss::stopStream(): the stream is already stopped!";
9366 error( RtAudioError::WARNING );
9370 MUTEX_LOCK( &stream_.mutex );
9372 // The state might change while waiting on a mutex.
9373 if ( stream_.state == STREAM_STOPPED ) {
9374 MUTEX_UNLOCK( &stream_.mutex );
9379 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9380 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
9382 // Flush the output with zeros a few times.
9385 RtAudioFormat format;
9387 if ( stream_.doConvertBuffer[0] ) {
9388 buffer = stream_.deviceBuffer;
9389 samples = stream_.bufferSize * stream_.nDeviceChannels[0];
9390 format = stream_.deviceFormat[0];
9393 buffer = stream_.userBuffer[0];
9394 samples = stream_.bufferSize * stream_.nUserChannels[0];
9395 format = stream_.userFormat;
9398 memset( buffer, 0, samples * formatBytes(format) );
9399 for ( unsigned int i=0; i<stream_.nBuffers+1; i++ ) {
9400 result = write( handle->id[0], buffer, samples * formatBytes(format) );
9401 if ( result == -1 ) {
9402 errorText_ = "RtApiOss::stopStream: audio write error.";
9403 error( RtAudioError::WARNING );
9407 result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
9408 if ( result == -1 ) {
9409 errorStream_ << "RtApiOss::stopStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
9410 errorText_ = errorStream_.str();
9413 handle->triggered = false;
9416 if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
9417 result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
9418 if ( result == -1 ) {
9419 errorStream_ << "RtApiOss::stopStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
9420 errorText_ = errorStream_.str();
9426 stream_.state = STREAM_STOPPED;
9427 MUTEX_UNLOCK( &stream_.mutex );
9429 if ( result != -1 ) return;
9430 error( RtAudioError::SYSTEM_ERROR );
9433 void RtApiOss :: abortStream()
9436 if ( stream_.state == STREAM_STOPPED ) {
9437 errorText_ = "RtApiOss::abortStream(): the stream is already stopped!";
9438 error( RtAudioError::WARNING );
9442 MUTEX_LOCK( &stream_.mutex );
9444 // The state might change while waiting on a mutex.
9445 if ( stream_.state == STREAM_STOPPED ) {
9446 MUTEX_UNLOCK( &stream_.mutex );
9451 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9452 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
9453 result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
9454 if ( result == -1 ) {
9455 errorStream_ << "RtApiOss::abortStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
9456 errorText_ = errorStream_.str();
9459 handle->triggered = false;
9462 if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
9463 result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
9464 if ( result == -1 ) {
9465 errorStream_ << "RtApiOss::abortStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
9466 errorText_ = errorStream_.str();
9472 stream_.state = STREAM_STOPPED;
9473 MUTEX_UNLOCK( &stream_.mutex );
9475 if ( result != -1 ) return;
9476 error( RtAudioError::SYSTEM_ERROR );
9479 void RtApiOss :: callbackEvent()
9481 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9482 if ( stream_.state == STREAM_STOPPED ) {
9483 MUTEX_LOCK( &stream_.mutex );
9484 pthread_cond_wait( &handle->runnable, &stream_.mutex );
9485 if ( stream_.state != STREAM_RUNNING ) {
9486 MUTEX_UNLOCK( &stream_.mutex );
9489 MUTEX_UNLOCK( &stream_.mutex );
9492 if ( stream_.state == STREAM_CLOSED ) {
9493 errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!";
9494 error( RtAudioError::WARNING );
9498 // Invoke user callback to get fresh output data.
9499 int doStopStream = 0;
9500 RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
9501 double streamTime = getStreamTime();
9502 RtAudioStreamStatus status = 0;
9503 if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
9504 status |= RTAUDIO_OUTPUT_UNDERFLOW;
9505 handle->xrun[0] = false;
9507 if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
9508 status |= RTAUDIO_INPUT_OVERFLOW;
9509 handle->xrun[1] = false;
9511 doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
9512 stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
9513 if ( doStopStream == 2 ) {
9514 this->abortStream();
9518 MUTEX_LOCK( &stream_.mutex );
9520 // The state might change while waiting on a mutex.
9521 if ( stream_.state == STREAM_STOPPED ) goto unlock;
9526 RtAudioFormat format;
9528 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
9530 // Setup parameters and do buffer conversion if necessary.
9531 if ( stream_.doConvertBuffer[0] ) {
9532 buffer = stream_.deviceBuffer;
9533 convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
9534 samples = stream_.bufferSize * stream_.nDeviceChannels[0];
9535 format = stream_.deviceFormat[0];
9538 buffer = stream_.userBuffer[0];
9539 samples = stream_.bufferSize * stream_.nUserChannels[0];
9540 format = stream_.userFormat;
9543 // Do byte swapping if necessary.
9544 if ( stream_.doByteSwap[0] )
9545 byteSwapBuffer( buffer, samples, format );
9547 if ( stream_.mode == DUPLEX && handle->triggered == false ) {
9549 ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
9550 result = write( handle->id[0], buffer, samples * formatBytes(format) );
9551 trig = PCM_ENABLE_INPUT|PCM_ENABLE_OUTPUT;
9552 ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
9553 handle->triggered = true;
9556 // Write samples to device.
9557 result = write( handle->id[0], buffer, samples * formatBytes(format) );
9559 if ( result == -1 ) {
9560 // We'll assume this is an underrun, though there isn't a
9561 // specific means for determining that.
9562 handle->xrun[0] = true;
9563 errorText_ = "RtApiOss::callbackEvent: audio write error.";
9564 error( RtAudioError::WARNING );
9565 // Continue on to input section.
9569 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
9571 // Setup parameters.
9572 if ( stream_.doConvertBuffer[1] ) {
9573 buffer = stream_.deviceBuffer;
9574 samples = stream_.bufferSize * stream_.nDeviceChannels[1];
9575 format = stream_.deviceFormat[1];
9578 buffer = stream_.userBuffer[1];
9579 samples = stream_.bufferSize * stream_.nUserChannels[1];
9580 format = stream_.userFormat;
9583 // Read samples from device.
9584 result = read( handle->id[1], buffer, samples * formatBytes(format) );
9586 if ( result == -1 ) {
9587 // We'll assume this is an overrun, though there isn't a
9588 // specific means for determining that.
9589 handle->xrun[1] = true;
9590 errorText_ = "RtApiOss::callbackEvent: audio read error.";
9591 error( RtAudioError::WARNING );
9595 // Do byte swapping if necessary.
9596 if ( stream_.doByteSwap[1] )
9597 byteSwapBuffer( buffer, samples, format );
9599 // Do buffer conversion if necessary.
9600 if ( stream_.doConvertBuffer[1] )
9601 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
9605 MUTEX_UNLOCK( &stream_.mutex );
9607 RtApi::tickStreamTime();
9608 if ( doStopStream == 1 ) this->stopStream();
9611 static void *ossCallbackHandler( void *ptr )
9613 CallbackInfo *info = (CallbackInfo *) ptr;
9614 RtApiOss *object = (RtApiOss *) info->object;
9615 bool *isRunning = &info->isRunning;
9617 while ( *isRunning == true ) {
9618 pthread_testcancel();
9619 object->callbackEvent();
9622 pthread_exit( NULL );
9625 //******************** End of __LINUX_OSS__ *********************//
9629 // *************************************************** //
9631 // Protected common (OS-independent) RtAudio methods.
9633 // *************************************************** //
9635 // This method can be modified to control the behavior of error
9636 // message printing.
9637 void RtApi :: error( RtAudioError::Type type )
9639 errorStream_.str(""); // clear the ostringstream
9641 RtAudioErrorCallback errorCallback = (RtAudioErrorCallback) stream_.callbackInfo.errorCallback;
9642 if ( errorCallback ) {
9643 // abortStream() can generate new error messages. Ignore them. Just keep original one.
9645 if ( firstErrorOccurred_ )
9648 firstErrorOccurred_ = true;
9649 const std::string errorMessage = errorText_;
9651 if ( type != RtAudioError::WARNING && stream_.state != STREAM_STOPPED) {
9652 stream_.callbackInfo.isRunning = false; // exit from the thread
9656 errorCallback( type, errorMessage );
9657 firstErrorOccurred_ = false;
9661 if ( type == RtAudioError::WARNING && showWarnings_ == true )
9662 std::cerr << '\n' << errorText_ << "\n\n";
9663 else if ( type != RtAudioError::WARNING )
9664 throw( RtAudioError( errorText_, type ) );
9667 void RtApi :: verifyStream()
9669 if ( stream_.state == STREAM_CLOSED ) {
9670 errorText_ = "RtApi:: a stream is not open!";
9671 error( RtAudioError::INVALID_USE );
9675 void RtApi :: clearStreamInfo()
9677 stream_.mode = UNINITIALIZED;
9678 stream_.state = STREAM_CLOSED;
9679 stream_.sampleRate = 0;
9680 stream_.bufferSize = 0;
9681 stream_.nBuffers = 0;
9682 stream_.userFormat = 0;
9683 stream_.userInterleaved = true;
9684 stream_.streamTime = 0.0;
9685 stream_.apiHandle = 0;
9686 stream_.deviceBuffer = 0;
9687 stream_.callbackInfo.callback = 0;
9688 stream_.callbackInfo.userData = 0;
9689 stream_.callbackInfo.isRunning = false;
9690 stream_.callbackInfo.errorCallback = 0;
9691 for ( int i=0; i<2; i++ ) {
9692 stream_.device[i] = 11111;
9693 stream_.doConvertBuffer[i] = false;
9694 stream_.deviceInterleaved[i] = true;
9695 stream_.doByteSwap[i] = false;
9696 stream_.nUserChannels[i] = 0;
9697 stream_.nDeviceChannels[i] = 0;
9698 stream_.channelOffset[i] = 0;
9699 stream_.deviceFormat[i] = 0;
9700 stream_.latency[i] = 0;
9701 stream_.userBuffer[i] = 0;
9702 stream_.convertInfo[i].channels = 0;
9703 stream_.convertInfo[i].inJump = 0;
9704 stream_.convertInfo[i].outJump = 0;
9705 stream_.convertInfo[i].inFormat = 0;
9706 stream_.convertInfo[i].outFormat = 0;
9707 stream_.convertInfo[i].inOffset.clear();
9708 stream_.convertInfo[i].outOffset.clear();
9712 unsigned int RtApi :: formatBytes( RtAudioFormat format )
9714 if ( format == RTAUDIO_SINT16 )
9716 else if ( format == RTAUDIO_SINT32 || format == RTAUDIO_FLOAT32 )
9718 else if ( format == RTAUDIO_FLOAT64 )
9720 else if ( format == RTAUDIO_SINT24 )
9722 else if ( format == RTAUDIO_SINT8 )
9725 errorText_ = "RtApi::formatBytes: undefined format.";
9726 error( RtAudioError::WARNING );
9731 void RtApi :: setConvertInfo( StreamMode mode, unsigned int firstChannel )
9733 if ( mode == INPUT ) { // convert device to user buffer
9734 stream_.convertInfo[mode].inJump = stream_.nDeviceChannels[1];
9735 stream_.convertInfo[mode].outJump = stream_.nUserChannels[1];
9736 stream_.convertInfo[mode].inFormat = stream_.deviceFormat[1];
9737 stream_.convertInfo[mode].outFormat = stream_.userFormat;
9739 else { // convert user to device buffer
9740 stream_.convertInfo[mode].inJump = stream_.nUserChannels[0];
9741 stream_.convertInfo[mode].outJump = stream_.nDeviceChannels[0];
9742 stream_.convertInfo[mode].inFormat = stream_.userFormat;
9743 stream_.convertInfo[mode].outFormat = stream_.deviceFormat[0];
9746 if ( stream_.convertInfo[mode].inJump < stream_.convertInfo[mode].outJump )
9747 stream_.convertInfo[mode].channels = stream_.convertInfo[mode].inJump;
9749 stream_.convertInfo[mode].channels = stream_.convertInfo[mode].outJump;
9751 // Set up the interleave/deinterleave offsets.
9752 if ( stream_.deviceInterleaved[mode] != stream_.userInterleaved ) {
9753 if ( ( mode == OUTPUT && stream_.deviceInterleaved[mode] ) ||
9754 ( mode == INPUT && stream_.userInterleaved ) ) {
9755 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
9756 stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
9757 stream_.convertInfo[mode].outOffset.push_back( k );
9758 stream_.convertInfo[mode].inJump = 1;
9762 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
9763 stream_.convertInfo[mode].inOffset.push_back( k );
9764 stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
9765 stream_.convertInfo[mode].outJump = 1;
9769 else { // no (de)interleaving
9770 if ( stream_.userInterleaved ) {
9771 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
9772 stream_.convertInfo[mode].inOffset.push_back( k );
9773 stream_.convertInfo[mode].outOffset.push_back( k );
9777 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
9778 stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
9779 stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
9780 stream_.convertInfo[mode].inJump = 1;
9781 stream_.convertInfo[mode].outJump = 1;
9786 // Add channel offset.
9787 if ( firstChannel > 0 ) {
9788 if ( stream_.deviceInterleaved[mode] ) {
9789 if ( mode == OUTPUT ) {
9790 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
9791 stream_.convertInfo[mode].outOffset[k] += firstChannel;
9794 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
9795 stream_.convertInfo[mode].inOffset[k] += firstChannel;
9799 if ( mode == OUTPUT ) {
9800 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
9801 stream_.convertInfo[mode].outOffset[k] += ( firstChannel * stream_.bufferSize );
9804 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
9805 stream_.convertInfo[mode].inOffset[k] += ( firstChannel * stream_.bufferSize );
9811 void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info )
9813 // This function does format conversion, input/output channel compensation, and
9814 // data interleaving/deinterleaving. 24-bit integers are assumed to occupy
9815 // the lower three bytes of a 32-bit integer.
9817 // Clear our device buffer when in/out duplex device channels are different
9818 if ( outBuffer == stream_.deviceBuffer && stream_.mode == DUPLEX &&
9819 ( stream_.nDeviceChannels[0] < stream_.nDeviceChannels[1] ) )
9820 memset( outBuffer, 0, stream_.bufferSize * info.outJump * formatBytes( info.outFormat ) );
9823 if (info.outFormat == RTAUDIO_FLOAT64) {
9825 Float64 *out = (Float64 *)outBuffer;
9827 if (info.inFormat == RTAUDIO_SINT8) {
9828 signed char *in = (signed char *)inBuffer;
9829 scale = 1.0 / 127.5;
9830 for (unsigned int i=0; i<stream_.bufferSize; i++) {
9831 for (j=0; j<info.channels; j++) {
9832 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
9833 out[info.outOffset[j]] += 0.5;
9834 out[info.outOffset[j]] *= scale;
9837 out += info.outJump;
9840 else if (info.inFormat == RTAUDIO_SINT16) {
9841 Int16 *in = (Int16 *)inBuffer;
9842 scale = 1.0 / 32767.5;
9843 for (unsigned int i=0; i<stream_.bufferSize; i++) {
9844 for (j=0; j<info.channels; j++) {
9845 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
9846 out[info.outOffset[j]] += 0.5;
9847 out[info.outOffset[j]] *= scale;
9850 out += info.outJump;
9853 else if (info.inFormat == RTAUDIO_SINT24) {
9854 Int24 *in = (Int24 *)inBuffer;
9855 scale = 1.0 / 8388607.5;
9856 for (unsigned int i=0; i<stream_.bufferSize; i++) {
9857 for (j=0; j<info.channels; j++) {
9858 out[info.outOffset[j]] = (Float64) (in[info.inOffset[j]].asInt());
9859 out[info.outOffset[j]] += 0.5;
9860 out[info.outOffset[j]] *= scale;
9863 out += info.outJump;
9866 else if (info.inFormat == RTAUDIO_SINT32) {
9867 Int32 *in = (Int32 *)inBuffer;
9868 scale = 1.0 / 2147483647.5;
9869 for (unsigned int i=0; i<stream_.bufferSize; i++) {
9870 for (j=0; j<info.channels; j++) {
9871 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
9872 out[info.outOffset[j]] += 0.5;
9873 out[info.outOffset[j]] *= scale;
9876 out += info.outJump;
9879 else if (info.inFormat == RTAUDIO_FLOAT32) {
9880 Float32 *in = (Float32 *)inBuffer;
9881 for (unsigned int i=0; i<stream_.bufferSize; i++) {
9882 for (j=0; j<info.channels; j++) {
9883 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
9886 out += info.outJump;
9889 else if (info.inFormat == RTAUDIO_FLOAT64) {
9890 // Channel compensation and/or (de)interleaving only.
9891 Float64 *in = (Float64 *)inBuffer;
9892 for (unsigned int i=0; i<stream_.bufferSize; i++) {
9893 for (j=0; j<info.channels; j++) {
9894 out[info.outOffset[j]] = in[info.inOffset[j]];
9897 out += info.outJump;
9901 else if (info.outFormat == RTAUDIO_FLOAT32) {
9903 Float32 *out = (Float32 *)outBuffer;
9905 if (info.inFormat == RTAUDIO_SINT8) {
9906 signed char *in = (signed char *)inBuffer;
9907 scale = (Float32) ( 1.0 / 127.5 );
9908 for (unsigned int i=0; i<stream_.bufferSize; i++) {
9909 for (j=0; j<info.channels; j++) {
9910 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
9911 out[info.outOffset[j]] += 0.5;
9912 out[info.outOffset[j]] *= scale;
9915 out += info.outJump;
9918 else if (info.inFormat == RTAUDIO_SINT16) {
9919 Int16 *in = (Int16 *)inBuffer;
9920 scale = (Float32) ( 1.0 / 32767.5 );
9921 for (unsigned int i=0; i<stream_.bufferSize; i++) {
9922 for (j=0; j<info.channels; j++) {
9923 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
9924 out[info.outOffset[j]] += 0.5;
9925 out[info.outOffset[j]] *= scale;
9928 out += info.outJump;
9931 else if (info.inFormat == RTAUDIO_SINT24) {
9932 Int24 *in = (Int24 *)inBuffer;
9933 scale = (Float32) ( 1.0 / 8388607.5 );
9934 for (unsigned int i=0; i<stream_.bufferSize; i++) {
9935 for (j=0; j<info.channels; j++) {
9936 out[info.outOffset[j]] = (Float32) (in[info.inOffset[j]].asInt());
9937 out[info.outOffset[j]] += 0.5;
9938 out[info.outOffset[j]] *= scale;
9941 out += info.outJump;
9944 else if (info.inFormat == RTAUDIO_SINT32) {
9945 Int32 *in = (Int32 *)inBuffer;
9946 scale = (Float32) ( 1.0 / 2147483647.5 );
9947 for (unsigned int i=0; i<stream_.bufferSize; i++) {
9948 for (j=0; j<info.channels; j++) {
9949 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
9950 out[info.outOffset[j]] += 0.5;
9951 out[info.outOffset[j]] *= scale;
9954 out += info.outJump;
9957 else if (info.inFormat == RTAUDIO_FLOAT32) {
9958 // Channel compensation and/or (de)interleaving only.
9959 Float32 *in = (Float32 *)inBuffer;
9960 for (unsigned int i=0; i<stream_.bufferSize; i++) {
9961 for (j=0; j<info.channels; j++) {
9962 out[info.outOffset[j]] = in[info.inOffset[j]];
9965 out += info.outJump;
9968 else if (info.inFormat == RTAUDIO_FLOAT64) {
9969 Float64 *in = (Float64 *)inBuffer;
9970 for (unsigned int i=0; i<stream_.bufferSize; i++) {
9971 for (j=0; j<info.channels; j++) {
9972 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
9975 out += info.outJump;
9979 else if (info.outFormat == RTAUDIO_SINT32) {
9980 Int32 *out = (Int32 *)outBuffer;
9981 if (info.inFormat == RTAUDIO_SINT8) {
9982 signed char *in = (signed char *)inBuffer;
9983 for (unsigned int i=0; i<stream_.bufferSize; i++) {
9984 for (j=0; j<info.channels; j++) {
9985 out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
9986 out[info.outOffset[j]] <<= 24;
9989 out += info.outJump;
9992 else if (info.inFormat == RTAUDIO_SINT16) {
9993 Int16 *in = (Int16 *)inBuffer;
9994 for (unsigned int i=0; i<stream_.bufferSize; i++) {
9995 for (j=0; j<info.channels; j++) {
9996 out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
9997 out[info.outOffset[j]] <<= 16;
10000 out += info.outJump;
10003 else if (info.inFormat == RTAUDIO_SINT24) {
10004 Int24 *in = (Int24 *)inBuffer;
10005 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10006 for (j=0; j<info.channels; j++) {
10007 out[info.outOffset[j]] = (Int32) in[info.inOffset[j]].asInt();
10008 out[info.outOffset[j]] <<= 8;
10011 out += info.outJump;
10014 else if (info.inFormat == RTAUDIO_SINT32) {
10015 // Channel compensation and/or (de)interleaving only.
10016 Int32 *in = (Int32 *)inBuffer;
10017 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10018 for (j=0; j<info.channels; j++) {
10019 out[info.outOffset[j]] = in[info.inOffset[j]];
10022 out += info.outJump;
10025 else if (info.inFormat == RTAUDIO_FLOAT32) {
10026 Float32 *in = (Float32 *)inBuffer;
10027 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10028 for (j=0; j<info.channels; j++) {
10029 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
10032 out += info.outJump;
10035 else if (info.inFormat == RTAUDIO_FLOAT64) {
10036 Float64 *in = (Float64 *)inBuffer;
10037 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10038 for (j=0; j<info.channels; j++) {
10039 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
10042 out += info.outJump;
10046 else if (info.outFormat == RTAUDIO_SINT24) {
10047 Int24 *out = (Int24 *)outBuffer;
10048 if (info.inFormat == RTAUDIO_SINT8) {
10049 signed char *in = (signed char *)inBuffer;
10050 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10051 for (j=0; j<info.channels; j++) {
10052 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] << 16);
10053 //out[info.outOffset[j]] <<= 16;
10056 out += info.outJump;
10059 else if (info.inFormat == RTAUDIO_SINT16) {
10060 Int16 *in = (Int16 *)inBuffer;
10061 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10062 for (j=0; j<info.channels; j++) {
10063 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] << 8);
10064 //out[info.outOffset[j]] <<= 8;
10067 out += info.outJump;
10070 else if (info.inFormat == RTAUDIO_SINT24) {
10071 // Channel compensation and/or (de)interleaving only.
10072 Int24 *in = (Int24 *)inBuffer;
10073 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10074 for (j=0; j<info.channels; j++) {
10075 out[info.outOffset[j]] = in[info.inOffset[j]];
10078 out += info.outJump;
10081 else if (info.inFormat == RTAUDIO_SINT32) {
10082 Int32 *in = (Int32 *)inBuffer;
10083 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10084 for (j=0; j<info.channels; j++) {
10085 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] >> 8);
10086 //out[info.outOffset[j]] >>= 8;
10089 out += info.outJump;
10092 else if (info.inFormat == RTAUDIO_FLOAT32) {
10093 Float32 *in = (Float32 *)inBuffer;
10094 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10095 for (j=0; j<info.channels; j++) {
10096 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
10099 out += info.outJump;
10102 else if (info.inFormat == RTAUDIO_FLOAT64) {
10103 Float64 *in = (Float64 *)inBuffer;
10104 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10105 for (j=0; j<info.channels; j++) {
10106 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
10109 out += info.outJump;
10113 else if (info.outFormat == RTAUDIO_SINT16) {
10114 Int16 *out = (Int16 *)outBuffer;
10115 if (info.inFormat == RTAUDIO_SINT8) {
10116 signed char *in = (signed char *)inBuffer;
10117 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10118 for (j=0; j<info.channels; j++) {
10119 out[info.outOffset[j]] = (Int16) in[info.inOffset[j]];
10120 out[info.outOffset[j]] <<= 8;
10123 out += info.outJump;
10126 else if (info.inFormat == RTAUDIO_SINT16) {
10127 // Channel compensation and/or (de)interleaving only.
10128 Int16 *in = (Int16 *)inBuffer;
10129 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10130 for (j=0; j<info.channels; j++) {
10131 out[info.outOffset[j]] = in[info.inOffset[j]];
10134 out += info.outJump;
10137 else if (info.inFormat == RTAUDIO_SINT24) {
10138 Int24 *in = (Int24 *)inBuffer;
10139 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10140 for (j=0; j<info.channels; j++) {
10141 out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]].asInt() >> 8);
10144 out += info.outJump;
10147 else if (info.inFormat == RTAUDIO_SINT32) {
10148 Int32 *in = (Int32 *)inBuffer;
10149 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10150 for (j=0; j<info.channels; j++) {
10151 out[info.outOffset[j]] = (Int16) ((in[info.inOffset[j]] >> 16) & 0x0000ffff);
10154 out += info.outJump;
10157 else if (info.inFormat == RTAUDIO_FLOAT32) {
10158 Float32 *in = (Float32 *)inBuffer;
10159 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10160 for (j=0; j<info.channels; j++) {
10161 out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
10164 out += info.outJump;
10167 else if (info.inFormat == RTAUDIO_FLOAT64) {
10168 Float64 *in = (Float64 *)inBuffer;
10169 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10170 for (j=0; j<info.channels; j++) {
10171 out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
10174 out += info.outJump;
10178 else if (info.outFormat == RTAUDIO_SINT8) {
10179 signed char *out = (signed char *)outBuffer;
10180 if (info.inFormat == RTAUDIO_SINT8) {
10181 // Channel compensation and/or (de)interleaving only.
10182 signed char *in = (signed char *)inBuffer;
10183 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10184 for (j=0; j<info.channels; j++) {
10185 out[info.outOffset[j]] = in[info.inOffset[j]];
10188 out += info.outJump;
10191 if (info.inFormat == RTAUDIO_SINT16) {
10192 Int16 *in = (Int16 *)inBuffer;
10193 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10194 for (j=0; j<info.channels; j++) {
10195 out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 8) & 0x00ff);
10198 out += info.outJump;
10201 else if (info.inFormat == RTAUDIO_SINT24) {
10202 Int24 *in = (Int24 *)inBuffer;
10203 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10204 for (j=0; j<info.channels; j++) {
10205 out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]].asInt() >> 16);
10208 out += info.outJump;
10211 else if (info.inFormat == RTAUDIO_SINT32) {
10212 Int32 *in = (Int32 *)inBuffer;
10213 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10214 for (j=0; j<info.channels; j++) {
10215 out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 24) & 0x000000ff);
10218 out += info.outJump;
10221 else if (info.inFormat == RTAUDIO_FLOAT32) {
10222 Float32 *in = (Float32 *)inBuffer;
10223 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10224 for (j=0; j<info.channels; j++) {
10225 out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
10228 out += info.outJump;
10231 else if (info.inFormat == RTAUDIO_FLOAT64) {
10232 Float64 *in = (Float64 *)inBuffer;
10233 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10234 for (j=0; j<info.channels; j++) {
10235 out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
10238 out += info.outJump;
10244 //static inline uint16_t bswap_16(uint16_t x) { return (x>>8) | (x<<8); }
10245 //static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); }
10246 //static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); }
10248 void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format )
10254 if ( format == RTAUDIO_SINT16 ) {
10255 for ( unsigned int i=0; i<samples; i++ ) {
10256 // Swap 1st and 2nd bytes.
10261 // Increment 2 bytes.
10265 else if ( format == RTAUDIO_SINT32 ||
10266 format == RTAUDIO_FLOAT32 ) {
10267 for ( unsigned int i=0; i<samples; i++ ) {
10268 // Swap 1st and 4th bytes.
10273 // Swap 2nd and 3rd bytes.
10279 // Increment 3 more bytes.
10283 else if ( format == RTAUDIO_SINT24 ) {
10284 for ( unsigned int i=0; i<samples; i++ ) {
10285 // Swap 1st and 3rd bytes.
10290 // Increment 2 more bytes.
10294 else if ( format == RTAUDIO_FLOAT64 ) {
10295 for ( unsigned int i=0; i<samples; i++ ) {
10296 // Swap 1st and 8th bytes
10301 // Swap 2nd and 7th bytes
10307 // Swap 3rd and 6th bytes
10313 // Swap 4th and 5th bytes
10319 // Increment 5 more bytes.
10325 // Indentation settings for Vim and Emacs
10327 // Local Variables:
10328 // c-basic-offset: 2
10329 // indent-tabs-mode: nil
10332 // vim: et sts=2 sw=2