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 // Define API names and display names.
102 // Must be in same order as API enum.
104 const char* rtaudio_api_names[][2] = {
105 { "unspecified" , "Unknown" },
107 { "pulse" , "Pulse" },
108 { "oss" , "OpenSoundSystem" },
110 { "core" , "CoreAudio" },
111 { "wasapi" , "WASAPI" },
113 { "ds" , "DirectSound" },
114 { "dummy" , "Dummy" },
116 const unsigned int rtaudio_num_api_names =
117 sizeof(rtaudio_api_names)/sizeof(rtaudio_api_names[0]);
119 // The order here will control the order of RtAudio's API search in
121 extern "C" const RtAudio::Api rtaudio_compiled_apis[] = {
122 #if defined(__UNIX_JACK__)
125 #if defined(__LINUX_PULSE__)
126 RtAudio::LINUX_PULSE,
128 #if defined(__LINUX_ALSA__)
131 #if defined(__LINUX_OSS__)
134 #if defined(__WINDOWS_ASIO__)
135 RtAudio::WINDOWS_ASIO,
137 #if defined(__WINDOWS_WASAPI__)
138 RtAudio::WINDOWS_WASAPI,
140 #if defined(__WINDOWS_DS__)
143 #if defined(__MACOSX_CORE__)
144 RtAudio::MACOSX_CORE,
146 #if defined(__RTAUDIO_DUMMY__)
147 RtAudio::RTAUDIO_DUMMY,
149 RtAudio::UNSPECIFIED,
151 extern "C" const unsigned int rtaudio_num_compiled_apis =
152 sizeof(rtaudio_compiled_apis)/sizeof(rtaudio_compiled_apis[0])-1;
155 // This is a compile-time check that rtaudio_num_api_names == RtAudio::NUM_APIS.
156 // If the build breaks here, check that they match.
157 template<bool b> class StaticAssert { private: StaticAssert() {} };
158 template<> class StaticAssert<true>{ public: StaticAssert() {} };
159 class StaticAssertions { StaticAssertions() {
160 StaticAssert<rtaudio_num_api_names == RtAudio::NUM_APIS>();
163 void RtAudio :: getCompiledApi( std::vector<RtAudio::Api> &apis )
165 apis = std::vector<RtAudio::Api>(rtaudio_compiled_apis,
166 rtaudio_compiled_apis + rtaudio_num_compiled_apis);
169 std::string RtAudio :: getApiName( RtAudio::Api api )
171 if (api < 0 || api >= RtAudio::NUM_APIS)
173 return rtaudio_api_names[api][0];
176 std::string RtAudio :: getApiDisplayName( RtAudio::Api api )
178 if (api < 0 || api >= RtAudio::NUM_APIS)
180 return rtaudio_api_names[api][1];
183 RtAudio::Api RtAudio :: getCompiledApiByName( const std::string &name )
186 for (i = 0; i < rtaudio_num_compiled_apis; ++i)
187 if (name == rtaudio_api_names[rtaudio_compiled_apis[i]][0])
188 return rtaudio_compiled_apis[i];
189 return RtAudio::UNSPECIFIED;
192 void RtAudio :: openRtApi( RtAudio::Api api )
198 #if defined(__UNIX_JACK__)
199 if ( api == UNIX_JACK )
200 rtapi_ = new RtApiJack();
202 #if defined(__LINUX_ALSA__)
203 if ( api == LINUX_ALSA )
204 rtapi_ = new RtApiAlsa();
206 #if defined(__LINUX_PULSE__)
207 if ( api == LINUX_PULSE )
208 rtapi_ = new RtApiPulse();
210 #if defined(__LINUX_OSS__)
211 if ( api == LINUX_OSS )
212 rtapi_ = new RtApiOss();
214 #if defined(__WINDOWS_ASIO__)
215 if ( api == WINDOWS_ASIO )
216 rtapi_ = new RtApiAsio();
218 #if defined(__WINDOWS_WASAPI__)
219 if ( api == WINDOWS_WASAPI )
220 rtapi_ = new RtApiWasapi();
222 #if defined(__WINDOWS_DS__)
223 if ( api == WINDOWS_DS )
224 rtapi_ = new RtApiDs();
226 #if defined(__MACOSX_CORE__)
227 if ( api == MACOSX_CORE )
228 rtapi_ = new RtApiCore();
230 #if defined(__RTAUDIO_DUMMY__)
231 if ( api == RTAUDIO_DUMMY )
232 rtapi_ = new RtApiDummy();
236 RtAudio :: RtAudio( RtAudio::Api api )
240 if ( api != UNSPECIFIED ) {
241 // Attempt to open the specified API.
243 if ( rtapi_ ) return;
245 // No compiled support for specified API value. Issue a debug
246 // warning and continue as if no API was specified.
247 std::cerr << "\nRtAudio: no compiled support for specified API argument!\n" << std::endl;
250 // Iterate through the compiled APIs and return as soon as we find
251 // one with at least one device or we reach the end of the list.
252 std::vector< RtAudio::Api > apis;
253 getCompiledApi( apis );
254 for ( unsigned int i=0; i<apis.size(); i++ ) {
255 openRtApi( apis[i] );
256 if ( rtapi_ && rtapi_->getDeviceCount() ) break;
259 if ( rtapi_ ) return;
261 // It should not be possible to get here because the preprocessor
262 // definition __RTAUDIO_DUMMY__ is automatically defined if no
263 // API-specific definitions are passed to the compiler. But just in
264 // case something weird happens, we'll thow an error.
265 std::string errorText = "\nRtAudio: no compiled API support found ... critical error!!\n\n";
266 throw( RtAudioError( errorText, RtAudioError::UNSPECIFIED ) );
269 RtAudio :: ~RtAudio()
275 void RtAudio :: openStream( RtAudio::StreamParameters *outputParameters,
276 RtAudio::StreamParameters *inputParameters,
277 RtAudioFormat format, unsigned int sampleRate,
278 unsigned int *bufferFrames,
279 RtAudioCallback callback, void *userData,
280 RtAudio::StreamOptions *options,
281 RtAudioErrorCallback errorCallback )
283 return rtapi_->openStream( outputParameters, inputParameters, format,
284 sampleRate, bufferFrames, callback,
285 userData, options, errorCallback );
288 // *************************************************** //
290 // Public RtApi definitions (see end of file for
291 // private or protected utility functions).
293 // *************************************************** //
297 stream_.state = STREAM_CLOSED;
298 stream_.mode = UNINITIALIZED;
299 stream_.apiHandle = 0;
300 stream_.userBuffer[0] = 0;
301 stream_.userBuffer[1] = 0;
302 MUTEX_INITIALIZE( &stream_.mutex );
303 showWarnings_ = true;
304 firstErrorOccurred_ = false;
309 MUTEX_DESTROY( &stream_.mutex );
312 void RtApi :: openStream( RtAudio::StreamParameters *oParams,
313 RtAudio::StreamParameters *iParams,
314 RtAudioFormat format, unsigned int sampleRate,
315 unsigned int *bufferFrames,
316 RtAudioCallback callback, void *userData,
317 RtAudio::StreamOptions *options,
318 RtAudioErrorCallback errorCallback )
320 if ( stream_.state != STREAM_CLOSED ) {
321 errorText_ = "RtApi::openStream: a stream is already open!";
322 error( RtAudioError::INVALID_USE );
326 // Clear stream information potentially left from a previously open stream.
329 if ( oParams && oParams->nChannels < 1 ) {
330 errorText_ = "RtApi::openStream: a non-NULL output StreamParameters structure cannot have an nChannels value less than one.";
331 error( RtAudioError::INVALID_USE );
335 if ( iParams && iParams->nChannels < 1 ) {
336 errorText_ = "RtApi::openStream: a non-NULL input StreamParameters structure cannot have an nChannels value less than one.";
337 error( RtAudioError::INVALID_USE );
341 if ( oParams == NULL && iParams == NULL ) {
342 errorText_ = "RtApi::openStream: input and output StreamParameters structures are both NULL!";
343 error( RtAudioError::INVALID_USE );
347 if ( formatBytes(format) == 0 ) {
348 errorText_ = "RtApi::openStream: 'format' parameter value is undefined.";
349 error( RtAudioError::INVALID_USE );
353 unsigned int nDevices = getDeviceCount();
354 unsigned int oChannels = 0;
356 oChannels = oParams->nChannels;
357 if ( oParams->deviceId >= nDevices ) {
358 errorText_ = "RtApi::openStream: output device parameter value is invalid.";
359 error( RtAudioError::INVALID_USE );
364 unsigned int iChannels = 0;
366 iChannels = iParams->nChannels;
367 if ( iParams->deviceId >= nDevices ) {
368 errorText_ = "RtApi::openStream: input device parameter value is invalid.";
369 error( RtAudioError::INVALID_USE );
376 if ( oChannels > 0 ) {
378 result = probeDeviceOpen( oParams->deviceId, OUTPUT, oChannels, oParams->firstChannel,
379 sampleRate, format, bufferFrames, options );
380 if ( result == false ) {
381 error( RtAudioError::SYSTEM_ERROR );
386 if ( iChannels > 0 ) {
388 result = probeDeviceOpen( iParams->deviceId, INPUT, iChannels, iParams->firstChannel,
389 sampleRate, format, bufferFrames, options );
390 if ( result == false ) {
391 if ( oChannels > 0 ) closeStream();
392 error( RtAudioError::SYSTEM_ERROR );
397 stream_.callbackInfo.callback = (void *) callback;
398 stream_.callbackInfo.userData = userData;
399 stream_.callbackInfo.errorCallback = (void *) errorCallback;
401 if ( options ) options->numberOfBuffers = stream_.nBuffers;
402 stream_.state = STREAM_STOPPED;
405 unsigned int RtApi :: getDefaultInputDevice( void )
407 // Should be implemented in subclasses if possible.
411 unsigned int RtApi :: getDefaultOutputDevice( void )
413 // Should be implemented in subclasses if possible.
417 void RtApi :: closeStream( void )
419 // MUST be implemented in subclasses!
423 bool RtApi :: probeDeviceOpen( unsigned int /*device*/, StreamMode /*mode*/, unsigned int /*channels*/,
424 unsigned int /*firstChannel*/, unsigned int /*sampleRate*/,
425 RtAudioFormat /*format*/, unsigned int * /*bufferSize*/,
426 RtAudio::StreamOptions * /*options*/ )
428 // MUST be implemented in subclasses!
432 void RtApi :: tickStreamTime( void )
434 // Subclasses that do not provide their own implementation of
435 // getStreamTime should call this function once per buffer I/O to
436 // provide basic stream time support.
438 stream_.streamTime += ( stream_.bufferSize * 1.0 / stream_.sampleRate );
440 #if defined( HAVE_GETTIMEOFDAY )
441 gettimeofday( &stream_.lastTickTimestamp, NULL );
445 long RtApi :: getStreamLatency( void )
449 long totalLatency = 0;
450 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
451 totalLatency = stream_.latency[0];
452 if ( stream_.mode == INPUT || stream_.mode == DUPLEX )
453 totalLatency += stream_.latency[1];
458 double RtApi :: getStreamTime( void )
462 #if defined( HAVE_GETTIMEOFDAY )
463 // Return a very accurate estimate of the stream time by
464 // adding in the elapsed time since the last tick.
468 if ( stream_.state != STREAM_RUNNING || stream_.streamTime == 0.0 )
469 return stream_.streamTime;
471 gettimeofday( &now, NULL );
472 then = stream_.lastTickTimestamp;
473 return stream_.streamTime +
474 ((now.tv_sec + 0.000001 * now.tv_usec) -
475 (then.tv_sec + 0.000001 * then.tv_usec));
477 return stream_.streamTime;
481 void RtApi :: setStreamTime( double time )
486 stream_.streamTime = time;
487 #if defined( HAVE_GETTIMEOFDAY )
488 gettimeofday( &stream_.lastTickTimestamp, NULL );
492 unsigned int RtApi :: getStreamSampleRate( void )
496 return stream_.sampleRate;
500 // *************************************************** //
502 // OS/API-specific methods.
504 // *************************************************** //
506 #if defined(__MACOSX_CORE__)
508 // The OS X CoreAudio API is designed to use a separate callback
509 // procedure for each of its audio devices. A single RtAudio duplex
510 // stream using two different devices is supported here, though it
511 // cannot be guaranteed to always behave correctly because we cannot
512 // synchronize these two callbacks.
514 // A property listener is installed for over/underrun information.
515 // However, no functionality is currently provided to allow property
516 // listeners to trigger user handlers because it is unclear what could
517 // be done if a critical stream parameter (buffer size, sample rate,
518 // device disconnect) notification arrived. The listeners entail
519 // quite a bit of extra code and most likely, a user program wouldn't
520 // be prepared for the result anyway. However, we do provide a flag
521 // to the client callback function to inform of an over/underrun.
523 // A structure to hold various information related to the CoreAudio API
526 AudioDeviceID id[2]; // device ids
527 #if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
528 AudioDeviceIOProcID procId[2];
530 UInt32 iStream[2]; // device stream index (or first if using multiple)
531 UInt32 nStreams[2]; // number of streams to use
534 pthread_cond_t condition;
535 int drainCounter; // Tracks callback counts when draining
536 bool internalDrain; // Indicates if stop is initiated from callback or not.
539 :deviceBuffer(0), drainCounter(0), internalDrain(false) { nStreams[0] = 1; nStreams[1] = 1; id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
542 RtApiCore:: RtApiCore()
544 #if defined( AVAILABLE_MAC_OS_X_VERSION_10_6_AND_LATER )
545 // This is a largely undocumented but absolutely necessary
546 // requirement starting with OS-X 10.6. If not called, queries and
547 // updates to various audio device properties are not handled
549 CFRunLoopRef theRunLoop = NULL;
550 AudioObjectPropertyAddress property = { kAudioHardwarePropertyRunLoop,
551 kAudioObjectPropertyScopeGlobal,
552 kAudioObjectPropertyElementMaster };
553 OSStatus result = AudioObjectSetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, sizeof(CFRunLoopRef), &theRunLoop);
554 if ( result != noErr ) {
555 errorText_ = "RtApiCore::RtApiCore: error setting run loop property!";
556 error( RtAudioError::WARNING );
561 RtApiCore :: ~RtApiCore()
563 // The subclass destructor gets called before the base class
564 // destructor, so close an existing stream before deallocating
565 // apiDeviceId memory.
566 if ( stream_.state != STREAM_CLOSED ) closeStream();
569 unsigned int RtApiCore :: getDeviceCount( void )
571 // Find out how many audio devices there are, if any.
573 AudioObjectPropertyAddress propertyAddress = { kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
574 OSStatus result = AudioObjectGetPropertyDataSize( kAudioObjectSystemObject, &propertyAddress, 0, NULL, &dataSize );
575 if ( result != noErr ) {
576 errorText_ = "RtApiCore::getDeviceCount: OS-X error getting device info!";
577 error( RtAudioError::WARNING );
581 return dataSize / sizeof( AudioDeviceID );
584 unsigned int RtApiCore :: getDefaultInputDevice( void )
586 unsigned int nDevices = getDeviceCount();
587 if ( nDevices <= 1 ) return 0;
590 UInt32 dataSize = sizeof( AudioDeviceID );
591 AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultInputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
592 OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id );
593 if ( result != noErr ) {
594 errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device.";
595 error( RtAudioError::WARNING );
599 dataSize *= nDevices;
600 AudioDeviceID deviceList[ nDevices ];
601 property.mSelector = kAudioHardwarePropertyDevices;
602 result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList );
603 if ( result != noErr ) {
604 errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device IDs.";
605 error( RtAudioError::WARNING );
609 for ( unsigned int i=0; i<nDevices; i++ )
610 if ( id == deviceList[i] ) return i;
612 errorText_ = "RtApiCore::getDefaultInputDevice: No default device found!";
613 error( RtAudioError::WARNING );
617 unsigned int RtApiCore :: getDefaultOutputDevice( void )
619 unsigned int nDevices = getDeviceCount();
620 if ( nDevices <= 1 ) return 0;
623 UInt32 dataSize = sizeof( AudioDeviceID );
624 AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultOutputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
625 OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id );
626 if ( result != noErr ) {
627 errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device.";
628 error( RtAudioError::WARNING );
632 dataSize = sizeof( AudioDeviceID ) * nDevices;
633 AudioDeviceID deviceList[ nDevices ];
634 property.mSelector = kAudioHardwarePropertyDevices;
635 result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList );
636 if ( result != noErr ) {
637 errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device IDs.";
638 error( RtAudioError::WARNING );
642 for ( unsigned int i=0; i<nDevices; i++ )
643 if ( id == deviceList[i] ) return i;
645 errorText_ = "RtApiCore::getDefaultOutputDevice: No default device found!";
646 error( RtAudioError::WARNING );
650 RtAudio::DeviceInfo RtApiCore :: getDeviceInfo( unsigned int device )
652 RtAudio::DeviceInfo info;
656 unsigned int nDevices = getDeviceCount();
657 if ( nDevices == 0 ) {
658 errorText_ = "RtApiCore::getDeviceInfo: no devices found!";
659 error( RtAudioError::INVALID_USE );
663 if ( device >= nDevices ) {
664 errorText_ = "RtApiCore::getDeviceInfo: device ID is invalid!";
665 error( RtAudioError::INVALID_USE );
669 AudioDeviceID deviceList[ nDevices ];
670 UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;
671 AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
672 kAudioObjectPropertyScopeGlobal,
673 kAudioObjectPropertyElementMaster };
674 OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property,
675 0, NULL, &dataSize, (void *) &deviceList );
676 if ( result != noErr ) {
677 errorText_ = "RtApiCore::getDeviceInfo: OS-X system error getting device IDs.";
678 error( RtAudioError::WARNING );
682 AudioDeviceID id = deviceList[ device ];
684 // Get the device name.
687 dataSize = sizeof( CFStringRef );
688 property.mSelector = kAudioObjectPropertyManufacturer;
689 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname );
690 if ( result != noErr ) {
691 errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device manufacturer.";
692 errorText_ = errorStream_.str();
693 error( RtAudioError::WARNING );
697 //const char *mname = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() );
698 int length = CFStringGetLength(cfname);
699 char *mname = (char *)malloc(length * 3 + 1);
700 #if defined( UNICODE ) || defined( _UNICODE )
701 CFStringGetCString(cfname, mname, length * 3 + 1, kCFStringEncodingUTF8);
703 CFStringGetCString(cfname, mname, length * 3 + 1, CFStringGetSystemEncoding());
705 info.name.append( (const char *)mname, strlen(mname) );
706 info.name.append( ": " );
710 property.mSelector = kAudioObjectPropertyName;
711 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname );
712 if ( result != noErr ) {
713 errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device name.";
714 errorText_ = errorStream_.str();
715 error( RtAudioError::WARNING );
719 //const char *name = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() );
720 length = CFStringGetLength(cfname);
721 char *name = (char *)malloc(length * 3 + 1);
722 #if defined( UNICODE ) || defined( _UNICODE )
723 CFStringGetCString(cfname, name, length * 3 + 1, kCFStringEncodingUTF8);
725 CFStringGetCString(cfname, name, length * 3 + 1, CFStringGetSystemEncoding());
727 info.name.append( (const char *)name, strlen(name) );
731 // Get the output stream "configuration".
732 AudioBufferList *bufferList = nil;
733 property.mSelector = kAudioDevicePropertyStreamConfiguration;
734 property.mScope = kAudioDevicePropertyScopeOutput;
735 // property.mElement = kAudioObjectPropertyElementWildcard;
737 result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
738 if ( result != noErr || dataSize == 0 ) {
739 errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration info for device (" << device << ").";
740 errorText_ = errorStream_.str();
741 error( RtAudioError::WARNING );
745 // Allocate the AudioBufferList.
746 bufferList = (AudioBufferList *) malloc( dataSize );
747 if ( bufferList == NULL ) {
748 errorText_ = "RtApiCore::getDeviceInfo: memory error allocating output AudioBufferList.";
749 error( RtAudioError::WARNING );
753 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
754 if ( result != noErr || dataSize == 0 ) {
756 errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration for device (" << device << ").";
757 errorText_ = errorStream_.str();
758 error( RtAudioError::WARNING );
762 // Get output channel information.
763 unsigned int i, nStreams = bufferList->mNumberBuffers;
764 for ( i=0; i<nStreams; i++ )
765 info.outputChannels += bufferList->mBuffers[i].mNumberChannels;
768 // Get the input stream "configuration".
769 property.mScope = kAudioDevicePropertyScopeInput;
770 result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
771 if ( result != noErr || dataSize == 0 ) {
772 errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration info for device (" << device << ").";
773 errorText_ = errorStream_.str();
774 error( RtAudioError::WARNING );
778 // Allocate the AudioBufferList.
779 bufferList = (AudioBufferList *) malloc( dataSize );
780 if ( bufferList == NULL ) {
781 errorText_ = "RtApiCore::getDeviceInfo: memory error allocating input AudioBufferList.";
782 error( RtAudioError::WARNING );
786 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
787 if (result != noErr || dataSize == 0) {
789 errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration for device (" << device << ").";
790 errorText_ = errorStream_.str();
791 error( RtAudioError::WARNING );
795 // Get input channel information.
796 nStreams = bufferList->mNumberBuffers;
797 for ( i=0; i<nStreams; i++ )
798 info.inputChannels += bufferList->mBuffers[i].mNumberChannels;
801 // If device opens for both playback and capture, we determine the channels.
802 if ( info.outputChannels > 0 && info.inputChannels > 0 )
803 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
805 // Probe the device sample rates.
806 bool isInput = false;
807 if ( info.outputChannels == 0 ) isInput = true;
809 // Determine the supported sample rates.
810 property.mSelector = kAudioDevicePropertyAvailableNominalSampleRates;
811 if ( isInput == false ) property.mScope = kAudioDevicePropertyScopeOutput;
812 result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
813 if ( result != kAudioHardwareNoError || dataSize == 0 ) {
814 errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rate info.";
815 errorText_ = errorStream_.str();
816 error( RtAudioError::WARNING );
820 UInt32 nRanges = dataSize / sizeof( AudioValueRange );
821 AudioValueRange rangeList[ nRanges ];
822 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &rangeList );
823 if ( result != kAudioHardwareNoError ) {
824 errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rates.";
825 errorText_ = errorStream_.str();
826 error( RtAudioError::WARNING );
830 // The sample rate reporting mechanism is a bit of a mystery. It
831 // seems that it can either return individual rates or a range of
832 // rates. I assume that if the min / max range values are the same,
833 // then that represents a single supported rate and if the min / max
834 // range values are different, the device supports an arbitrary
835 // range of values (though there might be multiple ranges, so we'll
836 // use the most conservative range).
837 Float64 minimumRate = 1.0, maximumRate = 10000000000.0;
838 bool haveValueRange = false;
839 info.sampleRates.clear();
840 for ( UInt32 i=0; i<nRanges; i++ ) {
841 if ( rangeList[i].mMinimum == rangeList[i].mMaximum ) {
842 unsigned int tmpSr = (unsigned int) rangeList[i].mMinimum;
843 info.sampleRates.push_back( tmpSr );
845 if ( !info.preferredSampleRate || ( tmpSr <= 48000 && tmpSr > info.preferredSampleRate ) )
846 info.preferredSampleRate = tmpSr;
849 haveValueRange = true;
850 if ( rangeList[i].mMinimum > minimumRate ) minimumRate = rangeList[i].mMinimum;
851 if ( rangeList[i].mMaximum < maximumRate ) maximumRate = rangeList[i].mMaximum;
855 if ( haveValueRange ) {
856 for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
857 if ( SAMPLE_RATES[k] >= (unsigned int) minimumRate && SAMPLE_RATES[k] <= (unsigned int) maximumRate ) {
858 info.sampleRates.push_back( SAMPLE_RATES[k] );
860 if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
861 info.preferredSampleRate = SAMPLE_RATES[k];
866 // Sort and remove any redundant values
867 std::sort( info.sampleRates.begin(), info.sampleRates.end() );
868 info.sampleRates.erase( unique( info.sampleRates.begin(), info.sampleRates.end() ), info.sampleRates.end() );
870 if ( info.sampleRates.size() == 0 ) {
871 errorStream_ << "RtApiCore::probeDeviceInfo: No supported sample rates found for device (" << device << ").";
872 errorText_ = errorStream_.str();
873 error( RtAudioError::WARNING );
877 // CoreAudio always uses 32-bit floating point data for PCM streams.
878 // Thus, any other "physical" formats supported by the device are of
879 // no interest to the client.
880 info.nativeFormats = RTAUDIO_FLOAT32;
882 if ( info.outputChannels > 0 )
883 if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true;
884 if ( info.inputChannels > 0 )
885 if ( getDefaultInputDevice() == device ) info.isDefaultInput = true;
891 static OSStatus callbackHandler( AudioDeviceID inDevice,
892 const AudioTimeStamp* /*inNow*/,
893 const AudioBufferList* inInputData,
894 const AudioTimeStamp* /*inInputTime*/,
895 AudioBufferList* outOutputData,
896 const AudioTimeStamp* /*inOutputTime*/,
899 CallbackInfo *info = (CallbackInfo *) infoPointer;
901 RtApiCore *object = (RtApiCore *) info->object;
902 if ( object->callbackEvent( inDevice, inInputData, outOutputData ) == false )
903 return kAudioHardwareUnspecifiedError;
905 return kAudioHardwareNoError;
908 static OSStatus xrunListener( AudioObjectID /*inDevice*/,
910 const AudioObjectPropertyAddress properties[],
911 void* handlePointer )
913 CoreHandle *handle = (CoreHandle *) handlePointer;
914 for ( UInt32 i=0; i<nAddresses; i++ ) {
915 if ( properties[i].mSelector == kAudioDeviceProcessorOverload ) {
916 if ( properties[i].mScope == kAudioDevicePropertyScopeInput )
917 handle->xrun[1] = true;
919 handle->xrun[0] = true;
923 return kAudioHardwareNoError;
926 static OSStatus rateListener( AudioObjectID inDevice,
927 UInt32 /*nAddresses*/,
928 const AudioObjectPropertyAddress /*properties*/[],
931 Float64 *rate = (Float64 *) ratePointer;
932 UInt32 dataSize = sizeof( Float64 );
933 AudioObjectPropertyAddress property = { kAudioDevicePropertyNominalSampleRate,
934 kAudioObjectPropertyScopeGlobal,
935 kAudioObjectPropertyElementMaster };
936 AudioObjectGetPropertyData( inDevice, &property, 0, NULL, &dataSize, rate );
937 return kAudioHardwareNoError;
940 bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
941 unsigned int firstChannel, unsigned int sampleRate,
942 RtAudioFormat format, unsigned int *bufferSize,
943 RtAudio::StreamOptions *options )
946 unsigned int nDevices = getDeviceCount();
947 if ( nDevices == 0 ) {
948 // This should not happen because a check is made before this function is called.
949 errorText_ = "RtApiCore::probeDeviceOpen: no devices found!";
953 if ( device >= nDevices ) {
954 // This should not happen because a check is made before this function is called.
955 errorText_ = "RtApiCore::probeDeviceOpen: device ID is invalid!";
959 AudioDeviceID deviceList[ nDevices ];
960 UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;
961 AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
962 kAudioObjectPropertyScopeGlobal,
963 kAudioObjectPropertyElementMaster };
964 OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property,
965 0, NULL, &dataSize, (void *) &deviceList );
966 if ( result != noErr ) {
967 errorText_ = "RtApiCore::probeDeviceOpen: OS-X system error getting device IDs.";
971 AudioDeviceID id = deviceList[ device ];
973 // Setup for stream mode.
974 bool isInput = false;
975 if ( mode == INPUT ) {
977 property.mScope = kAudioDevicePropertyScopeInput;
980 property.mScope = kAudioDevicePropertyScopeOutput;
982 // Get the stream "configuration".
983 AudioBufferList *bufferList = nil;
985 property.mSelector = kAudioDevicePropertyStreamConfiguration;
986 result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
987 if ( result != noErr || dataSize == 0 ) {
988 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration info for device (" << device << ").";
989 errorText_ = errorStream_.str();
993 // Allocate the AudioBufferList.
994 bufferList = (AudioBufferList *) malloc( dataSize );
995 if ( bufferList == NULL ) {
996 errorText_ = "RtApiCore::probeDeviceOpen: memory error allocating AudioBufferList.";
1000 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
1001 if (result != noErr || dataSize == 0) {
1003 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration for device (" << device << ").";
1004 errorText_ = errorStream_.str();
1008 // Search for one or more streams that contain the desired number of
1009 // channels. CoreAudio devices can have an arbitrary number of
1010 // streams and each stream can have an arbitrary number of channels.
1011 // For each stream, a single buffer of interleaved samples is
1012 // provided. RtAudio prefers the use of one stream of interleaved
1013 // data or multiple consecutive single-channel streams. However, we
1014 // now support multiple consecutive multi-channel streams of
1015 // interleaved data as well.
1016 UInt32 iStream, offsetCounter = firstChannel;
1017 UInt32 nStreams = bufferList->mNumberBuffers;
1018 bool monoMode = false;
1019 bool foundStream = false;
1021 // First check that the device supports the requested number of
1023 UInt32 deviceChannels = 0;
1024 for ( iStream=0; iStream<nStreams; iStream++ )
1025 deviceChannels += bufferList->mBuffers[iStream].mNumberChannels;
1027 if ( deviceChannels < ( channels + firstChannel ) ) {
1029 errorStream_ << "RtApiCore::probeDeviceOpen: the device (" << device << ") does not support the requested channel count.";
1030 errorText_ = errorStream_.str();
1034 // Look for a single stream meeting our needs.
1035 UInt32 firstStream, streamCount = 1, streamChannels = 0, channelOffset = 0;
1036 for ( iStream=0; iStream<nStreams; iStream++ ) {
1037 streamChannels = bufferList->mBuffers[iStream].mNumberChannels;
1038 if ( streamChannels >= channels + offsetCounter ) {
1039 firstStream = iStream;
1040 channelOffset = offsetCounter;
1044 if ( streamChannels > offsetCounter ) break;
1045 offsetCounter -= streamChannels;
1048 // If we didn't find a single stream above, then we should be able
1049 // to meet the channel specification with multiple streams.
1050 if ( foundStream == false ) {
1052 offsetCounter = firstChannel;
1053 for ( iStream=0; iStream<nStreams; iStream++ ) {
1054 streamChannels = bufferList->mBuffers[iStream].mNumberChannels;
1055 if ( streamChannels > offsetCounter ) break;
1056 offsetCounter -= streamChannels;
1059 firstStream = iStream;
1060 channelOffset = offsetCounter;
1061 Int32 channelCounter = channels + offsetCounter - streamChannels;
1063 if ( streamChannels > 1 ) monoMode = false;
1064 while ( channelCounter > 0 ) {
1065 streamChannels = bufferList->mBuffers[++iStream].mNumberChannels;
1066 if ( streamChannels > 1 ) monoMode = false;
1067 channelCounter -= streamChannels;
1074 // Determine the buffer size.
1075 AudioValueRange bufferRange;
1076 dataSize = sizeof( AudioValueRange );
1077 property.mSelector = kAudioDevicePropertyBufferFrameSizeRange;
1078 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &bufferRange );
1080 if ( result != noErr ) {
1081 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting buffer size range for device (" << device << ").";
1082 errorText_ = errorStream_.str();
1086 if ( bufferRange.mMinimum > *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMinimum;
1087 else if ( bufferRange.mMaximum < *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMaximum;
1088 if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) *bufferSize = (unsigned long) bufferRange.mMinimum;
1090 // Set the buffer size. For multiple streams, I'm assuming we only
1091 // need to make this setting for the master channel.
1092 UInt32 theSize = (UInt32) *bufferSize;
1093 dataSize = sizeof( UInt32 );
1094 property.mSelector = kAudioDevicePropertyBufferFrameSize;
1095 result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &theSize );
1097 if ( result != noErr ) {
1098 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting the buffer size for device (" << device << ").";
1099 errorText_ = errorStream_.str();
1103 // If attempting to setup a duplex stream, the bufferSize parameter
1104 // MUST be the same in both directions!
1105 *bufferSize = theSize;
1106 if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {
1107 errorStream_ << "RtApiCore::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << device << ").";
1108 errorText_ = errorStream_.str();
1112 stream_.bufferSize = *bufferSize;
1113 stream_.nBuffers = 1;
1115 // Try to set "hog" mode ... it's not clear to me this is working.
1116 if ( options && options->flags & RTAUDIO_HOG_DEVICE ) {
1118 dataSize = sizeof( hog_pid );
1119 property.mSelector = kAudioDevicePropertyHogMode;
1120 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &hog_pid );
1121 if ( result != noErr ) {
1122 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting 'hog' state!";
1123 errorText_ = errorStream_.str();
1127 if ( hog_pid != getpid() ) {
1129 result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &hog_pid );
1130 if ( result != noErr ) {
1131 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting 'hog' state!";
1132 errorText_ = errorStream_.str();
1138 // Check and if necessary, change the sample rate for the device.
1139 Float64 nominalRate;
1140 dataSize = sizeof( Float64 );
1141 property.mSelector = kAudioDevicePropertyNominalSampleRate;
1142 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &nominalRate );
1143 if ( result != noErr ) {
1144 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting current sample rate.";
1145 errorText_ = errorStream_.str();
1149 // Only change the sample rate if off by more than 1 Hz.
1150 if ( fabs( nominalRate - (double)sampleRate ) > 1.0 ) {
1152 // Set a property listener for the sample rate change
1153 Float64 reportedRate = 0.0;
1154 AudioObjectPropertyAddress tmp = { kAudioDevicePropertyNominalSampleRate, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
1155 result = AudioObjectAddPropertyListener( id, &tmp, rateListener, (void *) &reportedRate );
1156 if ( result != noErr ) {
1157 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate property listener for device (" << device << ").";
1158 errorText_ = errorStream_.str();
1162 nominalRate = (Float64) sampleRate;
1163 result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &nominalRate );
1164 if ( result != noErr ) {
1165 AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate );
1166 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate for device (" << device << ").";
1167 errorText_ = errorStream_.str();
1171 // Now wait until the reported nominal rate is what we just set.
1172 UInt32 microCounter = 0;
1173 while ( reportedRate != nominalRate ) {
1174 microCounter += 5000;
1175 if ( microCounter > 5000000 ) break;
1179 // Remove the property listener.
1180 AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate );
1182 if ( microCounter > 5000000 ) {
1183 errorStream_ << "RtApiCore::probeDeviceOpen: timeout waiting for sample rate update for device (" << device << ").";
1184 errorText_ = errorStream_.str();
1189 // Now set the stream format for all streams. Also, check the
1190 // physical format of the device and change that if necessary.
1191 AudioStreamBasicDescription description;
1192 dataSize = sizeof( AudioStreamBasicDescription );
1193 property.mSelector = kAudioStreamPropertyVirtualFormat;
1194 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &description );
1195 if ( result != noErr ) {
1196 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream format for device (" << device << ").";
1197 errorText_ = errorStream_.str();
1201 // Set the sample rate and data format id. However, only make the
1202 // change if the sample rate is not within 1.0 of the desired
1203 // rate and the format is not linear pcm.
1204 bool updateFormat = false;
1205 if ( fabs( description.mSampleRate - (Float64)sampleRate ) > 1.0 ) {
1206 description.mSampleRate = (Float64) sampleRate;
1207 updateFormat = true;
1210 if ( description.mFormatID != kAudioFormatLinearPCM ) {
1211 description.mFormatID = kAudioFormatLinearPCM;
1212 updateFormat = true;
1215 if ( updateFormat ) {
1216 result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &description );
1217 if ( result != noErr ) {
1218 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate or data format for device (" << device << ").";
1219 errorText_ = errorStream_.str();
1224 // Now check the physical format.
1225 property.mSelector = kAudioStreamPropertyPhysicalFormat;
1226 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &description );
1227 if ( result != noErr ) {
1228 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream physical format for device (" << device << ").";
1229 errorText_ = errorStream_.str();
1233 //std::cout << "Current physical stream format:" << std::endl;
1234 //std::cout << " mBitsPerChan = " << description.mBitsPerChannel << std::endl;
1235 //std::cout << " aligned high = " << (description.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (description.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl;
1236 //std::cout << " bytesPerFrame = " << description.mBytesPerFrame << std::endl;
1237 //std::cout << " sample rate = " << description.mSampleRate << std::endl;
1239 if ( description.mFormatID != kAudioFormatLinearPCM || description.mBitsPerChannel < 16 ) {
1240 description.mFormatID = kAudioFormatLinearPCM;
1241 //description.mSampleRate = (Float64) sampleRate;
1242 AudioStreamBasicDescription testDescription = description;
1245 // We'll try higher bit rates first and then work our way down.
1246 std::vector< std::pair<UInt32, UInt32> > physicalFormats;
1247 formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsFloat) & ~kLinearPCMFormatFlagIsSignedInteger;
1248 physicalFormats.push_back( std::pair<Float32, UInt32>( 32, formatFlags ) );
1249 formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat;
1250 physicalFormats.push_back( std::pair<Float32, UInt32>( 32, formatFlags ) );
1251 physicalFormats.push_back( std::pair<Float32, UInt32>( 24, formatFlags ) ); // 24-bit packed
1252 formatFlags &= ~( kAudioFormatFlagIsPacked | kAudioFormatFlagIsAlignedHigh );
1253 physicalFormats.push_back( std::pair<Float32, UInt32>( 24.2, formatFlags ) ); // 24-bit in 4 bytes, aligned low
1254 formatFlags |= kAudioFormatFlagIsAlignedHigh;
1255 physicalFormats.push_back( std::pair<Float32, UInt32>( 24.4, formatFlags ) ); // 24-bit in 4 bytes, aligned high
1256 formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat;
1257 physicalFormats.push_back( std::pair<Float32, UInt32>( 16, formatFlags ) );
1258 physicalFormats.push_back( std::pair<Float32, UInt32>( 8, formatFlags ) );
1260 bool setPhysicalFormat = false;
1261 for( unsigned int i=0; i<physicalFormats.size(); i++ ) {
1262 testDescription = description;
1263 testDescription.mBitsPerChannel = (UInt32) physicalFormats[i].first;
1264 testDescription.mFormatFlags = physicalFormats[i].second;
1265 if ( (24 == (UInt32)physicalFormats[i].first) && ~( physicalFormats[i].second & kAudioFormatFlagIsPacked ) )
1266 testDescription.mBytesPerFrame = 4 * testDescription.mChannelsPerFrame;
1268 testDescription.mBytesPerFrame = testDescription.mBitsPerChannel/8 * testDescription.mChannelsPerFrame;
1269 testDescription.mBytesPerPacket = testDescription.mBytesPerFrame * testDescription.mFramesPerPacket;
1270 result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &testDescription );
1271 if ( result == noErr ) {
1272 setPhysicalFormat = true;
1273 //std::cout << "Updated physical stream format:" << std::endl;
1274 //std::cout << " mBitsPerChan = " << testDescription.mBitsPerChannel << std::endl;
1275 //std::cout << " aligned high = " << (testDescription.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (testDescription.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl;
1276 //std::cout << " bytesPerFrame = " << testDescription.mBytesPerFrame << std::endl;
1277 //std::cout << " sample rate = " << testDescription.mSampleRate << std::endl;
1282 if ( !setPhysicalFormat ) {
1283 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting physical data format for device (" << device << ").";
1284 errorText_ = errorStream_.str();
1287 } // done setting virtual/physical formats.
1289 // Get the stream / device latency.
1291 dataSize = sizeof( UInt32 );
1292 property.mSelector = kAudioDevicePropertyLatency;
1293 if ( AudioObjectHasProperty( id, &property ) == true ) {
1294 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &latency );
1295 if ( result == kAudioHardwareNoError ) stream_.latency[ mode ] = latency;
1297 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting device latency for device (" << device << ").";
1298 errorText_ = errorStream_.str();
1299 error( RtAudioError::WARNING );
1303 // Byte-swapping: According to AudioHardware.h, the stream data will
1304 // always be presented in native-endian format, so we should never
1305 // need to byte swap.
1306 stream_.doByteSwap[mode] = false;
1308 // From the CoreAudio documentation, PCM data must be supplied as
1310 stream_.userFormat = format;
1311 stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
1313 if ( streamCount == 1 )
1314 stream_.nDeviceChannels[mode] = description.mChannelsPerFrame;
1315 else // multiple streams
1316 stream_.nDeviceChannels[mode] = channels;
1317 stream_.nUserChannels[mode] = channels;
1318 stream_.channelOffset[mode] = channelOffset; // offset within a CoreAudio stream
1319 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
1320 else stream_.userInterleaved = true;
1321 stream_.deviceInterleaved[mode] = true;
1322 if ( monoMode == true ) stream_.deviceInterleaved[mode] = false;
1324 // Set flags for buffer conversion.
1325 stream_.doConvertBuffer[mode] = false;
1326 if ( stream_.userFormat != stream_.deviceFormat[mode] )
1327 stream_.doConvertBuffer[mode] = true;
1328 if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
1329 stream_.doConvertBuffer[mode] = true;
1330 if ( streamCount == 1 ) {
1331 if ( stream_.nUserChannels[mode] > 1 &&
1332 stream_.userInterleaved != stream_.deviceInterleaved[mode] )
1333 stream_.doConvertBuffer[mode] = true;
1335 else if ( monoMode && stream_.userInterleaved )
1336 stream_.doConvertBuffer[mode] = true;
1338 // Allocate our CoreHandle structure for the stream.
1339 CoreHandle *handle = 0;
1340 if ( stream_.apiHandle == 0 ) {
1342 handle = new CoreHandle;
1344 catch ( std::bad_alloc& ) {
1345 errorText_ = "RtApiCore::probeDeviceOpen: error allocating CoreHandle memory.";
1349 if ( pthread_cond_init( &handle->condition, NULL ) ) {
1350 errorText_ = "RtApiCore::probeDeviceOpen: error initializing pthread condition variable.";
1353 stream_.apiHandle = (void *) handle;
1356 handle = (CoreHandle *) stream_.apiHandle;
1357 handle->iStream[mode] = firstStream;
1358 handle->nStreams[mode] = streamCount;
1359 handle->id[mode] = id;
1361 // Allocate necessary internal buffers.
1362 unsigned long bufferBytes;
1363 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
1364 // stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
1365 stream_.userBuffer[mode] = (char *) malloc( bufferBytes * sizeof(char) );
1366 memset( stream_.userBuffer[mode], 0, bufferBytes * sizeof(char) );
1367 if ( stream_.userBuffer[mode] == NULL ) {
1368 errorText_ = "RtApiCore::probeDeviceOpen: error allocating user buffer memory.";
1372 // If possible, we will make use of the CoreAudio stream buffers as
1373 // "device buffers". However, we can't do this if using multiple
1375 if ( stream_.doConvertBuffer[mode] && handle->nStreams[mode] > 1 ) {
1377 bool makeBuffer = true;
1378 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
1379 if ( mode == INPUT ) {
1380 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
1381 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
1382 if ( bufferBytes <= bytesOut ) makeBuffer = false;
1387 bufferBytes *= *bufferSize;
1388 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
1389 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
1390 if ( stream_.deviceBuffer == NULL ) {
1391 errorText_ = "RtApiCore::probeDeviceOpen: error allocating device buffer memory.";
1397 stream_.sampleRate = sampleRate;
1398 stream_.device[mode] = device;
1399 stream_.state = STREAM_STOPPED;
1400 stream_.callbackInfo.object = (void *) this;
1402 // Setup the buffer conversion information structure.
1403 if ( stream_.doConvertBuffer[mode] ) {
1404 if ( streamCount > 1 ) setConvertInfo( mode, 0 );
1405 else setConvertInfo( mode, channelOffset );
1408 if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device )
1409 // Only one callback procedure per device.
1410 stream_.mode = DUPLEX;
1412 #if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
1413 result = AudioDeviceCreateIOProcID( id, callbackHandler, (void *) &stream_.callbackInfo, &handle->procId[mode] );
1415 // deprecated in favor of AudioDeviceCreateIOProcID()
1416 result = AudioDeviceAddIOProc( id, callbackHandler, (void *) &stream_.callbackInfo );
1418 if ( result != noErr ) {
1419 errorStream_ << "RtApiCore::probeDeviceOpen: system error setting callback for device (" << device << ").";
1420 errorText_ = errorStream_.str();
1423 if ( stream_.mode == OUTPUT && mode == INPUT )
1424 stream_.mode = DUPLEX;
1426 stream_.mode = mode;
1429 // Setup the device property listener for over/underload.
1430 property.mSelector = kAudioDeviceProcessorOverload;
1431 property.mScope = kAudioObjectPropertyScopeGlobal;
1432 result = AudioObjectAddPropertyListener( id, &property, xrunListener, (void *) handle );
1438 pthread_cond_destroy( &handle->condition );
1440 stream_.apiHandle = 0;
1443 for ( int i=0; i<2; i++ ) {
1444 if ( stream_.userBuffer[i] ) {
1445 free( stream_.userBuffer[i] );
1446 stream_.userBuffer[i] = 0;
1450 if ( stream_.deviceBuffer ) {
1451 free( stream_.deviceBuffer );
1452 stream_.deviceBuffer = 0;
1455 stream_.state = STREAM_CLOSED;
1459 void RtApiCore :: closeStream( void )
1461 if ( stream_.state == STREAM_CLOSED ) {
1462 errorText_ = "RtApiCore::closeStream(): no open stream to close!";
1463 error( RtAudioError::WARNING );
1467 CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
1468 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
1470 AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
1471 kAudioObjectPropertyScopeGlobal,
1472 kAudioObjectPropertyElementMaster };
1474 property.mSelector = kAudioDeviceProcessorOverload;
1475 property.mScope = kAudioObjectPropertyScopeGlobal;
1476 if (AudioObjectRemovePropertyListener( handle->id[0], &property, xrunListener, (void *) handle ) != noErr) {
1477 errorText_ = "RtApiCore::closeStream(): error removing property listener!";
1478 error( RtAudioError::WARNING );
1481 if ( stream_.state == STREAM_RUNNING )
1482 AudioDeviceStop( handle->id[0], callbackHandler );
1483 #if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
1484 AudioDeviceDestroyIOProcID( handle->id[0], handle->procId[0] );
1486 // deprecated in favor of AudioDeviceDestroyIOProcID()
1487 AudioDeviceRemoveIOProc( handle->id[0], callbackHandler );
1491 if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
1493 AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
1494 kAudioObjectPropertyScopeGlobal,
1495 kAudioObjectPropertyElementMaster };
1497 property.mSelector = kAudioDeviceProcessorOverload;
1498 property.mScope = kAudioObjectPropertyScopeGlobal;
1499 if (AudioObjectRemovePropertyListener( handle->id[1], &property, xrunListener, (void *) handle ) != noErr) {
1500 errorText_ = "RtApiCore::closeStream(): error removing property listener!";
1501 error( RtAudioError::WARNING );
1504 if ( stream_.state == STREAM_RUNNING )
1505 AudioDeviceStop( handle->id[1], callbackHandler );
1506 #if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
1507 AudioDeviceDestroyIOProcID( handle->id[1], handle->procId[1] );
1509 // deprecated in favor of AudioDeviceDestroyIOProcID()
1510 AudioDeviceRemoveIOProc( handle->id[1], callbackHandler );
1514 for ( int i=0; i<2; i++ ) {
1515 if ( stream_.userBuffer[i] ) {
1516 free( stream_.userBuffer[i] );
1517 stream_.userBuffer[i] = 0;
1521 if ( stream_.deviceBuffer ) {
1522 free( stream_.deviceBuffer );
1523 stream_.deviceBuffer = 0;
1526 // Destroy pthread condition variable.
1527 pthread_cond_destroy( &handle->condition );
1529 stream_.apiHandle = 0;
1531 stream_.mode = UNINITIALIZED;
1532 stream_.state = STREAM_CLOSED;
1535 void RtApiCore :: startStream( void )
1538 if ( stream_.state == STREAM_RUNNING ) {
1539 errorText_ = "RtApiCore::startStream(): the stream is already running!";
1540 error( RtAudioError::WARNING );
1544 OSStatus result = noErr;
1545 CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
1546 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
1548 result = AudioDeviceStart( handle->id[0], callbackHandler );
1549 if ( result != noErr ) {
1550 errorStream_ << "RtApiCore::startStream: system error (" << getErrorCode( result ) << ") starting callback procedure on device (" << stream_.device[0] << ").";
1551 errorText_ = errorStream_.str();
1556 if ( stream_.mode == INPUT ||
1557 ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
1559 result = AudioDeviceStart( handle->id[1], callbackHandler );
1560 if ( result != noErr ) {
1561 errorStream_ << "RtApiCore::startStream: system error starting input callback procedure on device (" << stream_.device[1] << ").";
1562 errorText_ = errorStream_.str();
1567 handle->drainCounter = 0;
1568 handle->internalDrain = false;
1569 stream_.state = STREAM_RUNNING;
1572 if ( result == noErr ) return;
1573 error( RtAudioError::SYSTEM_ERROR );
1576 void RtApiCore :: stopStream( void )
1579 if ( stream_.state == STREAM_STOPPED ) {
1580 errorText_ = "RtApiCore::stopStream(): the stream is already stopped!";
1581 error( RtAudioError::WARNING );
1585 OSStatus result = noErr;
1586 CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
1587 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
1589 if ( handle->drainCounter == 0 ) {
1590 handle->drainCounter = 2;
1591 pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
1594 result = AudioDeviceStop( handle->id[0], callbackHandler );
1595 if ( result != noErr ) {
1596 errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping callback procedure on device (" << stream_.device[0] << ").";
1597 errorText_ = errorStream_.str();
1602 if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
1604 result = AudioDeviceStop( handle->id[1], callbackHandler );
1605 if ( result != noErr ) {
1606 errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping input callback procedure on device (" << stream_.device[1] << ").";
1607 errorText_ = errorStream_.str();
1612 stream_.state = STREAM_STOPPED;
1615 if ( result == noErr ) return;
1616 error( RtAudioError::SYSTEM_ERROR );
1619 void RtApiCore :: abortStream( void )
1622 if ( stream_.state == STREAM_STOPPED ) {
1623 errorText_ = "RtApiCore::abortStream(): the stream is already stopped!";
1624 error( RtAudioError::WARNING );
1628 CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
1629 handle->drainCounter = 2;
1634 // This function will be called by a spawned thread when the user
1635 // callback function signals that the stream should be stopped or
1636 // aborted. It is better to handle it this way because the
1637 // callbackEvent() function probably should return before the AudioDeviceStop()
1638 // function is called.
1639 static void *coreStopStream( void *ptr )
1641 CallbackInfo *info = (CallbackInfo *) ptr;
1642 RtApiCore *object = (RtApiCore *) info->object;
1644 object->stopStream();
1645 pthread_exit( NULL );
1648 bool RtApiCore :: callbackEvent( AudioDeviceID deviceId,
1649 const AudioBufferList *inBufferList,
1650 const AudioBufferList *outBufferList )
1652 if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;
1653 if ( stream_.state == STREAM_CLOSED ) {
1654 errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";
1655 error( RtAudioError::WARNING );
1659 CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
1660 CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
1662 // Check if we were draining the stream and signal is finished.
1663 if ( handle->drainCounter > 3 ) {
1664 ThreadHandle threadId;
1666 stream_.state = STREAM_STOPPING;
1667 if ( handle->internalDrain == true )
1668 pthread_create( &threadId, NULL, coreStopStream, info );
1669 else // external call to stopStream()
1670 pthread_cond_signal( &handle->condition );
1674 AudioDeviceID outputDevice = handle->id[0];
1676 // Invoke user callback to get fresh output data UNLESS we are
1677 // draining stream or duplex mode AND the input/output devices are
1678 // different AND this function is called for the input device.
1679 if ( handle->drainCounter == 0 && ( stream_.mode != DUPLEX || deviceId == outputDevice ) ) {
1680 RtAudioCallback callback = (RtAudioCallback) info->callback;
1681 double streamTime = getStreamTime();
1682 RtAudioStreamStatus status = 0;
1683 if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
1684 status |= RTAUDIO_OUTPUT_UNDERFLOW;
1685 handle->xrun[0] = false;
1687 if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
1688 status |= RTAUDIO_INPUT_OVERFLOW;
1689 handle->xrun[1] = false;
1692 int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
1693 stream_.bufferSize, streamTime, status, info->userData );
1694 if ( cbReturnValue == 2 ) {
1695 stream_.state = STREAM_STOPPING;
1696 handle->drainCounter = 2;
1700 else if ( cbReturnValue == 1 ) {
1701 handle->drainCounter = 1;
1702 handle->internalDrain = true;
1706 if ( stream_.mode == OUTPUT || ( stream_.mode == DUPLEX && deviceId == outputDevice ) ) {
1708 if ( handle->drainCounter > 1 ) { // write zeros to the output stream
1710 if ( handle->nStreams[0] == 1 ) {
1711 memset( outBufferList->mBuffers[handle->iStream[0]].mData,
1713 outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );
1715 else { // fill multiple streams with zeros
1716 for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) {
1717 memset( outBufferList->mBuffers[handle->iStream[0]+i].mData,
1719 outBufferList->mBuffers[handle->iStream[0]+i].mDataByteSize );
1723 else if ( handle->nStreams[0] == 1 ) {
1724 if ( stream_.doConvertBuffer[0] ) { // convert directly to CoreAudio stream buffer
1725 convertBuffer( (char *) outBufferList->mBuffers[handle->iStream[0]].mData,
1726 stream_.userBuffer[0], stream_.convertInfo[0] );
1728 else { // copy from user buffer
1729 memcpy( outBufferList->mBuffers[handle->iStream[0]].mData,
1730 stream_.userBuffer[0],
1731 outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );
1734 else { // fill multiple streams
1735 Float32 *inBuffer = (Float32 *) stream_.userBuffer[0];
1736 if ( stream_.doConvertBuffer[0] ) {
1737 convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
1738 inBuffer = (Float32 *) stream_.deviceBuffer;
1741 if ( stream_.deviceInterleaved[0] == false ) { // mono mode
1742 UInt32 bufferBytes = outBufferList->mBuffers[handle->iStream[0]].mDataByteSize;
1743 for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
1744 memcpy( outBufferList->mBuffers[handle->iStream[0]+i].mData,
1745 (void *)&inBuffer[i*stream_.bufferSize], bufferBytes );
1748 else { // fill multiple multi-channel streams with interleaved data
1749 UInt32 streamChannels, channelsLeft, inJump, outJump, inOffset;
1752 bool inInterleaved = ( stream_.userInterleaved ) ? true : false;
1753 UInt32 inChannels = stream_.nUserChannels[0];
1754 if ( stream_.doConvertBuffer[0] ) {
1755 inInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode
1756 inChannels = stream_.nDeviceChannels[0];
1759 if ( inInterleaved ) inOffset = 1;
1760 else inOffset = stream_.bufferSize;
1762 channelsLeft = inChannels;
1763 for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) {
1765 out = (Float32 *) outBufferList->mBuffers[handle->iStream[0]+i].mData;
1766 streamChannels = outBufferList->mBuffers[handle->iStream[0]+i].mNumberChannels;
1769 // Account for possible channel offset in first stream
1770 if ( i == 0 && stream_.channelOffset[0] > 0 ) {
1771 streamChannels -= stream_.channelOffset[0];
1772 outJump = stream_.channelOffset[0];
1776 // Account for possible unfilled channels at end of the last stream
1777 if ( streamChannels > channelsLeft ) {
1778 outJump = streamChannels - channelsLeft;
1779 streamChannels = channelsLeft;
1782 // Determine input buffer offsets and skips
1783 if ( inInterleaved ) {
1784 inJump = inChannels;
1785 in += inChannels - channelsLeft;
1789 in += (inChannels - channelsLeft) * inOffset;
1792 for ( unsigned int i=0; i<stream_.bufferSize; i++ ) {
1793 for ( unsigned int j=0; j<streamChannels; j++ ) {
1794 *out++ = in[j*inOffset];
1799 channelsLeft -= streamChannels;
1805 // Don't bother draining input
1806 if ( handle->drainCounter ) {
1807 handle->drainCounter++;
1811 AudioDeviceID inputDevice;
1812 inputDevice = handle->id[1];
1813 if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && deviceId == inputDevice ) ) {
1815 if ( handle->nStreams[1] == 1 ) {
1816 if ( stream_.doConvertBuffer[1] ) { // convert directly from CoreAudio stream buffer
1817 convertBuffer( stream_.userBuffer[1],
1818 (char *) inBufferList->mBuffers[handle->iStream[1]].mData,
1819 stream_.convertInfo[1] );
1821 else { // copy to user buffer
1822 memcpy( stream_.userBuffer[1],
1823 inBufferList->mBuffers[handle->iStream[1]].mData,
1824 inBufferList->mBuffers[handle->iStream[1]].mDataByteSize );
1827 else { // read from multiple streams
1828 Float32 *outBuffer = (Float32 *) stream_.userBuffer[1];
1829 if ( stream_.doConvertBuffer[1] ) outBuffer = (Float32 *) stream_.deviceBuffer;
1831 if ( stream_.deviceInterleaved[1] == false ) { // mono mode
1832 UInt32 bufferBytes = inBufferList->mBuffers[handle->iStream[1]].mDataByteSize;
1833 for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
1834 memcpy( (void *)&outBuffer[i*stream_.bufferSize],
1835 inBufferList->mBuffers[handle->iStream[1]+i].mData, bufferBytes );
1838 else { // read from multiple multi-channel streams
1839 UInt32 streamChannels, channelsLeft, inJump, outJump, outOffset;
1842 bool outInterleaved = ( stream_.userInterleaved ) ? true : false;
1843 UInt32 outChannels = stream_.nUserChannels[1];
1844 if ( stream_.doConvertBuffer[1] ) {
1845 outInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode
1846 outChannels = stream_.nDeviceChannels[1];
1849 if ( outInterleaved ) outOffset = 1;
1850 else outOffset = stream_.bufferSize;
1852 channelsLeft = outChannels;
1853 for ( unsigned int i=0; i<handle->nStreams[1]; i++ ) {
1855 in = (Float32 *) inBufferList->mBuffers[handle->iStream[1]+i].mData;
1856 streamChannels = inBufferList->mBuffers[handle->iStream[1]+i].mNumberChannels;
1859 // Account for possible channel offset in first stream
1860 if ( i == 0 && stream_.channelOffset[1] > 0 ) {
1861 streamChannels -= stream_.channelOffset[1];
1862 inJump = stream_.channelOffset[1];
1866 // Account for possible unread channels at end of the last stream
1867 if ( streamChannels > channelsLeft ) {
1868 inJump = streamChannels - channelsLeft;
1869 streamChannels = channelsLeft;
1872 // Determine output buffer offsets and skips
1873 if ( outInterleaved ) {
1874 outJump = outChannels;
1875 out += outChannels - channelsLeft;
1879 out += (outChannels - channelsLeft) * outOffset;
1882 for ( unsigned int i=0; i<stream_.bufferSize; i++ ) {
1883 for ( unsigned int j=0; j<streamChannels; j++ ) {
1884 out[j*outOffset] = *in++;
1889 channelsLeft -= streamChannels;
1893 if ( stream_.doConvertBuffer[1] ) { // convert from our internal "device" buffer
1894 convertBuffer( stream_.userBuffer[1],
1895 stream_.deviceBuffer,
1896 stream_.convertInfo[1] );
1902 //MUTEX_UNLOCK( &stream_.mutex );
1904 RtApi::tickStreamTime();
1908 const char* RtApiCore :: getErrorCode( OSStatus code )
1912 case kAudioHardwareNotRunningError:
1913 return "kAudioHardwareNotRunningError";
1915 case kAudioHardwareUnspecifiedError:
1916 return "kAudioHardwareUnspecifiedError";
1918 case kAudioHardwareUnknownPropertyError:
1919 return "kAudioHardwareUnknownPropertyError";
1921 case kAudioHardwareBadPropertySizeError:
1922 return "kAudioHardwareBadPropertySizeError";
1924 case kAudioHardwareIllegalOperationError:
1925 return "kAudioHardwareIllegalOperationError";
1927 case kAudioHardwareBadObjectError:
1928 return "kAudioHardwareBadObjectError";
1930 case kAudioHardwareBadDeviceError:
1931 return "kAudioHardwareBadDeviceError";
1933 case kAudioHardwareBadStreamError:
1934 return "kAudioHardwareBadStreamError";
1936 case kAudioHardwareUnsupportedOperationError:
1937 return "kAudioHardwareUnsupportedOperationError";
1939 case kAudioDeviceUnsupportedFormatError:
1940 return "kAudioDeviceUnsupportedFormatError";
1942 case kAudioDevicePermissionsError:
1943 return "kAudioDevicePermissionsError";
1946 return "CoreAudio unknown error";
1950 //******************** End of __MACOSX_CORE__ *********************//
1953 #if defined(__UNIX_JACK__)
1955 // JACK is a low-latency audio server, originally written for the
1956 // GNU/Linux operating system and now also ported to OS-X. It can
1957 // connect a number of different applications to an audio device, as
1958 // well as allowing them to share audio between themselves.
1960 // When using JACK with RtAudio, "devices" refer to JACK clients that
1961 // have ports connected to the server. The JACK server is typically
1962 // started in a terminal as follows:
1964 // .jackd -d alsa -d hw:0
1966 // or through an interface program such as qjackctl. Many of the
1967 // parameters normally set for a stream are fixed by the JACK server
1968 // and can be specified when the JACK server is started. In
1971 // .jackd -d alsa -d hw:0 -r 44100 -p 512 -n 4
1973 // specifies a sample rate of 44100 Hz, a buffer size of 512 sample
1974 // frames, and number of buffers = 4. Once the server is running, it
1975 // is not possible to override these values. If the values are not
1976 // specified in the command-line, the JACK server uses default values.
1978 // The JACK server does not have to be running when an instance of
1979 // RtApiJack is created, though the function getDeviceCount() will
1980 // report 0 devices found until JACK has been started. When no
1981 // devices are available (i.e., the JACK server is not running), a
1982 // stream cannot be opened.
1984 #include <jack/jack.h>
1988 // A structure to hold various information related to the Jack API
1991 jack_client_t *client;
1992 jack_port_t **ports[2];
1993 std::string deviceName[2];
1995 pthread_cond_t condition;
1996 int drainCounter; // Tracks callback counts when draining
1997 bool internalDrain; // Indicates if stop is initiated from callback or not.
2000 :client(0), drainCounter(0), internalDrain(false) { ports[0] = 0; ports[1] = 0; xrun[0] = false; xrun[1] = false; }
2003 #if !defined(__RTAUDIO_DEBUG__)
2004 static void jackSilentError( const char * ) {};
2007 RtApiJack :: RtApiJack()
2008 :shouldAutoconnect_(true) {
2009 // Nothing to do here.
2010 #if !defined(__RTAUDIO_DEBUG__)
2011 // Turn off Jack's internal error reporting.
2012 jack_set_error_function( &jackSilentError );
2016 RtApiJack :: ~RtApiJack()
2018 if ( stream_.state != STREAM_CLOSED ) closeStream();
2021 unsigned int RtApiJack :: getDeviceCount( void )
2023 // See if we can become a jack client.
2024 jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption;
2025 jack_status_t *status = NULL;
2026 jack_client_t *client = jack_client_open( "RtApiJackCount", options, status );
2027 if ( client == 0 ) return 0;
2030 std::string port, previousPort;
2031 unsigned int nChannels = 0, nDevices = 0;
2032 ports = jack_get_ports( client, NULL, JACK_DEFAULT_AUDIO_TYPE, 0 );
2034 // Parse the port names up to the first colon (:).
2037 port = (char *) ports[ nChannels ];
2038 iColon = port.find(":");
2039 if ( iColon != std::string::npos ) {
2040 port = port.substr( 0, iColon + 1 );
2041 if ( port != previousPort ) {
2043 previousPort = port;
2046 } while ( ports[++nChannels] );
2050 jack_client_close( client );
2054 RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device )
2056 RtAudio::DeviceInfo info;
2057 info.probed = false;
2059 jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption
2060 jack_status_t *status = NULL;
2061 jack_client_t *client = jack_client_open( "RtApiJackInfo", options, status );
2062 if ( client == 0 ) {
2063 errorText_ = "RtApiJack::getDeviceInfo: Jack server not found or connection error!";
2064 error( RtAudioError::WARNING );
2069 std::string port, previousPort;
2070 unsigned int nPorts = 0, nDevices = 0;
2071 ports = jack_get_ports( client, NULL, JACK_DEFAULT_AUDIO_TYPE, 0 );
2073 // Parse the port names up to the first colon (:).
2076 port = (char *) ports[ nPorts ];
2077 iColon = port.find(":");
2078 if ( iColon != std::string::npos ) {
2079 port = port.substr( 0, iColon );
2080 if ( port != previousPort ) {
2081 if ( nDevices == device ) info.name = port;
2083 previousPort = port;
2086 } while ( ports[++nPorts] );
2090 if ( device >= nDevices ) {
2091 jack_client_close( client );
2092 errorText_ = "RtApiJack::getDeviceInfo: device ID is invalid!";
2093 error( RtAudioError::INVALID_USE );
2097 // Get the current jack server sample rate.
2098 info.sampleRates.clear();
2100 info.preferredSampleRate = jack_get_sample_rate( client );
2101 info.sampleRates.push_back( info.preferredSampleRate );
2103 // Count the available ports containing the client name as device
2104 // channels. Jack "input ports" equal RtAudio output channels.
2105 unsigned int nChannels = 0;
2106 ports = jack_get_ports( client, info.name.c_str(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput );
2108 while ( ports[ nChannels ] ) nChannels++;
2110 info.outputChannels = nChannels;
2113 // Jack "output ports" equal RtAudio input channels.
2115 ports = jack_get_ports( client, info.name.c_str(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput );
2117 while ( ports[ nChannels ] ) nChannels++;
2119 info.inputChannels = nChannels;
2122 if ( info.outputChannels == 0 && info.inputChannels == 0 ) {
2123 jack_client_close(client);
2124 errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!";
2125 error( RtAudioError::WARNING );
2129 // If device opens for both playback and capture, we determine the channels.
2130 if ( info.outputChannels > 0 && info.inputChannels > 0 )
2131 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
2133 // Jack always uses 32-bit floats.
2134 info.nativeFormats = RTAUDIO_FLOAT32;
2136 // Jack doesn't provide default devices so we'll use the first available one.
2137 if ( device == 0 && info.outputChannels > 0 )
2138 info.isDefaultOutput = true;
2139 if ( device == 0 && info.inputChannels > 0 )
2140 info.isDefaultInput = true;
2142 jack_client_close(client);
2147 static int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer )
2149 CallbackInfo *info = (CallbackInfo *) infoPointer;
2151 RtApiJack *object = (RtApiJack *) info->object;
2152 if ( object->callbackEvent( (unsigned long) nframes ) == false ) return 1;
2157 // This function will be called by a spawned thread when the Jack
2158 // server signals that it is shutting down. It is necessary to handle
2159 // it this way because the jackShutdown() function must return before
2160 // the jack_deactivate() function (in closeStream()) will return.
2161 static void *jackCloseStream( void *ptr )
2163 CallbackInfo *info = (CallbackInfo *) ptr;
2164 RtApiJack *object = (RtApiJack *) info->object;
2166 object->closeStream();
2168 pthread_exit( NULL );
2170 static void jackShutdown( void *infoPointer )
2172 CallbackInfo *info = (CallbackInfo *) infoPointer;
2173 RtApiJack *object = (RtApiJack *) info->object;
2175 // Check current stream state. If stopped, then we'll assume this
2176 // was called as a result of a call to RtApiJack::stopStream (the
2177 // deactivation of a client handle causes this function to be called).
2178 // If not, we'll assume the Jack server is shutting down or some
2179 // other problem occurred and we should close the stream.
2180 if ( object->isStreamRunning() == false ) return;
2182 ThreadHandle threadId;
2183 pthread_create( &threadId, NULL, jackCloseStream, info );
2184 std::cerr << "\nRtApiJack: the Jack server is shutting down this client ... stream stopped and closed!!\n" << std::endl;
2187 static int jackXrun( void *infoPointer )
2189 JackHandle *handle = *((JackHandle **) infoPointer);
2191 if ( handle->ports[0] ) handle->xrun[0] = true;
2192 if ( handle->ports[1] ) handle->xrun[1] = true;
2197 bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
2198 unsigned int firstChannel, unsigned int sampleRate,
2199 RtAudioFormat format, unsigned int *bufferSize,
2200 RtAudio::StreamOptions *options )
2202 JackHandle *handle = (JackHandle *) stream_.apiHandle;
2204 // Look for jack server and try to become a client (only do once per stream).
2205 jack_client_t *client = 0;
2206 if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) {
2207 jack_options_t jackoptions = (jack_options_t) ( JackNoStartServer ); //JackNullOption;
2208 jack_status_t *status = NULL;
2209 if ( options && !options->streamName.empty() )
2210 client = jack_client_open( options->streamName.c_str(), jackoptions, status );
2212 client = jack_client_open( "RtApiJack", jackoptions, status );
2213 if ( client == 0 ) {
2214 errorText_ = "RtApiJack::probeDeviceOpen: Jack server not found or connection error!";
2215 error( RtAudioError::WARNING );
2220 // The handle must have been created on an earlier pass.
2221 client = handle->client;
2225 std::string port, previousPort, deviceName;
2226 unsigned int nPorts = 0, nDevices = 0;
2227 ports = jack_get_ports( client, NULL, JACK_DEFAULT_AUDIO_TYPE, 0 );
2229 // Parse the port names up to the first colon (:).
2232 port = (char *) ports[ nPorts ];
2233 iColon = port.find(":");
2234 if ( iColon != std::string::npos ) {
2235 port = port.substr( 0, iColon );
2236 if ( port != previousPort ) {
2237 if ( nDevices == device ) deviceName = port;
2239 previousPort = port;
2242 } while ( ports[++nPorts] );
2246 if ( device >= nDevices ) {
2247 errorText_ = "RtApiJack::probeDeviceOpen: device ID is invalid!";
2251 unsigned long flag = JackPortIsInput;
2252 if ( mode == INPUT ) flag = JackPortIsOutput;
2254 if ( ! (options && (options->flags & RTAUDIO_JACK_DONT_CONNECT)) ) {
2255 // Count the available ports containing the client name as device
2256 // channels. Jack "input ports" equal RtAudio output channels.
2257 unsigned int nChannels = 0;
2258 ports = jack_get_ports( client, deviceName.c_str(), JACK_DEFAULT_AUDIO_TYPE, flag );
2260 while ( ports[ nChannels ] ) nChannels++;
2263 // Compare the jack ports for specified client to the requested number of channels.
2264 if ( nChannels < (channels + firstChannel) ) {
2265 errorStream_ << "RtApiJack::probeDeviceOpen: requested number of channels (" << channels << ") + offset (" << firstChannel << ") not found for specified device (" << device << ":" << deviceName << ").";
2266 errorText_ = errorStream_.str();
2271 // Check the jack server sample rate.
2272 unsigned int jackRate = jack_get_sample_rate( client );
2273 if ( sampleRate != jackRate ) {
2274 jack_client_close( client );
2275 errorStream_ << "RtApiJack::probeDeviceOpen: the requested sample rate (" << sampleRate << ") is different than the JACK server rate (" << jackRate << ").";
2276 errorText_ = errorStream_.str();
2279 stream_.sampleRate = jackRate;
2281 // Get the latency of the JACK port.
2282 ports = jack_get_ports( client, deviceName.c_str(), JACK_DEFAULT_AUDIO_TYPE, flag );
2283 if ( ports[ firstChannel ] ) {
2285 jack_latency_callback_mode_t cbmode = (mode == INPUT ? JackCaptureLatency : JackPlaybackLatency);
2286 // the range (usually the min and max are equal)
2287 jack_latency_range_t latrange; latrange.min = latrange.max = 0;
2288 // get the latency range
2289 jack_port_get_latency_range( jack_port_by_name( client, ports[firstChannel] ), cbmode, &latrange );
2290 // be optimistic, use the min!
2291 stream_.latency[mode] = latrange.min;
2292 //stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) );
2296 // The jack server always uses 32-bit floating-point data.
2297 stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
2298 stream_.userFormat = format;
2300 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
2301 else stream_.userInterleaved = true;
2303 // Jack always uses non-interleaved buffers.
2304 stream_.deviceInterleaved[mode] = false;
2306 // Jack always provides host byte-ordered data.
2307 stream_.doByteSwap[mode] = false;
2309 // Get the buffer size. The buffer size and number of buffers
2310 // (periods) is set when the jack server is started.
2311 stream_.bufferSize = (int) jack_get_buffer_size( client );
2312 *bufferSize = stream_.bufferSize;
2314 stream_.nDeviceChannels[mode] = channels;
2315 stream_.nUserChannels[mode] = channels;
2317 // Set flags for buffer conversion.
2318 stream_.doConvertBuffer[mode] = false;
2319 if ( stream_.userFormat != stream_.deviceFormat[mode] )
2320 stream_.doConvertBuffer[mode] = true;
2321 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
2322 stream_.nUserChannels[mode] > 1 )
2323 stream_.doConvertBuffer[mode] = true;
2325 // Allocate our JackHandle structure for the stream.
2326 if ( handle == 0 ) {
2328 handle = new JackHandle;
2330 catch ( std::bad_alloc& ) {
2331 errorText_ = "RtApiJack::probeDeviceOpen: error allocating JackHandle memory.";
2335 if ( pthread_cond_init(&handle->condition, NULL) ) {
2336 errorText_ = "RtApiJack::probeDeviceOpen: error initializing pthread condition variable.";
2339 stream_.apiHandle = (void *) handle;
2340 handle->client = client;
2342 handle->deviceName[mode] = deviceName;
2344 // Allocate necessary internal buffers.
2345 unsigned long bufferBytes;
2346 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
2347 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
2348 if ( stream_.userBuffer[mode] == NULL ) {
2349 errorText_ = "RtApiJack::probeDeviceOpen: error allocating user buffer memory.";
2353 if ( stream_.doConvertBuffer[mode] ) {
2355 bool makeBuffer = true;
2356 if ( mode == OUTPUT )
2357 bufferBytes = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
2358 else { // mode == INPUT
2359 bufferBytes = stream_.nDeviceChannels[1] * formatBytes( stream_.deviceFormat[1] );
2360 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
2361 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]);
2362 if ( bufferBytes < bytesOut ) makeBuffer = false;
2367 bufferBytes *= *bufferSize;
2368 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
2369 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
2370 if ( stream_.deviceBuffer == NULL ) {
2371 errorText_ = "RtApiJack::probeDeviceOpen: error allocating device buffer memory.";
2377 // Allocate memory for the Jack ports (channels) identifiers.
2378 handle->ports[mode] = (jack_port_t **) malloc ( sizeof (jack_port_t *) * channels );
2379 if ( handle->ports[mode] == NULL ) {
2380 errorText_ = "RtApiJack::probeDeviceOpen: error allocating port memory.";
2384 stream_.device[mode] = device;
2385 stream_.channelOffset[mode] = firstChannel;
2386 stream_.state = STREAM_STOPPED;
2387 stream_.callbackInfo.object = (void *) this;
2389 if ( stream_.mode == OUTPUT && mode == INPUT )
2390 // We had already set up the stream for output.
2391 stream_.mode = DUPLEX;
2393 stream_.mode = mode;
2394 jack_set_process_callback( handle->client, jackCallbackHandler, (void *) &stream_.callbackInfo );
2395 jack_set_xrun_callback( handle->client, jackXrun, (void *) &stream_.apiHandle );
2396 jack_on_shutdown( handle->client, jackShutdown, (void *) &stream_.callbackInfo );
2399 // Register our ports.
2401 if ( mode == OUTPUT ) {
2402 for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
2403 snprintf( label, 64, "outport %d", i );
2404 handle->ports[0][i] = jack_port_register( handle->client, (const char *)label,
2405 JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0 );
2409 for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
2410 snprintf( label, 64, "inport %d", i );
2411 handle->ports[1][i] = jack_port_register( handle->client, (const char *)label,
2412 JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0 );
2416 // Setup the buffer conversion information structure. We don't use
2417 // buffers to do channel offsets, so we override that parameter
2419 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );
2421 if ( options && options->flags & RTAUDIO_JACK_DONT_CONNECT ) shouldAutoconnect_ = false;
2427 pthread_cond_destroy( &handle->condition );
2428 jack_client_close( handle->client );
2430 if ( handle->ports[0] ) free( handle->ports[0] );
2431 if ( handle->ports[1] ) free( handle->ports[1] );
2434 stream_.apiHandle = 0;
2437 for ( int i=0; i<2; i++ ) {
2438 if ( stream_.userBuffer[i] ) {
2439 free( stream_.userBuffer[i] );
2440 stream_.userBuffer[i] = 0;
2444 if ( stream_.deviceBuffer ) {
2445 free( stream_.deviceBuffer );
2446 stream_.deviceBuffer = 0;
2452 void RtApiJack :: closeStream( void )
2454 if ( stream_.state == STREAM_CLOSED ) {
2455 errorText_ = "RtApiJack::closeStream(): no open stream to close!";
2456 error( RtAudioError::WARNING );
2460 JackHandle *handle = (JackHandle *) stream_.apiHandle;
2463 if ( stream_.state == STREAM_RUNNING )
2464 jack_deactivate( handle->client );
2466 jack_client_close( handle->client );
2470 if ( handle->ports[0] ) free( handle->ports[0] );
2471 if ( handle->ports[1] ) free( handle->ports[1] );
2472 pthread_cond_destroy( &handle->condition );
2474 stream_.apiHandle = 0;
2477 for ( int i=0; i<2; i++ ) {
2478 if ( stream_.userBuffer[i] ) {
2479 free( stream_.userBuffer[i] );
2480 stream_.userBuffer[i] = 0;
2484 if ( stream_.deviceBuffer ) {
2485 free( stream_.deviceBuffer );
2486 stream_.deviceBuffer = 0;
2489 stream_.mode = UNINITIALIZED;
2490 stream_.state = STREAM_CLOSED;
2493 void RtApiJack :: startStream( void )
2496 if ( stream_.state == STREAM_RUNNING ) {
2497 errorText_ = "RtApiJack::startStream(): the stream is already running!";
2498 error( RtAudioError::WARNING );
2502 JackHandle *handle = (JackHandle *) stream_.apiHandle;
2503 int result = jack_activate( handle->client );
2505 errorText_ = "RtApiJack::startStream(): unable to activate JACK client!";
2511 // Get the list of available ports.
2512 if ( shouldAutoconnect_ && (stream_.mode == OUTPUT || stream_.mode == DUPLEX) ) {
2514 ports = jack_get_ports( handle->client, handle->deviceName[0].c_str(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput);
2515 if ( ports == NULL) {
2516 errorText_ = "RtApiJack::startStream(): error determining available JACK input ports!";
2520 // Now make the port connections. Since RtAudio wasn't designed to
2521 // allow the user to select particular channels of a device, we'll
2522 // just open the first "nChannels" ports with offset.
2523 for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
2525 if ( ports[ stream_.channelOffset[0] + i ] )
2526 result = jack_connect( handle->client, jack_port_name( handle->ports[0][i] ), ports[ stream_.channelOffset[0] + i ] );
2529 errorText_ = "RtApiJack::startStream(): error connecting output ports!";
2536 if ( shouldAutoconnect_ && (stream_.mode == INPUT || stream_.mode == DUPLEX) ) {
2538 ports = jack_get_ports( handle->client, handle->deviceName[1].c_str(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput );
2539 if ( ports == NULL) {
2540 errorText_ = "RtApiJack::startStream(): error determining available JACK output ports!";
2544 // Now make the port connections. See note above.
2545 for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
2547 if ( ports[ stream_.channelOffset[1] + i ] )
2548 result = jack_connect( handle->client, ports[ stream_.channelOffset[1] + i ], jack_port_name( handle->ports[1][i] ) );
2551 errorText_ = "RtApiJack::startStream(): error connecting input ports!";
2558 handle->drainCounter = 0;
2559 handle->internalDrain = false;
2560 stream_.state = STREAM_RUNNING;
2563 if ( result == 0 ) return;
2564 error( RtAudioError::SYSTEM_ERROR );
2567 void RtApiJack :: stopStream( void )
2570 if ( stream_.state == STREAM_STOPPED ) {
2571 errorText_ = "RtApiJack::stopStream(): the stream is already stopped!";
2572 error( RtAudioError::WARNING );
2576 JackHandle *handle = (JackHandle *) stream_.apiHandle;
2577 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
2579 if ( handle->drainCounter == 0 ) {
2580 handle->drainCounter = 2;
2581 pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
2585 jack_deactivate( handle->client );
2586 stream_.state = STREAM_STOPPED;
2589 void RtApiJack :: abortStream( void )
2592 if ( stream_.state == STREAM_STOPPED ) {
2593 errorText_ = "RtApiJack::abortStream(): the stream is already stopped!";
2594 error( RtAudioError::WARNING );
2598 JackHandle *handle = (JackHandle *) stream_.apiHandle;
2599 handle->drainCounter = 2;
2604 // This function will be called by a spawned thread when the user
2605 // callback function signals that the stream should be stopped or
2606 // aborted. It is necessary to handle it this way because the
2607 // callbackEvent() function must return before the jack_deactivate()
2608 // function will return.
2609 static void *jackStopStream( void *ptr )
2611 CallbackInfo *info = (CallbackInfo *) ptr;
2612 RtApiJack *object = (RtApiJack *) info->object;
2614 object->stopStream();
2615 pthread_exit( NULL );
2618 bool RtApiJack :: callbackEvent( unsigned long nframes )
2620 if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;
2621 if ( stream_.state == STREAM_CLOSED ) {
2622 errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";
2623 error( RtAudioError::WARNING );
2626 if ( stream_.bufferSize != nframes ) {
2627 errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!";
2628 error( RtAudioError::WARNING );
2632 CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
2633 JackHandle *handle = (JackHandle *) stream_.apiHandle;
2635 // Check if we were draining the stream and signal is finished.
2636 if ( handle->drainCounter > 3 ) {
2637 ThreadHandle threadId;
2639 stream_.state = STREAM_STOPPING;
2640 if ( handle->internalDrain == true )
2641 pthread_create( &threadId, NULL, jackStopStream, info );
2643 pthread_cond_signal( &handle->condition );
2647 // Invoke user callback first, to get fresh output data.
2648 if ( handle->drainCounter == 0 ) {
2649 RtAudioCallback callback = (RtAudioCallback) info->callback;
2650 double streamTime = getStreamTime();
2651 RtAudioStreamStatus status = 0;
2652 if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
2653 status |= RTAUDIO_OUTPUT_UNDERFLOW;
2654 handle->xrun[0] = false;
2656 if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
2657 status |= RTAUDIO_INPUT_OVERFLOW;
2658 handle->xrun[1] = false;
2660 int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
2661 stream_.bufferSize, streamTime, status, info->userData );
2662 if ( cbReturnValue == 2 ) {
2663 stream_.state = STREAM_STOPPING;
2664 handle->drainCounter = 2;
2666 pthread_create( &id, NULL, jackStopStream, info );
2669 else if ( cbReturnValue == 1 ) {
2670 handle->drainCounter = 1;
2671 handle->internalDrain = true;
2675 jack_default_audio_sample_t *jackbuffer;
2676 unsigned long bufferBytes = nframes * sizeof( jack_default_audio_sample_t );
2677 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
2679 if ( handle->drainCounter > 1 ) { // write zeros to the output stream
2681 for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
2682 jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
2683 memset( jackbuffer, 0, bufferBytes );
2687 else if ( stream_.doConvertBuffer[0] ) {
2689 convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
2691 for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
2692 jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
2693 memcpy( jackbuffer, &stream_.deviceBuffer[i*bufferBytes], bufferBytes );
2696 else { // no buffer conversion
2697 for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
2698 jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
2699 memcpy( jackbuffer, &stream_.userBuffer[0][i*bufferBytes], bufferBytes );
2704 // Don't bother draining input
2705 if ( handle->drainCounter ) {
2706 handle->drainCounter++;
2710 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
2712 if ( stream_.doConvertBuffer[1] ) {
2713 for ( unsigned int i=0; i<stream_.nDeviceChannels[1]; i++ ) {
2714 jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );
2715 memcpy( &stream_.deviceBuffer[i*bufferBytes], jackbuffer, bufferBytes );
2717 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
2719 else { // no buffer conversion
2720 for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
2721 jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );
2722 memcpy( &stream_.userBuffer[1][i*bufferBytes], jackbuffer, bufferBytes );
2728 RtApi::tickStreamTime();
2731 //******************** End of __UNIX_JACK__ *********************//
2734 #if defined(__WINDOWS_ASIO__) // ASIO API on Windows
2736 // The ASIO API is designed around a callback scheme, so this
2737 // implementation is similar to that used for OS-X CoreAudio and Linux
2738 // Jack. The primary constraint with ASIO is that it only allows
2739 // access to a single driver at a time. Thus, it is not possible to
2740 // have more than one simultaneous RtAudio stream.
2742 // This implementation also requires a number of external ASIO files
2743 // and a few global variables. The ASIO callback scheme does not
2744 // allow for the passing of user data, so we must create a global
2745 // pointer to our callbackInfo structure.
2747 // On unix systems, we make use of a pthread condition variable.
2748 // Since there is no equivalent in Windows, I hacked something based
2749 // on information found in
2750 // http://www.cs.wustl.edu/~schmidt/win32-cv-1.html.
2752 #include "asiosys.h"
2754 #include "iasiothiscallresolver.h"
2755 #include "asiodrivers.h"
2758 static AsioDrivers drivers;
2759 static ASIOCallbacks asioCallbacks;
2760 static ASIODriverInfo driverInfo;
2761 static CallbackInfo *asioCallbackInfo;
2762 static bool asioXRun;
2765 int drainCounter; // Tracks callback counts when draining
2766 bool internalDrain; // Indicates if stop is initiated from callback or not.
2767 ASIOBufferInfo *bufferInfos;
2771 :drainCounter(0), internalDrain(false), bufferInfos(0) {}
2774 // Function declarations (definitions at end of section)
2775 static const char* getAsioErrorString( ASIOError result );
2776 static void sampleRateChanged( ASIOSampleRate sRate );
2777 static long asioMessages( long selector, long value, void* message, double* opt );
2779 RtApiAsio :: RtApiAsio()
2781 // ASIO cannot run on a multi-threaded appartment. You can call
2782 // CoInitialize beforehand, but it must be for appartment threading
2783 // (in which case, CoInitilialize will return S_FALSE here).
2784 coInitialized_ = false;
2785 HRESULT hr = CoInitialize( NULL );
2787 errorText_ = "RtApiAsio::ASIO requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)";
2788 error( RtAudioError::WARNING );
2790 coInitialized_ = true;
2792 drivers.removeCurrentDriver();
2793 driverInfo.asioVersion = 2;
2795 // See note in DirectSound implementation about GetDesktopWindow().
2796 driverInfo.sysRef = GetForegroundWindow();
2799 RtApiAsio :: ~RtApiAsio()
2801 if ( stream_.state != STREAM_CLOSED ) closeStream();
2802 if ( coInitialized_ ) CoUninitialize();
2805 unsigned int RtApiAsio :: getDeviceCount( void )
2807 return (unsigned int) drivers.asioGetNumDev();
2810 RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device )
2812 RtAudio::DeviceInfo info;
2813 info.probed = false;
2816 unsigned int nDevices = getDeviceCount();
2817 if ( nDevices == 0 ) {
2818 errorText_ = "RtApiAsio::getDeviceInfo: no devices found!";
2819 error( RtAudioError::INVALID_USE );
2823 if ( device >= nDevices ) {
2824 errorText_ = "RtApiAsio::getDeviceInfo: device ID is invalid!";
2825 error( RtAudioError::INVALID_USE );
2829 // If a stream is already open, we cannot probe other devices. Thus, use the saved results.
2830 if ( stream_.state != STREAM_CLOSED ) {
2831 if ( device >= devices_.size() ) {
2832 errorText_ = "RtApiAsio::getDeviceInfo: device ID was not present before stream was opened.";
2833 error( RtAudioError::WARNING );
2836 return devices_[ device ];
2839 char driverName[32];
2840 ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 );
2841 if ( result != ASE_OK ) {
2842 errorStream_ << "RtApiAsio::getDeviceInfo: unable to get driver name (" << getAsioErrorString( result ) << ").";
2843 errorText_ = errorStream_.str();
2844 error( RtAudioError::WARNING );
2848 info.name = driverName;
2850 if ( !drivers.loadDriver( driverName ) ) {
2851 errorStream_ << "RtApiAsio::getDeviceInfo: unable to load driver (" << driverName << ").";
2852 errorText_ = errorStream_.str();
2853 error( RtAudioError::WARNING );
2857 result = ASIOInit( &driverInfo );
2858 if ( result != ASE_OK ) {
2859 errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
2860 errorText_ = errorStream_.str();
2861 error( RtAudioError::WARNING );
2865 // Determine the device channel information.
2866 long inputChannels, outputChannels;
2867 result = ASIOGetChannels( &inputChannels, &outputChannels );
2868 if ( result != ASE_OK ) {
2869 drivers.removeCurrentDriver();
2870 errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
2871 errorText_ = errorStream_.str();
2872 error( RtAudioError::WARNING );
2876 info.outputChannels = outputChannels;
2877 info.inputChannels = inputChannels;
2878 if ( info.outputChannels > 0 && info.inputChannels > 0 )
2879 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
2881 // Determine the supported sample rates.
2882 info.sampleRates.clear();
2883 for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {
2884 result = ASIOCanSampleRate( (ASIOSampleRate) SAMPLE_RATES[i] );
2885 if ( result == ASE_OK ) {
2886 info.sampleRates.push_back( SAMPLE_RATES[i] );
2888 if ( !info.preferredSampleRate || ( SAMPLE_RATES[i] <= 48000 && SAMPLE_RATES[i] > info.preferredSampleRate ) )
2889 info.preferredSampleRate = SAMPLE_RATES[i];
2893 // Determine supported data types ... just check first channel and assume rest are the same.
2894 ASIOChannelInfo channelInfo;
2895 channelInfo.channel = 0;
2896 channelInfo.isInput = true;
2897 if ( info.inputChannels <= 0 ) channelInfo.isInput = false;
2898 result = ASIOGetChannelInfo( &channelInfo );
2899 if ( result != ASE_OK ) {
2900 drivers.removeCurrentDriver();
2901 errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting driver channel info (" << driverName << ").";
2902 errorText_ = errorStream_.str();
2903 error( RtAudioError::WARNING );
2907 info.nativeFormats = 0;
2908 if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB )
2909 info.nativeFormats |= RTAUDIO_SINT16;
2910 else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB )
2911 info.nativeFormats |= RTAUDIO_SINT32;
2912 else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB )
2913 info.nativeFormats |= RTAUDIO_FLOAT32;
2914 else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB )
2915 info.nativeFormats |= RTAUDIO_FLOAT64;
2916 else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB )
2917 info.nativeFormats |= RTAUDIO_SINT24;
2919 if ( info.outputChannels > 0 )
2920 if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true;
2921 if ( info.inputChannels > 0 )
2922 if ( getDefaultInputDevice() == device ) info.isDefaultInput = true;
2925 drivers.removeCurrentDriver();
2929 static void bufferSwitch( long index, ASIOBool /*processNow*/ )
2931 RtApiAsio *object = (RtApiAsio *) asioCallbackInfo->object;
2932 object->callbackEvent( index );
2935 void RtApiAsio :: saveDeviceInfo( void )
2939 unsigned int nDevices = getDeviceCount();
2940 devices_.resize( nDevices );
2941 for ( unsigned int i=0; i<nDevices; i++ )
2942 devices_[i] = getDeviceInfo( i );
2945 bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
2946 unsigned int firstChannel, unsigned int sampleRate,
2947 RtAudioFormat format, unsigned int *bufferSize,
2948 RtAudio::StreamOptions *options )
2949 {////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
2951 bool isDuplexInput = mode == INPUT && stream_.mode == OUTPUT;
2953 // For ASIO, a duplex stream MUST use the same driver.
2954 if ( isDuplexInput && stream_.device[0] != device ) {
2955 errorText_ = "RtApiAsio::probeDeviceOpen: an ASIO duplex stream must use the same device for input and output!";
2959 char driverName[32];
2960 ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 );
2961 if ( result != ASE_OK ) {
2962 errorStream_ << "RtApiAsio::probeDeviceOpen: unable to get driver name (" << getAsioErrorString( result ) << ").";
2963 errorText_ = errorStream_.str();
2967 // Only load the driver once for duplex stream.
2968 if ( !isDuplexInput ) {
2969 // The getDeviceInfo() function will not work when a stream is open
2970 // because ASIO does not allow multiple devices to run at the same
2971 // time. Thus, we'll probe the system before opening a stream and
2972 // save the results for use by getDeviceInfo().
2973 this->saveDeviceInfo();
2975 if ( !drivers.loadDriver( driverName ) ) {
2976 errorStream_ << "RtApiAsio::probeDeviceOpen: unable to load driver (" << driverName << ").";
2977 errorText_ = errorStream_.str();
2981 result = ASIOInit( &driverInfo );
2982 if ( result != ASE_OK ) {
2983 errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
2984 errorText_ = errorStream_.str();
2989 // keep them before any "goto error", they are used for error cleanup + goto device boundary checks
2990 bool buffersAllocated = false;
2991 AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
2992 unsigned int nChannels;
2995 // Check the device channel count.
2996 long inputChannels, outputChannels;
2997 result = ASIOGetChannels( &inputChannels, &outputChannels );
2998 if ( result != ASE_OK ) {
2999 errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
3000 errorText_ = errorStream_.str();
3004 if ( ( mode == OUTPUT && (channels+firstChannel) > (unsigned int) outputChannels) ||
3005 ( mode == INPUT && (channels+firstChannel) > (unsigned int) inputChannels) ) {
3006 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested channel count (" << channels << ") + offset (" << firstChannel << ").";
3007 errorText_ = errorStream_.str();
3010 stream_.nDeviceChannels[mode] = channels;
3011 stream_.nUserChannels[mode] = channels;
3012 stream_.channelOffset[mode] = firstChannel;
3014 // Verify the sample rate is supported.
3015 result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate );
3016 if ( result != ASE_OK ) {
3017 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested sample rate (" << sampleRate << ").";
3018 errorText_ = errorStream_.str();
3022 // Get the current sample rate
3023 ASIOSampleRate currentRate;
3024 result = ASIOGetSampleRate( ¤tRate );
3025 if ( result != ASE_OK ) {
3026 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error getting sample rate.";
3027 errorText_ = errorStream_.str();
3031 // Set the sample rate only if necessary
3032 if ( currentRate != sampleRate ) {
3033 result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate );
3034 if ( result != ASE_OK ) {
3035 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ").";
3036 errorText_ = errorStream_.str();
3041 // Determine the driver data type.
3042 ASIOChannelInfo channelInfo;
3043 channelInfo.channel = 0;
3044 if ( mode == OUTPUT ) channelInfo.isInput = false;
3045 else channelInfo.isInput = true;
3046 result = ASIOGetChannelInfo( &channelInfo );
3047 if ( result != ASE_OK ) {
3048 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting data format.";
3049 errorText_ = errorStream_.str();
3053 // Assuming WINDOWS host is always little-endian.
3054 stream_.doByteSwap[mode] = false;
3055 stream_.userFormat = format;
3056 stream_.deviceFormat[mode] = 0;
3057 if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) {
3058 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
3059 if ( channelInfo.type == ASIOSTInt16MSB ) stream_.doByteSwap[mode] = true;
3061 else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) {
3062 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
3063 if ( channelInfo.type == ASIOSTInt32MSB ) stream_.doByteSwap[mode] = true;
3065 else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) {
3066 stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
3067 if ( channelInfo.type == ASIOSTFloat32MSB ) stream_.doByteSwap[mode] = true;
3069 else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) {
3070 stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
3071 if ( channelInfo.type == ASIOSTFloat64MSB ) stream_.doByteSwap[mode] = true;
3073 else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB ) {
3074 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
3075 if ( channelInfo.type == ASIOSTInt24MSB ) stream_.doByteSwap[mode] = true;
3078 if ( stream_.deviceFormat[mode] == 0 ) {
3079 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") data format not supported by RtAudio.";
3080 errorText_ = errorStream_.str();
3084 // Set the buffer size. For a duplex stream, this will end up
3085 // setting the buffer size based on the input constraints, which
3087 long minSize, maxSize, preferSize, granularity;
3088 result = ASIOGetBufferSize( &minSize, &maxSize, &preferSize, &granularity );
3089 if ( result != ASE_OK ) {
3090 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting buffer size.";
3091 errorText_ = errorStream_.str();
3095 if ( isDuplexInput ) {
3096 // When this is the duplex input (output was opened before), then we have to use the same
3097 // buffersize as the output, because it might use the preferred buffer size, which most
3098 // likely wasn't passed as input to this. The buffer sizes have to be identically anyway,
3099 // So instead of throwing an error, make them equal. The caller uses the reference
3100 // to the "bufferSize" param as usual to set up processing buffers.
3102 *bufferSize = stream_.bufferSize;
3105 if ( *bufferSize == 0 ) *bufferSize = preferSize;
3106 else if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;
3107 else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;
3108 else if ( granularity == -1 ) {
3109 // Make sure bufferSize is a power of two.
3110 int log2_of_min_size = 0;
3111 int log2_of_max_size = 0;
3113 for ( unsigned int i = 0; i < sizeof(long) * 8; i++ ) {
3114 if ( minSize & ((long)1 << i) ) log2_of_min_size = i;
3115 if ( maxSize & ((long)1 << i) ) log2_of_max_size = i;
3118 long min_delta = std::abs( (long)*bufferSize - ((long)1 << log2_of_min_size) );
3119 int min_delta_num = log2_of_min_size;
3121 for (int i = log2_of_min_size + 1; i <= log2_of_max_size; i++) {
3122 long current_delta = std::abs( (long)*bufferSize - ((long)1 << i) );
3123 if (current_delta < min_delta) {
3124 min_delta = current_delta;
3129 *bufferSize = ( (unsigned int)1 << min_delta_num );
3130 if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;
3131 else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;
3133 else if ( granularity != 0 ) {
3134 // Set to an even multiple of granularity, rounding up.
3135 *bufferSize = (*bufferSize + granularity-1) / granularity * granularity;
3140 // we don't use it anymore, see above!
3141 // Just left it here for the case...
3142 if ( isDuplexInput && stream_.bufferSize != *bufferSize ) {
3143 errorText_ = "RtApiAsio::probeDeviceOpen: input/output buffersize discrepancy!";
3148 stream_.bufferSize = *bufferSize;
3149 stream_.nBuffers = 2;
3151 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
3152 else stream_.userInterleaved = true;
3154 // ASIO always uses non-interleaved buffers.
3155 stream_.deviceInterleaved[mode] = false;
3157 // Allocate, if necessary, our AsioHandle structure for the stream.
3158 if ( handle == 0 ) {
3160 handle = new AsioHandle;
3162 catch ( std::bad_alloc& ) {
3163 errorText_ = "RtApiAsio::probeDeviceOpen: error allocating AsioHandle memory.";
3166 handle->bufferInfos = 0;
3168 // Create a manual-reset event.
3169 handle->condition = CreateEvent( NULL, // no security
3170 TRUE, // manual-reset
3171 FALSE, // non-signaled initially
3173 stream_.apiHandle = (void *) handle;
3176 // Create the ASIO internal buffers. Since RtAudio sets up input
3177 // and output separately, we'll have to dispose of previously
3178 // created output buffers for a duplex stream.
3179 if ( mode == INPUT && stream_.mode == OUTPUT ) {
3180 ASIODisposeBuffers();
3181 if ( handle->bufferInfos ) free( handle->bufferInfos );
3184 // Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure.
3186 nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
3187 handle->bufferInfos = (ASIOBufferInfo *) malloc( nChannels * sizeof(ASIOBufferInfo) );
3188 if ( handle->bufferInfos == NULL ) {
3189 errorStream_ << "RtApiAsio::probeDeviceOpen: error allocating bufferInfo memory for driver (" << driverName << ").";
3190 errorText_ = errorStream_.str();
3194 ASIOBufferInfo *infos;
3195 infos = handle->bufferInfos;
3196 for ( i=0; i<stream_.nDeviceChannels[0]; i++, infos++ ) {
3197 infos->isInput = ASIOFalse;
3198 infos->channelNum = i + stream_.channelOffset[0];
3199 infos->buffers[0] = infos->buffers[1] = 0;
3201 for ( i=0; i<stream_.nDeviceChannels[1]; i++, infos++ ) {
3202 infos->isInput = ASIOTrue;
3203 infos->channelNum = i + stream_.channelOffset[1];
3204 infos->buffers[0] = infos->buffers[1] = 0;
3207 // prepare for callbacks
3208 stream_.sampleRate = sampleRate;
3209 stream_.device[mode] = device;
3210 stream_.mode = isDuplexInput ? DUPLEX : mode;
3212 // store this class instance before registering callbacks, that are going to use it
3213 asioCallbackInfo = &stream_.callbackInfo;
3214 stream_.callbackInfo.object = (void *) this;
3216 // Set up the ASIO callback structure and create the ASIO data buffers.
3217 asioCallbacks.bufferSwitch = &bufferSwitch;
3218 asioCallbacks.sampleRateDidChange = &sampleRateChanged;
3219 asioCallbacks.asioMessage = &asioMessages;
3220 asioCallbacks.bufferSwitchTimeInfo = NULL;
3221 result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks );
3222 if ( result != ASE_OK ) {
3223 // Standard method failed. This can happen with strict/misbehaving drivers that return valid buffer size ranges
3224 // but only accept the preferred buffer size as parameter for ASIOCreateBuffers (e.g. Creative's ASIO driver).
3225 // In that case, let's be naïve and try that instead.
3226 *bufferSize = preferSize;
3227 stream_.bufferSize = *bufferSize;
3228 result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks );
3231 if ( result != ASE_OK ) {
3232 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") creating buffers.";
3233 errorText_ = errorStream_.str();
3236 buffersAllocated = true;
3237 stream_.state = STREAM_STOPPED;
3239 // Set flags for buffer conversion.
3240 stream_.doConvertBuffer[mode] = false;
3241 if ( stream_.userFormat != stream_.deviceFormat[mode] )
3242 stream_.doConvertBuffer[mode] = true;
3243 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
3244 stream_.nUserChannels[mode] > 1 )
3245 stream_.doConvertBuffer[mode] = true;
3247 // Allocate necessary internal buffers
3248 unsigned long bufferBytes;
3249 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
3250 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
3251 if ( stream_.userBuffer[mode] == NULL ) {
3252 errorText_ = "RtApiAsio::probeDeviceOpen: error allocating user buffer memory.";
3256 if ( stream_.doConvertBuffer[mode] ) {
3258 bool makeBuffer = true;
3259 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
3260 if ( isDuplexInput && stream_.deviceBuffer ) {
3261 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
3262 if ( bufferBytes <= bytesOut ) makeBuffer = false;
3266 bufferBytes *= *bufferSize;
3267 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
3268 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
3269 if ( stream_.deviceBuffer == NULL ) {
3270 errorText_ = "RtApiAsio::probeDeviceOpen: error allocating device buffer memory.";
3276 // Determine device latencies
3277 long inputLatency, outputLatency;
3278 result = ASIOGetLatencies( &inputLatency, &outputLatency );
3279 if ( result != ASE_OK ) {
3280 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting latency.";
3281 errorText_ = errorStream_.str();
3282 error( RtAudioError::WARNING); // warn but don't fail
3285 stream_.latency[0] = outputLatency;
3286 stream_.latency[1] = inputLatency;
3289 // Setup the buffer conversion information structure. We don't use
3290 // buffers to do channel offsets, so we override that parameter
3292 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );
3297 if ( !isDuplexInput ) {
3298 // the cleanup for error in the duplex input, is done by RtApi::openStream
3299 // So we clean up for single channel only
3301 if ( buffersAllocated )
3302 ASIODisposeBuffers();
3304 drivers.removeCurrentDriver();
3307 CloseHandle( handle->condition );
3308 if ( handle->bufferInfos )
3309 free( handle->bufferInfos );
3312 stream_.apiHandle = 0;
3316 if ( stream_.userBuffer[mode] ) {
3317 free( stream_.userBuffer[mode] );
3318 stream_.userBuffer[mode] = 0;
3321 if ( stream_.deviceBuffer ) {
3322 free( stream_.deviceBuffer );
3323 stream_.deviceBuffer = 0;
3328 }////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
3330 void RtApiAsio :: closeStream()
3332 if ( stream_.state == STREAM_CLOSED ) {
3333 errorText_ = "RtApiAsio::closeStream(): no open stream to close!";
3334 error( RtAudioError::WARNING );
3338 if ( stream_.state == STREAM_RUNNING ) {
3339 stream_.state = STREAM_STOPPED;
3342 ASIODisposeBuffers();
3343 drivers.removeCurrentDriver();
3345 AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
3347 CloseHandle( handle->condition );
3348 if ( handle->bufferInfos )
3349 free( handle->bufferInfos );
3351 stream_.apiHandle = 0;
3354 for ( int i=0; i<2; i++ ) {
3355 if ( stream_.userBuffer[i] ) {
3356 free( stream_.userBuffer[i] );
3357 stream_.userBuffer[i] = 0;
3361 if ( stream_.deviceBuffer ) {
3362 free( stream_.deviceBuffer );
3363 stream_.deviceBuffer = 0;
3366 stream_.mode = UNINITIALIZED;
3367 stream_.state = STREAM_CLOSED;
3370 bool stopThreadCalled = false;
3372 void RtApiAsio :: startStream()
3375 if ( stream_.state == STREAM_RUNNING ) {
3376 errorText_ = "RtApiAsio::startStream(): the stream is already running!";
3377 error( RtAudioError::WARNING );
3381 AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
3382 ASIOError result = ASIOStart();
3383 if ( result != ASE_OK ) {
3384 errorStream_ << "RtApiAsio::startStream: error (" << getAsioErrorString( result ) << ") starting device.";
3385 errorText_ = errorStream_.str();
3389 handle->drainCounter = 0;
3390 handle->internalDrain = false;
3391 ResetEvent( handle->condition );
3392 stream_.state = STREAM_RUNNING;
3396 stopThreadCalled = false;
3398 if ( result == ASE_OK ) return;
3399 error( RtAudioError::SYSTEM_ERROR );
3402 void RtApiAsio :: stopStream()
3405 if ( stream_.state == STREAM_STOPPED ) {
3406 errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!";
3407 error( RtAudioError::WARNING );
3411 AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
3412 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
3413 if ( handle->drainCounter == 0 ) {
3414 handle->drainCounter = 2;
3415 WaitForSingleObject( handle->condition, INFINITE ); // block until signaled
3419 stream_.state = STREAM_STOPPED;
3421 ASIOError result = ASIOStop();
3422 if ( result != ASE_OK ) {
3423 errorStream_ << "RtApiAsio::stopStream: error (" << getAsioErrorString( result ) << ") stopping device.";
3424 errorText_ = errorStream_.str();
3427 if ( result == ASE_OK ) return;
3428 error( RtAudioError::SYSTEM_ERROR );
3431 void RtApiAsio :: abortStream()
3434 if ( stream_.state == STREAM_STOPPED ) {
3435 errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!";
3436 error( RtAudioError::WARNING );
3440 // The following lines were commented-out because some behavior was
3441 // noted where the device buffers need to be zeroed to avoid
3442 // continuing sound, even when the device buffers are completely
3443 // disposed. So now, calling abort is the same as calling stop.
3444 // AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
3445 // handle->drainCounter = 2;
3449 // This function will be called by a spawned thread when the user
3450 // callback function signals that the stream should be stopped or
3451 // aborted. It is necessary to handle it this way because the
3452 // callbackEvent() function must return before the ASIOStop()
3453 // function will return.
3454 static unsigned __stdcall asioStopStream( void *ptr )
3456 CallbackInfo *info = (CallbackInfo *) ptr;
3457 RtApiAsio *object = (RtApiAsio *) info->object;
3459 object->stopStream();
3464 bool RtApiAsio :: callbackEvent( long bufferIndex )
3466 if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;
3467 if ( stream_.state == STREAM_CLOSED ) {
3468 errorText_ = "RtApiAsio::callbackEvent(): the stream is closed ... this shouldn't happen!";
3469 error( RtAudioError::WARNING );
3473 CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
3474 AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
3476 // Check if we were draining the stream and signal if finished.
3477 if ( handle->drainCounter > 3 ) {
3479 stream_.state = STREAM_STOPPING;
3480 if ( handle->internalDrain == false )
3481 SetEvent( handle->condition );
3482 else { // spawn a thread to stop the stream
3484 stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream,
3485 &stream_.callbackInfo, 0, &threadId );
3490 // Invoke user callback to get fresh output data UNLESS we are
3492 if ( handle->drainCounter == 0 ) {
3493 RtAudioCallback callback = (RtAudioCallback) info->callback;
3494 double streamTime = getStreamTime();
3495 RtAudioStreamStatus status = 0;
3496 if ( stream_.mode != INPUT && asioXRun == true ) {
3497 status |= RTAUDIO_OUTPUT_UNDERFLOW;
3500 if ( stream_.mode != OUTPUT && asioXRun == true ) {
3501 status |= RTAUDIO_INPUT_OVERFLOW;
3504 int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
3505 stream_.bufferSize, streamTime, status, info->userData );
3506 if ( cbReturnValue == 2 ) {
3507 stream_.state = STREAM_STOPPING;
3508 handle->drainCounter = 2;
3510 stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream,
3511 &stream_.callbackInfo, 0, &threadId );
3514 else if ( cbReturnValue == 1 ) {
3515 handle->drainCounter = 1;
3516 handle->internalDrain = true;
3520 unsigned int nChannels, bufferBytes, i, j;
3521 nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
3522 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
3524 bufferBytes = stream_.bufferSize * formatBytes( stream_.deviceFormat[0] );
3526 if ( handle->drainCounter > 1 ) { // write zeros to the output stream
3528 for ( i=0, j=0; i<nChannels; i++ ) {
3529 if ( handle->bufferInfos[i].isInput != ASIOTrue )
3530 memset( handle->bufferInfos[i].buffers[bufferIndex], 0, bufferBytes );
3534 else if ( stream_.doConvertBuffer[0] ) {
3536 convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
3537 if ( stream_.doByteSwap[0] )
3538 byteSwapBuffer( stream_.deviceBuffer,
3539 stream_.bufferSize * stream_.nDeviceChannels[0],
3540 stream_.deviceFormat[0] );
3542 for ( i=0, j=0; i<nChannels; i++ ) {
3543 if ( handle->bufferInfos[i].isInput != ASIOTrue )
3544 memcpy( handle->bufferInfos[i].buffers[bufferIndex],
3545 &stream_.deviceBuffer[j++*bufferBytes], bufferBytes );
3551 if ( stream_.doByteSwap[0] )
3552 byteSwapBuffer( stream_.userBuffer[0],
3553 stream_.bufferSize * stream_.nUserChannels[0],
3554 stream_.userFormat );
3556 for ( i=0, j=0; i<nChannels; i++ ) {
3557 if ( handle->bufferInfos[i].isInput != ASIOTrue )
3558 memcpy( handle->bufferInfos[i].buffers[bufferIndex],
3559 &stream_.userBuffer[0][bufferBytes*j++], bufferBytes );
3565 // Don't bother draining input
3566 if ( handle->drainCounter ) {
3567 handle->drainCounter++;
3571 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
3573 bufferBytes = stream_.bufferSize * formatBytes(stream_.deviceFormat[1]);
3575 if (stream_.doConvertBuffer[1]) {
3577 // Always interleave ASIO input data.
3578 for ( i=0, j=0; i<nChannels; i++ ) {
3579 if ( handle->bufferInfos[i].isInput == ASIOTrue )
3580 memcpy( &stream_.deviceBuffer[j++*bufferBytes],
3581 handle->bufferInfos[i].buffers[bufferIndex],
3585 if ( stream_.doByteSwap[1] )
3586 byteSwapBuffer( stream_.deviceBuffer,
3587 stream_.bufferSize * stream_.nDeviceChannels[1],
3588 stream_.deviceFormat[1] );
3589 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
3593 for ( i=0, j=0; i<nChannels; i++ ) {
3594 if ( handle->bufferInfos[i].isInput == ASIOTrue ) {
3595 memcpy( &stream_.userBuffer[1][bufferBytes*j++],
3596 handle->bufferInfos[i].buffers[bufferIndex],
3601 if ( stream_.doByteSwap[1] )
3602 byteSwapBuffer( stream_.userBuffer[1],
3603 stream_.bufferSize * stream_.nUserChannels[1],
3604 stream_.userFormat );
3609 // The following call was suggested by Malte Clasen. While the API
3610 // documentation indicates it should not be required, some device
3611 // drivers apparently do not function correctly without it.
3614 RtApi::tickStreamTime();
3618 static void sampleRateChanged( ASIOSampleRate sRate )
3620 // The ASIO documentation says that this usually only happens during
3621 // external sync. Audio processing is not stopped by the driver,
3622 // actual sample rate might not have even changed, maybe only the
3623 // sample rate status of an AES/EBU or S/PDIF digital input at the
3626 RtApi *object = (RtApi *) asioCallbackInfo->object;
3628 object->stopStream();
3630 catch ( RtAudioError &exception ) {
3631 std::cerr << "\nRtApiAsio: sampleRateChanged() error (" << exception.getMessage() << ")!\n" << std::endl;
3635 std::cerr << "\nRtApiAsio: driver reports sample rate changed to " << sRate << " ... stream stopped!!!\n" << std::endl;
3638 static long asioMessages( long selector, long value, void* /*message*/, double* /*opt*/ )
3642 switch( selector ) {
3643 case kAsioSelectorSupported:
3644 if ( value == kAsioResetRequest
3645 || value == kAsioEngineVersion
3646 || value == kAsioResyncRequest
3647 || value == kAsioLatenciesChanged
3648 // The following three were added for ASIO 2.0, you don't
3649 // necessarily have to support them.
3650 || value == kAsioSupportsTimeInfo
3651 || value == kAsioSupportsTimeCode
3652 || value == kAsioSupportsInputMonitor)
3655 case kAsioResetRequest:
3656 // Defer the task and perform the reset of the driver during the
3657 // next "safe" situation. You cannot reset the driver right now,
3658 // as this code is called from the driver. Reset the driver is
3659 // done by completely destruct is. I.e. ASIOStop(),
3660 // ASIODisposeBuffers(), Destruction Afterwards you initialize the
3662 std::cerr << "\nRtApiAsio: driver reset requested!!!" << std::endl;
3665 case kAsioResyncRequest:
3666 // This informs the application that the driver encountered some
3667 // non-fatal data loss. It is used for synchronization purposes
3668 // of different media. Added mainly to work around the Win16Mutex
3669 // problems in Windows 95/98 with the Windows Multimedia system,
3670 // which could lose data because the Mutex was held too long by
3671 // another thread. However a driver can issue it in other
3673 // std::cerr << "\nRtApiAsio: driver resync requested!!!" << std::endl;
3677 case kAsioLatenciesChanged:
3678 // This will inform the host application that the drivers were
3679 // latencies changed. Beware, it this does not mean that the
3680 // buffer sizes have changed! You might need to update internal
3682 std::cerr << "\nRtApiAsio: driver latency may have changed!!!" << std::endl;
3685 case kAsioEngineVersion:
3686 // Return the supported ASIO version of the host application. If
3687 // a host application does not implement this selector, ASIO 1.0
3688 // is assumed by the driver.
3691 case kAsioSupportsTimeInfo:
3692 // Informs the driver whether the
3693 // asioCallbacks.bufferSwitchTimeInfo() callback is supported.
3694 // For compatibility with ASIO 1.0 drivers the host application
3695 // should always support the "old" bufferSwitch method, too.
3698 case kAsioSupportsTimeCode:
3699 // Informs the driver whether application is interested in time
3700 // code info. If an application does not need to know about time
3701 // code, the driver has less work to do.
3708 static const char* getAsioErrorString( ASIOError result )
3716 static const Messages m[] =
3718 { ASE_NotPresent, "Hardware input or output is not present or available." },
3719 { ASE_HWMalfunction, "Hardware is malfunctioning." },
3720 { ASE_InvalidParameter, "Invalid input parameter." },
3721 { ASE_InvalidMode, "Invalid mode." },
3722 { ASE_SPNotAdvancing, "Sample position not advancing." },
3723 { ASE_NoClock, "Sample clock or rate cannot be determined or is not present." },
3724 { ASE_NoMemory, "Not enough memory to complete the request." }
3727 for ( unsigned int i = 0; i < sizeof(m)/sizeof(m[0]); ++i )
3728 if ( m[i].value == result ) return m[i].message;
3730 return "Unknown error.";
3733 //******************** End of __WINDOWS_ASIO__ *********************//
3737 #if defined(__WINDOWS_WASAPI__) // Windows WASAPI API
3739 // Authored by Marcus Tomlinson <themarcustomlinson@gmail.com>, April 2014
3740 // - Introduces support for the Windows WASAPI API
3741 // - Aims to deliver bit streams to and from hardware at the lowest possible latency, via the absolute minimum buffer sizes required
3742 // - Provides flexible stream configuration to an otherwise strict and inflexible WASAPI interface
3743 // - Includes automatic internal conversion of sample rate and buffer size between hardware and the user
3750 #include <mferror.h>
3752 #include <mftransform.h>
3753 #include <wmcodecdsp.h>
3755 #include <audioclient.h>
3757 #include <mmdeviceapi.h>
3758 #include <functiondiscoverykeys_devpkey.h>
3760 #ifndef MF_E_TRANSFORM_NEED_MORE_INPUT
3761 #define MF_E_TRANSFORM_NEED_MORE_INPUT _HRESULT_TYPEDEF_(0xc00d6d72)
3764 #ifndef MFSTARTUP_NOSOCKET
3765 #define MFSTARTUP_NOSOCKET 0x1
3769 #pragma comment( lib, "ksuser" )
3770 #pragma comment( lib, "mfplat.lib" )
3771 #pragma comment( lib, "mfuuid.lib" )
3772 #pragma comment( lib, "wmcodecdspuuid" )
3775 //=============================================================================
3777 #define SAFE_RELEASE( objectPtr )\
3780 objectPtr->Release();\
3784 typedef HANDLE ( __stdcall *TAvSetMmThreadCharacteristicsPtr )( LPCWSTR TaskName, LPDWORD TaskIndex );
3786 //-----------------------------------------------------------------------------
3788 // WASAPI dictates stream sample rate, format, channel count, and in some cases, buffer size.
3789 // Therefore we must perform all necessary conversions to user buffers in order to satisfy these
3790 // requirements. WasapiBuffer ring buffers are used between HwIn->UserIn and UserOut->HwOut to
3791 // provide intermediate storage for read / write synchronization.
3805 // sets the length of the internal ring buffer
3806 void setBufferSize( unsigned int bufferSize, unsigned int formatBytes ) {
3809 buffer_ = ( char* ) calloc( bufferSize, formatBytes );
3811 bufferSize_ = bufferSize;
3816 // attempt to push a buffer into the ring buffer at the current "in" index
3817 bool pushBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format )
3819 if ( !buffer || // incoming buffer is NULL
3820 bufferSize == 0 || // incoming buffer has no data
3821 bufferSize > bufferSize_ ) // incoming buffer too large
3826 unsigned int relOutIndex = outIndex_;
3827 unsigned int inIndexEnd = inIndex_ + bufferSize;
3828 if ( relOutIndex < inIndex_ && inIndexEnd >= bufferSize_ ) {
3829 relOutIndex += bufferSize_;
3832 // "in" index can end on the "out" index but cannot begin at it
3833 if ( inIndex_ <= relOutIndex && inIndexEnd > relOutIndex ) {
3834 return false; // not enough space between "in" index and "out" index
3837 // copy buffer from external to internal
3838 int fromZeroSize = inIndex_ + bufferSize - bufferSize_;
3839 fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize;
3840 int fromInSize = bufferSize - fromZeroSize;
3845 memcpy( &( ( char* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( char ) );
3846 memcpy( buffer_, &( ( char* ) buffer )[fromInSize], fromZeroSize * sizeof( char ) );
3848 case RTAUDIO_SINT16:
3849 memcpy( &( ( short* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( short ) );
3850 memcpy( buffer_, &( ( short* ) buffer )[fromInSize], fromZeroSize * sizeof( short ) );
3852 case RTAUDIO_SINT24:
3853 memcpy( &( ( S24* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( S24 ) );
3854 memcpy( buffer_, &( ( S24* ) buffer )[fromInSize], fromZeroSize * sizeof( S24 ) );
3856 case RTAUDIO_SINT32:
3857 memcpy( &( ( int* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( int ) );
3858 memcpy( buffer_, &( ( int* ) buffer )[fromInSize], fromZeroSize * sizeof( int ) );
3860 case RTAUDIO_FLOAT32:
3861 memcpy( &( ( float* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( float ) );
3862 memcpy( buffer_, &( ( float* ) buffer )[fromInSize], fromZeroSize * sizeof( float ) );
3864 case RTAUDIO_FLOAT64:
3865 memcpy( &( ( double* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( double ) );
3866 memcpy( buffer_, &( ( double* ) buffer )[fromInSize], fromZeroSize * sizeof( double ) );
3870 // update "in" index
3871 inIndex_ += bufferSize;
3872 inIndex_ %= bufferSize_;
3877 // attempt to pull a buffer from the ring buffer from the current "out" index
3878 bool pullBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format )
3880 if ( !buffer || // incoming buffer is NULL
3881 bufferSize == 0 || // incoming buffer has no data
3882 bufferSize > bufferSize_ ) // incoming buffer too large
3887 unsigned int relInIndex = inIndex_;
3888 unsigned int outIndexEnd = outIndex_ + bufferSize;
3889 if ( relInIndex < outIndex_ && outIndexEnd >= bufferSize_ ) {
3890 relInIndex += bufferSize_;
3893 // "out" index can begin at and end on the "in" index
3894 if ( outIndex_ < relInIndex && outIndexEnd > relInIndex ) {
3895 return false; // not enough space between "out" index and "in" index
3898 // copy buffer from internal to external
3899 int fromZeroSize = outIndex_ + bufferSize - bufferSize_;
3900 fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize;
3901 int fromOutSize = bufferSize - fromZeroSize;
3906 memcpy( buffer, &( ( char* ) buffer_ )[outIndex_], fromOutSize * sizeof( char ) );
3907 memcpy( &( ( char* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( char ) );
3909 case RTAUDIO_SINT16:
3910 memcpy( buffer, &( ( short* ) buffer_ )[outIndex_], fromOutSize * sizeof( short ) );
3911 memcpy( &( ( short* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( short ) );
3913 case RTAUDIO_SINT24:
3914 memcpy( buffer, &( ( S24* ) buffer_ )[outIndex_], fromOutSize * sizeof( S24 ) );
3915 memcpy( &( ( S24* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( S24 ) );
3917 case RTAUDIO_SINT32:
3918 memcpy( buffer, &( ( int* ) buffer_ )[outIndex_], fromOutSize * sizeof( int ) );
3919 memcpy( &( ( int* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( int ) );
3921 case RTAUDIO_FLOAT32:
3922 memcpy( buffer, &( ( float* ) buffer_ )[outIndex_], fromOutSize * sizeof( float ) );
3923 memcpy( &( ( float* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( float ) );
3925 case RTAUDIO_FLOAT64:
3926 memcpy( buffer, &( ( double* ) buffer_ )[outIndex_], fromOutSize * sizeof( double ) );
3927 memcpy( &( ( double* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( double ) );
3931 // update "out" index
3932 outIndex_ += bufferSize;
3933 outIndex_ %= bufferSize_;
3940 unsigned int bufferSize_;
3941 unsigned int inIndex_;
3942 unsigned int outIndex_;
3945 //-----------------------------------------------------------------------------
3947 // In order to satisfy WASAPI's buffer requirements, we need a means of converting sample rate
3948 // between HW and the user. The WasapiResampler class is used to perform this conversion between
3949 // HwIn->UserIn and UserOut->HwOut during the stream callback loop.
3950 class WasapiResampler
3953 WasapiResampler( bool isFloat, unsigned int bitsPerSample, unsigned int channelCount,
3954 unsigned int inSampleRate, unsigned int outSampleRate )
3955 : _bytesPerSample( bitsPerSample / 8 )
3956 , _channelCount( channelCount )
3957 , _sampleRatio( ( float ) outSampleRate / inSampleRate )
3958 , _transformUnk( NULL )
3959 , _transform( NULL )
3960 , _mediaType( NULL )
3961 , _inputMediaType( NULL )
3962 , _outputMediaType( NULL )
3964 #ifdef __IWMResamplerProps_FWD_DEFINED__
3965 , _resamplerProps( NULL )
3968 // 1. Initialization
3970 MFStartup( MF_VERSION, MFSTARTUP_NOSOCKET );
3972 // 2. Create Resampler Transform Object
3974 CoCreateInstance( CLSID_CResamplerMediaObject, NULL, CLSCTX_INPROC_SERVER,
3975 IID_IUnknown, ( void** ) &_transformUnk );
3977 _transformUnk->QueryInterface( IID_PPV_ARGS( &_transform ) );
3979 #ifdef __IWMResamplerProps_FWD_DEFINED__
3980 _transformUnk->QueryInterface( IID_PPV_ARGS( &_resamplerProps ) );
3981 _resamplerProps->SetHalfFilterLength( 60 ); // best conversion quality
3984 // 3. Specify input / output format
3986 MFCreateMediaType( &_mediaType );
3987 _mediaType->SetGUID( MF_MT_MAJOR_TYPE, MFMediaType_Audio );
3988 _mediaType->SetGUID( MF_MT_SUBTYPE, isFloat ? MFAudioFormat_Float : MFAudioFormat_PCM );
3989 _mediaType->SetUINT32( MF_MT_AUDIO_NUM_CHANNELS, channelCount );
3990 _mediaType->SetUINT32( MF_MT_AUDIO_SAMPLES_PER_SECOND, inSampleRate );
3991 _mediaType->SetUINT32( MF_MT_AUDIO_BLOCK_ALIGNMENT, _bytesPerSample * channelCount );
3992 _mediaType->SetUINT32( MF_MT_AUDIO_AVG_BYTES_PER_SECOND, _bytesPerSample * channelCount * inSampleRate );
3993 _mediaType->SetUINT32( MF_MT_AUDIO_BITS_PER_SAMPLE, bitsPerSample );
3994 _mediaType->SetUINT32( MF_MT_ALL_SAMPLES_INDEPENDENT, TRUE );
3996 MFCreateMediaType( &_inputMediaType );
3997 _mediaType->CopyAllItems( _inputMediaType );
3999 _transform->SetInputType( 0, _inputMediaType, 0 );
4001 MFCreateMediaType( &_outputMediaType );
4002 _mediaType->CopyAllItems( _outputMediaType );
4004 _outputMediaType->SetUINT32( MF_MT_AUDIO_SAMPLES_PER_SECOND, outSampleRate );
4005 _outputMediaType->SetUINT32( MF_MT_AUDIO_AVG_BYTES_PER_SECOND, _bytesPerSample * channelCount * outSampleRate );
4007 _transform->SetOutputType( 0, _outputMediaType, 0 );
4009 // 4. Send stream start messages to Resampler
4011 _transform->ProcessMessage( MFT_MESSAGE_COMMAND_FLUSH, 0 );
4012 _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_BEGIN_STREAMING, 0 );
4013 _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_START_OF_STREAM, 0 );
4018 // 8. Send stream stop messages to Resampler
4020 _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_END_OF_STREAM, 0 );
4021 _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_END_STREAMING, 0 );
4027 SAFE_RELEASE( _transformUnk );
4028 SAFE_RELEASE( _transform );
4029 SAFE_RELEASE( _mediaType );
4030 SAFE_RELEASE( _inputMediaType );
4031 SAFE_RELEASE( _outputMediaType );
4033 #ifdef __IWMResamplerProps_FWD_DEFINED__
4034 SAFE_RELEASE( _resamplerProps );
4038 void Convert( char* outBuffer, const char* inBuffer, unsigned int inSampleCount, unsigned int& outSampleCount )
4040 unsigned int inputBufferSize = _bytesPerSample * _channelCount * inSampleCount;
4041 if ( _sampleRatio == 1 )
4043 // no sample rate conversion required
4044 memcpy( outBuffer, inBuffer, inputBufferSize );
4045 outSampleCount = inSampleCount;
4049 unsigned int outputBufferSize = ( unsigned int ) ceilf( inputBufferSize * _sampleRatio ) + ( _bytesPerSample * _channelCount );
4051 IMFMediaBuffer* rInBuffer;
4052 IMFSample* rInSample;
4053 BYTE* rInByteBuffer = NULL;
4055 // 5. Create Sample object from input data
4057 MFCreateMemoryBuffer( inputBufferSize, &rInBuffer );
4059 rInBuffer->Lock( &rInByteBuffer, NULL, NULL );
4060 memcpy( rInByteBuffer, inBuffer, inputBufferSize );
4061 rInBuffer->Unlock();
4062 rInByteBuffer = NULL;
4064 rInBuffer->SetCurrentLength( inputBufferSize );
4066 MFCreateSample( &rInSample );
4067 rInSample->AddBuffer( rInBuffer );
4069 // 6. Pass input data to Resampler
4071 _transform->ProcessInput( 0, rInSample, 0 );
4073 SAFE_RELEASE( rInBuffer );
4074 SAFE_RELEASE( rInSample );
4076 // 7. Perform sample rate conversion
4078 IMFMediaBuffer* rOutBuffer = NULL;
4079 BYTE* rOutByteBuffer = NULL;
4081 MFT_OUTPUT_DATA_BUFFER rOutDataBuffer;
4083 DWORD rBytes = outputBufferSize; // maximum bytes accepted per ProcessOutput
4085 // 7.1 Create Sample object for output data
4087 memset( &rOutDataBuffer, 0, sizeof rOutDataBuffer );
4088 MFCreateSample( &( rOutDataBuffer.pSample ) );
4089 MFCreateMemoryBuffer( rBytes, &rOutBuffer );
4090 rOutDataBuffer.pSample->AddBuffer( rOutBuffer );
4091 rOutDataBuffer.dwStreamID = 0;
4092 rOutDataBuffer.dwStatus = 0;
4093 rOutDataBuffer.pEvents = NULL;
4095 // 7.2 Get output data from Resampler
4097 if ( _transform->ProcessOutput( 0, 1, &rOutDataBuffer, &rStatus ) == MF_E_TRANSFORM_NEED_MORE_INPUT )
4100 SAFE_RELEASE( rOutBuffer );
4101 SAFE_RELEASE( rOutDataBuffer.pSample );
4105 // 7.3 Write output data to outBuffer
4107 SAFE_RELEASE( rOutBuffer );
4108 rOutDataBuffer.pSample->ConvertToContiguousBuffer( &rOutBuffer );
4109 rOutBuffer->GetCurrentLength( &rBytes );
4111 rOutBuffer->Lock( &rOutByteBuffer, NULL, NULL );
4112 memcpy( outBuffer, rOutByteBuffer, rBytes );
4113 rOutBuffer->Unlock();
4114 rOutByteBuffer = NULL;
4116 outSampleCount = rBytes / _bytesPerSample / _channelCount;
4117 SAFE_RELEASE( rOutBuffer );
4118 SAFE_RELEASE( rOutDataBuffer.pSample );
4122 unsigned int _bytesPerSample;
4123 unsigned int _channelCount;
4126 IUnknown* _transformUnk;
4127 IMFTransform* _transform;
4128 IMFMediaType* _mediaType;
4129 IMFMediaType* _inputMediaType;
4130 IMFMediaType* _outputMediaType;
4132 #ifdef __IWMResamplerProps_FWD_DEFINED__
4133 IWMResamplerProps* _resamplerProps;
4137 //-----------------------------------------------------------------------------
4139 // A structure to hold various information related to the WASAPI implementation.
4142 IAudioClient* captureAudioClient;
4143 IAudioClient* renderAudioClient;
4144 IAudioCaptureClient* captureClient;
4145 IAudioRenderClient* renderClient;
4146 HANDLE captureEvent;
4150 : captureAudioClient( NULL ),
4151 renderAudioClient( NULL ),
4152 captureClient( NULL ),
4153 renderClient( NULL ),
4154 captureEvent( NULL ),
4155 renderEvent( NULL ) {}
4158 //=============================================================================
4160 RtApiWasapi::RtApiWasapi()
4161 : coInitialized_( false ), deviceEnumerator_( NULL )
4163 // WASAPI can run either apartment or multi-threaded
4164 HRESULT hr = CoInitialize( NULL );
4165 if ( !FAILED( hr ) )
4166 coInitialized_ = true;
4168 // Instantiate device enumerator
4169 hr = CoCreateInstance( __uuidof( MMDeviceEnumerator ), NULL,
4170 CLSCTX_ALL, __uuidof( IMMDeviceEnumerator ),
4171 ( void** ) &deviceEnumerator_ );
4173 // If this runs on an old Windows, it will fail. Ignore and proceed.
4175 deviceEnumerator_ = NULL;
4178 //-----------------------------------------------------------------------------
4180 RtApiWasapi::~RtApiWasapi()
4182 if ( stream_.state != STREAM_CLOSED )
4185 SAFE_RELEASE( deviceEnumerator_ );
4187 // If this object previously called CoInitialize()
4188 if ( coInitialized_ )
4192 //=============================================================================
4194 unsigned int RtApiWasapi::getDeviceCount( void )
4196 unsigned int captureDeviceCount = 0;
4197 unsigned int renderDeviceCount = 0;
4199 IMMDeviceCollection* captureDevices = NULL;
4200 IMMDeviceCollection* renderDevices = NULL;
4202 if ( !deviceEnumerator_ )
4205 // Count capture devices
4207 HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
4208 if ( FAILED( hr ) ) {
4209 errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device collection.";
4213 hr = captureDevices->GetCount( &captureDeviceCount );
4214 if ( FAILED( hr ) ) {
4215 errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device count.";
4219 // Count render devices
4220 hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
4221 if ( FAILED( hr ) ) {
4222 errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device collection.";
4226 hr = renderDevices->GetCount( &renderDeviceCount );
4227 if ( FAILED( hr ) ) {
4228 errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device count.";
4233 // release all references
4234 SAFE_RELEASE( captureDevices );
4235 SAFE_RELEASE( renderDevices );
4237 if ( errorText_.empty() )
4238 return captureDeviceCount + renderDeviceCount;
4240 error( RtAudioError::DRIVER_ERROR );
4244 //-----------------------------------------------------------------------------
4246 RtAudio::DeviceInfo RtApiWasapi::getDeviceInfo( unsigned int device )
4248 RtAudio::DeviceInfo info;
4249 unsigned int captureDeviceCount = 0;
4250 unsigned int renderDeviceCount = 0;
4251 std::string defaultDeviceName;
4252 bool isCaptureDevice = false;
4254 PROPVARIANT deviceNameProp;
4255 PROPVARIANT defaultDeviceNameProp;
4257 IMMDeviceCollection* captureDevices = NULL;
4258 IMMDeviceCollection* renderDevices = NULL;
4259 IMMDevice* devicePtr = NULL;
4260 IMMDevice* defaultDevicePtr = NULL;
4261 IAudioClient* audioClient = NULL;
4262 IPropertyStore* devicePropStore = NULL;
4263 IPropertyStore* defaultDevicePropStore = NULL;
4265 WAVEFORMATEX* deviceFormat = NULL;
4266 WAVEFORMATEX* closestMatchFormat = NULL;
4269 info.probed = false;
4271 // Count capture devices
4273 RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
4274 HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
4275 if ( FAILED( hr ) ) {
4276 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device collection.";
4280 hr = captureDevices->GetCount( &captureDeviceCount );
4281 if ( FAILED( hr ) ) {
4282 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device count.";
4286 // Count render devices
4287 hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
4288 if ( FAILED( hr ) ) {
4289 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device collection.";
4293 hr = renderDevices->GetCount( &renderDeviceCount );
4294 if ( FAILED( hr ) ) {
4295 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device count.";
4299 // validate device index
4300 if ( device >= captureDeviceCount + renderDeviceCount ) {
4301 errorText_ = "RtApiWasapi::getDeviceInfo: Invalid device index.";
4302 errorType = RtAudioError::INVALID_USE;
4306 // determine whether index falls within capture or render devices
4307 if ( device >= renderDeviceCount ) {
4308 hr = captureDevices->Item( device - renderDeviceCount, &devicePtr );
4309 if ( FAILED( hr ) ) {
4310 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device handle.";
4313 isCaptureDevice = true;
4316 hr = renderDevices->Item( device, &devicePtr );
4317 if ( FAILED( hr ) ) {
4318 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device handle.";
4321 isCaptureDevice = false;
4324 // get default device name
4325 if ( isCaptureDevice ) {
4326 hr = deviceEnumerator_->GetDefaultAudioEndpoint( eCapture, eConsole, &defaultDevicePtr );
4327 if ( FAILED( hr ) ) {
4328 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default capture device handle.";
4333 hr = deviceEnumerator_->GetDefaultAudioEndpoint( eRender, eConsole, &defaultDevicePtr );
4334 if ( FAILED( hr ) ) {
4335 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default render device handle.";
4340 hr = defaultDevicePtr->OpenPropertyStore( STGM_READ, &defaultDevicePropStore );
4341 if ( FAILED( hr ) ) {
4342 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open default device property store.";
4345 PropVariantInit( &defaultDeviceNameProp );
4347 hr = defaultDevicePropStore->GetValue( PKEY_Device_FriendlyName, &defaultDeviceNameProp );
4348 if ( FAILED( hr ) ) {
4349 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default device property: PKEY_Device_FriendlyName.";
4353 defaultDeviceName = convertCharPointerToStdString(defaultDeviceNameProp.pwszVal);
4356 hr = devicePtr->OpenPropertyStore( STGM_READ, &devicePropStore );
4357 if ( FAILED( hr ) ) {
4358 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open device property store.";
4362 PropVariantInit( &deviceNameProp );
4364 hr = devicePropStore->GetValue( PKEY_Device_FriendlyName, &deviceNameProp );
4365 if ( FAILED( hr ) ) {
4366 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device property: PKEY_Device_FriendlyName.";
4370 info.name =convertCharPointerToStdString(deviceNameProp.pwszVal);
4373 if ( isCaptureDevice ) {
4374 info.isDefaultInput = info.name == defaultDeviceName;
4375 info.isDefaultOutput = false;
4378 info.isDefaultInput = false;
4379 info.isDefaultOutput = info.name == defaultDeviceName;
4383 hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, NULL, ( void** ) &audioClient );
4384 if ( FAILED( hr ) ) {
4385 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device audio client.";
4389 hr = audioClient->GetMixFormat( &deviceFormat );
4390 if ( FAILED( hr ) ) {
4391 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device mix format.";
4395 if ( isCaptureDevice ) {
4396 info.inputChannels = deviceFormat->nChannels;
4397 info.outputChannels = 0;
4398 info.duplexChannels = 0;
4401 info.inputChannels = 0;
4402 info.outputChannels = deviceFormat->nChannels;
4403 info.duplexChannels = 0;
4407 info.sampleRates.clear();
4409 // allow support for all sample rates as we have a built-in sample rate converter
4410 for ( unsigned int i = 0; i < MAX_SAMPLE_RATES; i++ ) {
4411 info.sampleRates.push_back( SAMPLE_RATES[i] );
4413 info.preferredSampleRate = deviceFormat->nSamplesPerSec;
4416 info.nativeFormats = 0;
4418 if ( deviceFormat->wFormatTag == WAVE_FORMAT_IEEE_FLOAT ||
4419 ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
4420 ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_IEEE_FLOAT ) )
4422 if ( deviceFormat->wBitsPerSample == 32 ) {
4423 info.nativeFormats |= RTAUDIO_FLOAT32;
4425 else if ( deviceFormat->wBitsPerSample == 64 ) {
4426 info.nativeFormats |= RTAUDIO_FLOAT64;
4429 else if ( deviceFormat->wFormatTag == WAVE_FORMAT_PCM ||
4430 ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
4431 ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_PCM ) )
4433 if ( deviceFormat->wBitsPerSample == 8 ) {
4434 info.nativeFormats |= RTAUDIO_SINT8;
4436 else if ( deviceFormat->wBitsPerSample == 16 ) {
4437 info.nativeFormats |= RTAUDIO_SINT16;
4439 else if ( deviceFormat->wBitsPerSample == 24 ) {
4440 info.nativeFormats |= RTAUDIO_SINT24;
4442 else if ( deviceFormat->wBitsPerSample == 32 ) {
4443 info.nativeFormats |= RTAUDIO_SINT32;
4451 // release all references
4452 PropVariantClear( &deviceNameProp );
4453 PropVariantClear( &defaultDeviceNameProp );
4455 SAFE_RELEASE( captureDevices );
4456 SAFE_RELEASE( renderDevices );
4457 SAFE_RELEASE( devicePtr );
4458 SAFE_RELEASE( defaultDevicePtr );
4459 SAFE_RELEASE( audioClient );
4460 SAFE_RELEASE( devicePropStore );
4461 SAFE_RELEASE( defaultDevicePropStore );
4463 CoTaskMemFree( deviceFormat );
4464 CoTaskMemFree( closestMatchFormat );
4466 if ( !errorText_.empty() )
4471 //-----------------------------------------------------------------------------
4473 unsigned int RtApiWasapi::getDefaultOutputDevice( void )
4475 for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {
4476 if ( getDeviceInfo( i ).isDefaultOutput ) {
4484 //-----------------------------------------------------------------------------
4486 unsigned int RtApiWasapi::getDefaultInputDevice( void )
4488 for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {
4489 if ( getDeviceInfo( i ).isDefaultInput ) {
4497 //-----------------------------------------------------------------------------
4499 void RtApiWasapi::closeStream( void )
4501 if ( stream_.state == STREAM_CLOSED ) {
4502 errorText_ = "RtApiWasapi::closeStream: No open stream to close.";
4503 error( RtAudioError::WARNING );
4507 if ( stream_.state != STREAM_STOPPED )
4510 // clean up stream memory
4511 SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient )
4512 SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient )
4514 SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureClient )
4515 SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderClient )
4517 if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent )
4518 CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent );
4520 if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent )
4521 CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent );
4523 delete ( WasapiHandle* ) stream_.apiHandle;
4524 stream_.apiHandle = NULL;
4526 for ( int i = 0; i < 2; i++ ) {
4527 if ( stream_.userBuffer[i] ) {
4528 free( stream_.userBuffer[i] );
4529 stream_.userBuffer[i] = 0;
4533 if ( stream_.deviceBuffer ) {
4534 free( stream_.deviceBuffer );
4535 stream_.deviceBuffer = 0;
4538 // update stream state
4539 stream_.state = STREAM_CLOSED;
4542 //-----------------------------------------------------------------------------
4544 void RtApiWasapi::startStream( void )
4548 if ( stream_.state == STREAM_RUNNING ) {
4549 errorText_ = "RtApiWasapi::startStream: The stream is already running.";
4550 error( RtAudioError::WARNING );
4554 // update stream state
4555 stream_.state = STREAM_RUNNING;
4557 // create WASAPI stream thread
4558 stream_.callbackInfo.thread = ( ThreadHandle ) CreateThread( NULL, 0, runWasapiThread, this, CREATE_SUSPENDED, NULL );
4560 if ( !stream_.callbackInfo.thread ) {
4561 errorText_ = "RtApiWasapi::startStream: Unable to instantiate callback thread.";
4562 error( RtAudioError::THREAD_ERROR );
4565 SetThreadPriority( ( void* ) stream_.callbackInfo.thread, stream_.callbackInfo.priority );
4566 ResumeThread( ( void* ) stream_.callbackInfo.thread );
4570 //-----------------------------------------------------------------------------
4572 void RtApiWasapi::stopStream( void )
4576 if ( stream_.state == STREAM_STOPPED ) {
4577 errorText_ = "RtApiWasapi::stopStream: The stream is already stopped.";
4578 error( RtAudioError::WARNING );
4582 // inform stream thread by setting stream state to STREAM_STOPPING
4583 stream_.state = STREAM_STOPPING;
4585 // wait until stream thread is stopped
4586 while( stream_.state != STREAM_STOPPED ) {
4590 // Wait for the last buffer to play before stopping.
4591 Sleep( 1000 * stream_.bufferSize / stream_.sampleRate );
4593 // stop capture client if applicable
4594 if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) {
4595 HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop();
4596 if ( FAILED( hr ) ) {
4597 errorText_ = "RtApiWasapi::stopStream: Unable to stop capture stream.";
4598 error( RtAudioError::DRIVER_ERROR );
4603 // stop render client if applicable
4604 if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) {
4605 HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop();
4606 if ( FAILED( hr ) ) {
4607 errorText_ = "RtApiWasapi::stopStream: Unable to stop render stream.";
4608 error( RtAudioError::DRIVER_ERROR );
4613 // close thread handle
4614 if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
4615 errorText_ = "RtApiWasapi::stopStream: Unable to close callback thread.";
4616 error( RtAudioError::THREAD_ERROR );
4620 stream_.callbackInfo.thread = (ThreadHandle) NULL;
4623 //-----------------------------------------------------------------------------
4625 void RtApiWasapi::abortStream( void )
4629 if ( stream_.state == STREAM_STOPPED ) {
4630 errorText_ = "RtApiWasapi::abortStream: The stream is already stopped.";
4631 error( RtAudioError::WARNING );
4635 // inform stream thread by setting stream state to STREAM_STOPPING
4636 stream_.state = STREAM_STOPPING;
4638 // wait until stream thread is stopped
4639 while ( stream_.state != STREAM_STOPPED ) {
4643 // stop capture client if applicable
4644 if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) {
4645 HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop();
4646 if ( FAILED( hr ) ) {
4647 errorText_ = "RtApiWasapi::abortStream: Unable to stop capture stream.";
4648 error( RtAudioError::DRIVER_ERROR );
4653 // stop render client if applicable
4654 if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) {
4655 HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop();
4656 if ( FAILED( hr ) ) {
4657 errorText_ = "RtApiWasapi::abortStream: Unable to stop render stream.";
4658 error( RtAudioError::DRIVER_ERROR );
4663 // close thread handle
4664 if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
4665 errorText_ = "RtApiWasapi::abortStream: Unable to close callback thread.";
4666 error( RtAudioError::THREAD_ERROR );
4670 stream_.callbackInfo.thread = (ThreadHandle) NULL;
4673 //-----------------------------------------------------------------------------
4675 bool RtApiWasapi::probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
4676 unsigned int firstChannel, unsigned int sampleRate,
4677 RtAudioFormat format, unsigned int* bufferSize,
4678 RtAudio::StreamOptions* options )
4680 bool methodResult = FAILURE;
4681 unsigned int captureDeviceCount = 0;
4682 unsigned int renderDeviceCount = 0;
4684 IMMDeviceCollection* captureDevices = NULL;
4685 IMMDeviceCollection* renderDevices = NULL;
4686 IMMDevice* devicePtr = NULL;
4687 WAVEFORMATEX* deviceFormat = NULL;
4688 unsigned int bufferBytes;
4689 stream_.state = STREAM_STOPPED;
4691 // create API Handle if not already created
4692 if ( !stream_.apiHandle )
4693 stream_.apiHandle = ( void* ) new WasapiHandle();
4695 // Count capture devices
4697 RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
4698 HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
4699 if ( FAILED( hr ) ) {
4700 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device collection.";
4704 hr = captureDevices->GetCount( &captureDeviceCount );
4705 if ( FAILED( hr ) ) {
4706 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device count.";
4710 // Count render devices
4711 hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
4712 if ( FAILED( hr ) ) {
4713 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device collection.";
4717 hr = renderDevices->GetCount( &renderDeviceCount );
4718 if ( FAILED( hr ) ) {
4719 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device count.";
4723 // validate device index
4724 if ( device >= captureDeviceCount + renderDeviceCount ) {
4725 errorType = RtAudioError::INVALID_USE;
4726 errorText_ = "RtApiWasapi::probeDeviceOpen: Invalid device index.";
4730 // if device index falls within capture devices
4731 if ( device >= renderDeviceCount ) {
4732 if ( mode != INPUT ) {
4733 errorType = RtAudioError::INVALID_USE;
4734 errorText_ = "RtApiWasapi::probeDeviceOpen: Capture device selected as output device.";
4738 // retrieve captureAudioClient from devicePtr
4739 IAudioClient*& captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
4741 hr = captureDevices->Item( device - renderDeviceCount, &devicePtr );
4742 if ( FAILED( hr ) ) {
4743 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device handle.";
4747 hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
4748 NULL, ( void** ) &captureAudioClient );
4749 if ( FAILED( hr ) ) {
4750 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device audio client.";
4754 hr = captureAudioClient->GetMixFormat( &deviceFormat );
4755 if ( FAILED( hr ) ) {
4756 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device mix format.";
4760 stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
4761 captureAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
4764 // if device index falls within render devices and is configured for loopback
4765 if ( device < renderDeviceCount && mode == INPUT )
4767 // if renderAudioClient is not initialised, initialise it now
4768 IAudioClient*& renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
4769 if ( !renderAudioClient )
4771 probeDeviceOpen( device, OUTPUT, channels, firstChannel, sampleRate, format, bufferSize, options );
4774 // retrieve captureAudioClient from devicePtr
4775 IAudioClient*& captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
4777 hr = renderDevices->Item( device, &devicePtr );
4778 if ( FAILED( hr ) ) {
4779 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device handle.";
4783 hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
4784 NULL, ( void** ) &captureAudioClient );
4785 if ( FAILED( hr ) ) {
4786 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device audio client.";
4790 hr = captureAudioClient->GetMixFormat( &deviceFormat );
4791 if ( FAILED( hr ) ) {
4792 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device mix format.";
4796 stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
4797 captureAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
4800 // if device index falls within render devices and is configured for output
4801 if ( device < renderDeviceCount && mode == OUTPUT )
4803 // if renderAudioClient is already initialised, don't initialise it again
4804 IAudioClient*& renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
4805 if ( renderAudioClient )
4807 methodResult = SUCCESS;
4811 hr = renderDevices->Item( device, &devicePtr );
4812 if ( FAILED( hr ) ) {
4813 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device handle.";
4817 hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
4818 NULL, ( void** ) &renderAudioClient );
4819 if ( FAILED( hr ) ) {
4820 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device audio client.";
4824 hr = renderAudioClient->GetMixFormat( &deviceFormat );
4825 if ( FAILED( hr ) ) {
4826 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device mix format.";
4830 stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
4831 renderAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
4835 if ( ( stream_.mode == OUTPUT && mode == INPUT ) ||
4836 ( stream_.mode == INPUT && mode == OUTPUT ) ) {
4837 stream_.mode = DUPLEX;
4840 stream_.mode = mode;
4843 stream_.device[mode] = device;
4844 stream_.doByteSwap[mode] = false;
4845 stream_.sampleRate = sampleRate;
4846 stream_.bufferSize = *bufferSize;
4847 stream_.nBuffers = 1;
4848 stream_.nUserChannels[mode] = channels;
4849 stream_.channelOffset[mode] = firstChannel;
4850 stream_.userFormat = format;
4851 stream_.deviceFormat[mode] = getDeviceInfo( device ).nativeFormats;
4853 if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
4854 stream_.userInterleaved = false;
4856 stream_.userInterleaved = true;
4857 stream_.deviceInterleaved[mode] = true;
4859 // Set flags for buffer conversion.
4860 stream_.doConvertBuffer[mode] = false;
4861 if ( stream_.userFormat != stream_.deviceFormat[mode] ||
4862 stream_.nUserChannels[0] != stream_.nDeviceChannels[0] ||
4863 stream_.nUserChannels[1] != stream_.nDeviceChannels[1] ||
4864 stream_.userInterleaved )
4865 stream_.doConvertBuffer[mode] = true;
4866 else if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
4867 stream_.nUserChannels[mode] > 1 )
4868 stream_.doConvertBuffer[mode] = true;
4870 if ( stream_.doConvertBuffer[mode] )
4871 setConvertInfo( mode, 0 );
4873 // Allocate necessary internal buffers
4874 bufferBytes = stream_.nUserChannels[mode] * stream_.bufferSize * formatBytes( stream_.userFormat );
4876 stream_.userBuffer[mode] = ( char* ) calloc( bufferBytes, 1 );
4877 if ( !stream_.userBuffer[mode] ) {
4878 errorType = RtAudioError::MEMORY_ERROR;
4879 errorText_ = "RtApiWasapi::probeDeviceOpen: Error allocating user buffer memory.";
4883 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME )
4884 stream_.callbackInfo.priority = 15;
4886 stream_.callbackInfo.priority = 0;
4888 ///! TODO: RTAUDIO_MINIMIZE_LATENCY // Provide stream buffers directly to callback
4889 ///! TODO: RTAUDIO_HOG_DEVICE // Exclusive mode
4891 methodResult = SUCCESS;
4895 SAFE_RELEASE( captureDevices );
4896 SAFE_RELEASE( renderDevices );
4897 SAFE_RELEASE( devicePtr );
4898 CoTaskMemFree( deviceFormat );
4900 // if method failed, close the stream
4901 if ( methodResult == FAILURE )
4904 if ( !errorText_.empty() )
4906 return methodResult;
4909 //=============================================================================
4911 DWORD WINAPI RtApiWasapi::runWasapiThread( void* wasapiPtr )
4914 ( ( RtApiWasapi* ) wasapiPtr )->wasapiThread();
4919 DWORD WINAPI RtApiWasapi::stopWasapiThread( void* wasapiPtr )
4922 ( ( RtApiWasapi* ) wasapiPtr )->stopStream();
4927 DWORD WINAPI RtApiWasapi::abortWasapiThread( void* wasapiPtr )
4930 ( ( RtApiWasapi* ) wasapiPtr )->abortStream();
4935 //-----------------------------------------------------------------------------
4937 void RtApiWasapi::wasapiThread()
4939 // as this is a new thread, we must CoInitialize it
4940 CoInitialize( NULL );
4944 IAudioClient* captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
4945 IAudioClient* renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
4946 IAudioCaptureClient* captureClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureClient;
4947 IAudioRenderClient* renderClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderClient;
4948 HANDLE captureEvent = ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent;
4949 HANDLE renderEvent = ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent;
4951 WAVEFORMATEX* captureFormat = NULL;
4952 WAVEFORMATEX* renderFormat = NULL;
4953 float captureSrRatio = 0.0f;
4954 float renderSrRatio = 0.0f;
4955 WasapiBuffer captureBuffer;
4956 WasapiBuffer renderBuffer;
4957 WasapiResampler* captureResampler = NULL;
4958 WasapiResampler* renderResampler = NULL;
4960 // declare local stream variables
4961 RtAudioCallback callback = ( RtAudioCallback ) stream_.callbackInfo.callback;
4962 BYTE* streamBuffer = NULL;
4963 unsigned long captureFlags = 0;
4964 unsigned int bufferFrameCount = 0;
4965 unsigned int numFramesPadding = 0;
4966 unsigned int convBufferSize = 0;
4967 bool loopbackEnabled = stream_.device[INPUT] == stream_.device[OUTPUT];
4968 bool callbackPushed = true;
4969 bool callbackPulled = false;
4970 bool callbackStopped = false;
4971 int callbackResult = 0;
4973 // convBuffer is used to store converted buffers between WASAPI and the user
4974 char* convBuffer = NULL;
4975 unsigned int convBuffSize = 0;
4976 unsigned int deviceBuffSize = 0;
4978 std::string errorText;
4979 RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
4981 // Attempt to assign "Pro Audio" characteristic to thread
4982 HMODULE AvrtDll = LoadLibrary( (LPCTSTR) "AVRT.dll" );
4984 DWORD taskIndex = 0;
4985 TAvSetMmThreadCharacteristicsPtr AvSetMmThreadCharacteristicsPtr = ( TAvSetMmThreadCharacteristicsPtr ) GetProcAddress( AvrtDll, "AvSetMmThreadCharacteristicsW" );
4986 AvSetMmThreadCharacteristicsPtr( L"Pro Audio", &taskIndex );
4987 FreeLibrary( AvrtDll );
4990 // start capture stream if applicable
4991 if ( captureAudioClient ) {
4992 hr = captureAudioClient->GetMixFormat( &captureFormat );
4993 if ( FAILED( hr ) ) {
4994 errorText = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
4998 // init captureResampler
4999 captureResampler = new WasapiResampler( stream_.deviceFormat[INPUT] == RTAUDIO_FLOAT32 || stream_.deviceFormat[INPUT] == RTAUDIO_FLOAT64,
5000 formatBytes( stream_.deviceFormat[INPUT] ) * 8, stream_.nDeviceChannels[INPUT],
5001 captureFormat->nSamplesPerSec, stream_.sampleRate );
5003 captureSrRatio = ( ( float ) captureFormat->nSamplesPerSec / stream_.sampleRate );
5005 if ( !captureClient ) {
5006 hr = captureAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
5007 loopbackEnabled ? AUDCLNT_STREAMFLAGS_LOOPBACK : AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
5012 if ( FAILED( hr ) ) {
5013 errorText = "RtApiWasapi::wasapiThread: Unable to initialize capture audio client.";
5017 hr = captureAudioClient->GetService( __uuidof( IAudioCaptureClient ),
5018 ( void** ) &captureClient );
5019 if ( FAILED( hr ) ) {
5020 errorText = "RtApiWasapi::wasapiThread: Unable to retrieve capture client handle.";
5024 // don't configure captureEvent if in loopback mode
5025 if ( !loopbackEnabled )
5027 // configure captureEvent to trigger on every available capture buffer
5028 captureEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
5029 if ( !captureEvent ) {
5030 errorType = RtAudioError::SYSTEM_ERROR;
5031 errorText = "RtApiWasapi::wasapiThread: Unable to create capture event.";
5035 hr = captureAudioClient->SetEventHandle( captureEvent );
5036 if ( FAILED( hr ) ) {
5037 errorText = "RtApiWasapi::wasapiThread: Unable to set capture event handle.";
5041 ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent = captureEvent;
5044 ( ( WasapiHandle* ) stream_.apiHandle )->captureClient = captureClient;
5047 unsigned int inBufferSize = 0;
5048 hr = captureAudioClient->GetBufferSize( &inBufferSize );
5049 if ( FAILED( hr ) ) {
5050 errorText = "RtApiWasapi::wasapiThread: Unable to get capture buffer size.";
5054 // scale outBufferSize according to stream->user sample rate ratio
5055 unsigned int outBufferSize = ( unsigned int ) ceilf( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT];
5056 inBufferSize *= stream_.nDeviceChannels[INPUT];
5058 // set captureBuffer size
5059 captureBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[INPUT] ) );
5061 // reset the capture stream
5062 hr = captureAudioClient->Reset();
5063 if ( FAILED( hr ) ) {
5064 errorText = "RtApiWasapi::wasapiThread: Unable to reset capture stream.";
5068 // start the capture stream
5069 hr = captureAudioClient->Start();
5070 if ( FAILED( hr ) ) {
5071 errorText = "RtApiWasapi::wasapiThread: Unable to start capture stream.";
5076 // start render stream if applicable
5077 if ( renderAudioClient ) {
5078 hr = renderAudioClient->GetMixFormat( &renderFormat );
5079 if ( FAILED( hr ) ) {
5080 errorText = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
5084 // init renderResampler
5085 renderResampler = new WasapiResampler( stream_.deviceFormat[OUTPUT] == RTAUDIO_FLOAT32 || stream_.deviceFormat[OUTPUT] == RTAUDIO_FLOAT64,
5086 formatBytes( stream_.deviceFormat[OUTPUT] ) * 8, stream_.nDeviceChannels[OUTPUT],
5087 stream_.sampleRate, renderFormat->nSamplesPerSec );
5089 renderSrRatio = ( ( float ) renderFormat->nSamplesPerSec / stream_.sampleRate );
5091 if ( !renderClient ) {
5092 hr = renderAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
5093 AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
5098 if ( FAILED( hr ) ) {
5099 errorText = "RtApiWasapi::wasapiThread: Unable to initialize render audio client.";
5103 hr = renderAudioClient->GetService( __uuidof( IAudioRenderClient ),
5104 ( void** ) &renderClient );
5105 if ( FAILED( hr ) ) {
5106 errorText = "RtApiWasapi::wasapiThread: Unable to retrieve render client handle.";
5110 // configure renderEvent to trigger on every available render buffer
5111 renderEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
5112 if ( !renderEvent ) {
5113 errorType = RtAudioError::SYSTEM_ERROR;
5114 errorText = "RtApiWasapi::wasapiThread: Unable to create render event.";
5118 hr = renderAudioClient->SetEventHandle( renderEvent );
5119 if ( FAILED( hr ) ) {
5120 errorText = "RtApiWasapi::wasapiThread: Unable to set render event handle.";
5124 ( ( WasapiHandle* ) stream_.apiHandle )->renderClient = renderClient;
5125 ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent = renderEvent;
5128 unsigned int outBufferSize = 0;
5129 hr = renderAudioClient->GetBufferSize( &outBufferSize );
5130 if ( FAILED( hr ) ) {
5131 errorText = "RtApiWasapi::wasapiThread: Unable to get render buffer size.";
5135 // scale inBufferSize according to user->stream sample rate ratio
5136 unsigned int inBufferSize = ( unsigned int ) ceilf( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT];
5137 outBufferSize *= stream_.nDeviceChannels[OUTPUT];
5139 // set renderBuffer size
5140 renderBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[OUTPUT] ) );
5142 // reset the render stream
5143 hr = renderAudioClient->Reset();
5144 if ( FAILED( hr ) ) {
5145 errorText = "RtApiWasapi::wasapiThread: Unable to reset render stream.";
5149 // start the render stream
5150 hr = renderAudioClient->Start();
5151 if ( FAILED( hr ) ) {
5152 errorText = "RtApiWasapi::wasapiThread: Unable to start render stream.";
5157 // malloc buffer memory
5158 if ( stream_.mode == INPUT )
5160 using namespace std; // for ceilf
5161 convBuffSize = ( size_t ) ( ceilf( stream_.bufferSize * captureSrRatio ) ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
5162 deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
5164 else if ( stream_.mode == OUTPUT )
5166 convBuffSize = ( size_t ) ( ceilf( stream_.bufferSize * renderSrRatio ) ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
5167 deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
5169 else if ( stream_.mode == DUPLEX )
5171 convBuffSize = std::max( ( size_t ) ( ceilf( stream_.bufferSize * captureSrRatio ) ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
5172 ( size_t ) ( ceilf( stream_.bufferSize * renderSrRatio ) ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
5173 deviceBuffSize = std::max( stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
5174 stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
5177 convBuffSize *= 2; // allow overflow for *SrRatio remainders
5178 convBuffer = ( char* ) malloc( convBuffSize );
5179 stream_.deviceBuffer = ( char* ) malloc( deviceBuffSize );
5180 if ( !convBuffer || !stream_.deviceBuffer ) {
5181 errorType = RtAudioError::MEMORY_ERROR;
5182 errorText = "RtApiWasapi::wasapiThread: Error allocating device buffer memory.";
5186 // stream process loop
5187 while ( stream_.state != STREAM_STOPPING ) {
5188 if ( !callbackPulled ) {
5191 // 1. Pull callback buffer from inputBuffer
5192 // 2. If 1. was successful: Convert callback buffer to user sample rate and channel count
5193 // Convert callback buffer to user format
5195 if ( captureAudioClient )
5197 int samplesToPull = ( unsigned int ) floorf( stream_.bufferSize * captureSrRatio );
5198 if ( captureSrRatio != 1 )
5200 // account for remainders
5205 while ( convBufferSize < stream_.bufferSize )
5207 // Pull callback buffer from inputBuffer
5208 callbackPulled = captureBuffer.pullBuffer( convBuffer,
5209 samplesToPull * stream_.nDeviceChannels[INPUT],
5210 stream_.deviceFormat[INPUT] );
5212 if ( !callbackPulled )
5217 // Convert callback buffer to user sample rate
5218 unsigned int deviceBufferOffset = convBufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
5219 unsigned int convSamples = 0;
5221 captureResampler->Convert( stream_.deviceBuffer + deviceBufferOffset,
5226 convBufferSize += convSamples;
5227 samplesToPull = 1; // now pull one sample at a time until we have stream_.bufferSize samples
5230 if ( callbackPulled )
5232 if ( stream_.doConvertBuffer[INPUT] ) {
5233 // Convert callback buffer to user format
5234 convertBuffer( stream_.userBuffer[INPUT],
5235 stream_.deviceBuffer,
5236 stream_.convertInfo[INPUT] );
5239 // no further conversion, simple copy deviceBuffer to userBuffer
5240 memcpy( stream_.userBuffer[INPUT],
5241 stream_.deviceBuffer,
5242 stream_.bufferSize * stream_.nUserChannels[INPUT] * formatBytes( stream_.userFormat ) );
5247 // if there is no capture stream, set callbackPulled flag
5248 callbackPulled = true;
5253 // 1. Execute user callback method
5254 // 2. Handle return value from callback
5256 // if callback has not requested the stream to stop
5257 if ( callbackPulled && !callbackStopped ) {
5258 // Execute user callback method
5259 callbackResult = callback( stream_.userBuffer[OUTPUT],
5260 stream_.userBuffer[INPUT],
5263 captureFlags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY ? RTAUDIO_INPUT_OVERFLOW : 0,
5264 stream_.callbackInfo.userData );
5266 // Handle return value from callback
5267 if ( callbackResult == 1 ) {
5268 // instantiate a thread to stop this thread
5269 HANDLE threadHandle = CreateThread( NULL, 0, stopWasapiThread, this, 0, NULL );
5270 if ( !threadHandle ) {
5271 errorType = RtAudioError::THREAD_ERROR;
5272 errorText = "RtApiWasapi::wasapiThread: Unable to instantiate stream stop thread.";
5275 else if ( !CloseHandle( threadHandle ) ) {
5276 errorType = RtAudioError::THREAD_ERROR;
5277 errorText = "RtApiWasapi::wasapiThread: Unable to close stream stop thread handle.";
5281 callbackStopped = true;
5283 else if ( callbackResult == 2 ) {
5284 // instantiate a thread to stop this thread
5285 HANDLE threadHandle = CreateThread( NULL, 0, abortWasapiThread, this, 0, NULL );
5286 if ( !threadHandle ) {
5287 errorType = RtAudioError::THREAD_ERROR;
5288 errorText = "RtApiWasapi::wasapiThread: Unable to instantiate stream abort thread.";
5291 else if ( !CloseHandle( threadHandle ) ) {
5292 errorType = RtAudioError::THREAD_ERROR;
5293 errorText = "RtApiWasapi::wasapiThread: Unable to close stream abort thread handle.";
5297 callbackStopped = true;
5304 // 1. Convert callback buffer to stream format
5305 // 2. Convert callback buffer to stream sample rate and channel count
5306 // 3. Push callback buffer into outputBuffer
5308 if ( renderAudioClient && callbackPulled )
5310 // if the last call to renderBuffer.PushBuffer() was successful
5311 if ( callbackPushed || convBufferSize == 0 )
5313 if ( stream_.doConvertBuffer[OUTPUT] )
5315 // Convert callback buffer to stream format
5316 convertBuffer( stream_.deviceBuffer,
5317 stream_.userBuffer[OUTPUT],
5318 stream_.convertInfo[OUTPUT] );
5322 // Convert callback buffer to stream sample rate
5323 renderResampler->Convert( convBuffer,
5324 stream_.deviceBuffer,
5329 // Push callback buffer into outputBuffer
5330 callbackPushed = renderBuffer.pushBuffer( convBuffer,
5331 convBufferSize * stream_.nDeviceChannels[OUTPUT],
5332 stream_.deviceFormat[OUTPUT] );
5335 // if there is no render stream, set callbackPushed flag
5336 callbackPushed = true;
5341 // 1. Get capture buffer from stream
5342 // 2. Push capture buffer into inputBuffer
5343 // 3. If 2. was successful: Release capture buffer
5345 if ( captureAudioClient ) {
5346 // if the callback input buffer was not pulled from captureBuffer, wait for next capture event
5347 if ( !callbackPulled ) {
5348 WaitForSingleObject( loopbackEnabled ? renderEvent : captureEvent, INFINITE );
5351 // Get capture buffer from stream
5352 hr = captureClient->GetBuffer( &streamBuffer,
5354 &captureFlags, NULL, NULL );
5355 if ( FAILED( hr ) ) {
5356 errorText = "RtApiWasapi::wasapiThread: Unable to retrieve capture buffer.";
5360 if ( bufferFrameCount != 0 ) {
5361 // Push capture buffer into inputBuffer
5362 if ( captureBuffer.pushBuffer( ( char* ) streamBuffer,
5363 bufferFrameCount * stream_.nDeviceChannels[INPUT],
5364 stream_.deviceFormat[INPUT] ) )
5366 // Release capture buffer
5367 hr = captureClient->ReleaseBuffer( bufferFrameCount );
5368 if ( FAILED( hr ) ) {
5369 errorText = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
5375 // Inform WASAPI that capture was unsuccessful
5376 hr = captureClient->ReleaseBuffer( 0 );
5377 if ( FAILED( hr ) ) {
5378 errorText = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
5385 // Inform WASAPI that capture was unsuccessful
5386 hr = captureClient->ReleaseBuffer( 0 );
5387 if ( FAILED( hr ) ) {
5388 errorText = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
5396 // 1. Get render buffer from stream
5397 // 2. Pull next buffer from outputBuffer
5398 // 3. If 2. was successful: Fill render buffer with next buffer
5399 // Release render buffer
5401 if ( renderAudioClient ) {
5402 // if the callback output buffer was not pushed to renderBuffer, wait for next render event
5403 if ( callbackPulled && !callbackPushed ) {
5404 WaitForSingleObject( renderEvent, INFINITE );
5407 // Get render buffer from stream
5408 hr = renderAudioClient->GetBufferSize( &bufferFrameCount );
5409 if ( FAILED( hr ) ) {
5410 errorText = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer size.";
5414 hr = renderAudioClient->GetCurrentPadding( &numFramesPadding );
5415 if ( FAILED( hr ) ) {
5416 errorText = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer padding.";
5420 bufferFrameCount -= numFramesPadding;
5422 if ( bufferFrameCount != 0 ) {
5423 hr = renderClient->GetBuffer( bufferFrameCount, &streamBuffer );
5424 if ( FAILED( hr ) ) {
5425 errorText = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer.";
5429 // Pull next buffer from outputBuffer
5430 // Fill render buffer with next buffer
5431 if ( renderBuffer.pullBuffer( ( char* ) streamBuffer,
5432 bufferFrameCount * stream_.nDeviceChannels[OUTPUT],
5433 stream_.deviceFormat[OUTPUT] ) )
5435 // Release render buffer
5436 hr = renderClient->ReleaseBuffer( bufferFrameCount, 0 );
5437 if ( FAILED( hr ) ) {
5438 errorText = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
5444 // Inform WASAPI that render was unsuccessful
5445 hr = renderClient->ReleaseBuffer( 0, 0 );
5446 if ( FAILED( hr ) ) {
5447 errorText = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
5454 // Inform WASAPI that render was unsuccessful
5455 hr = renderClient->ReleaseBuffer( 0, 0 );
5456 if ( FAILED( hr ) ) {
5457 errorText = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
5463 // if the callback buffer was pushed renderBuffer reset callbackPulled flag
5464 if ( callbackPushed ) {
5465 // unsetting the callbackPulled flag lets the stream know that
5466 // the audio device is ready for another callback output buffer.
5467 callbackPulled = false;
5470 RtApi::tickStreamTime();
5477 CoTaskMemFree( captureFormat );
5478 CoTaskMemFree( renderFormat );
5480 free ( convBuffer );
5481 delete renderResampler;
5482 delete captureResampler;
5486 // update stream state
5487 stream_.state = STREAM_STOPPED;
5489 if ( !errorText.empty() )
5491 errorText_ = errorText;
5496 //******************** End of __WINDOWS_WASAPI__ *********************//
5500 #if defined(__WINDOWS_DS__) // Windows DirectSound API
5502 // Modified by Robin Davies, October 2005
5503 // - Improvements to DirectX pointer chasing.
5504 // - Bug fix for non-power-of-two Asio granularity used by Edirol PCR-A30.
5505 // - Auto-call CoInitialize for DSOUND and ASIO platforms.
5506 // Various revisions for RtAudio 4.0 by Gary Scavone, April 2007
5507 // Changed device query structure for RtAudio 4.0.7, January 2010
5509 #include <windows.h>
5510 #include <process.h>
5511 #include <mmsystem.h>
5515 #include <algorithm>
5517 #if defined(__MINGW32__)
5518 // missing from latest mingw winapi
5519 #define WAVE_FORMAT_96M08 0x00010000 /* 96 kHz, Mono, 8-bit */
5520 #define WAVE_FORMAT_96S08 0x00020000 /* 96 kHz, Stereo, 8-bit */
5521 #define WAVE_FORMAT_96M16 0x00040000 /* 96 kHz, Mono, 16-bit */
5522 #define WAVE_FORMAT_96S16 0x00080000 /* 96 kHz, Stereo, 16-bit */
5525 #define MINIMUM_DEVICE_BUFFER_SIZE 32768
5527 #ifdef _MSC_VER // if Microsoft Visual C++
5528 #pragma comment( lib, "winmm.lib" ) // then, auto-link winmm.lib. Otherwise, it has to be added manually.
5531 static inline DWORD dsPointerBetween( DWORD pointer, DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize )
5533 if ( pointer > bufferSize ) pointer -= bufferSize;
5534 if ( laterPointer < earlierPointer ) laterPointer += bufferSize;
5535 if ( pointer < earlierPointer ) pointer += bufferSize;
5536 return pointer >= earlierPointer && pointer < laterPointer;
5539 // A structure to hold various information related to the DirectSound
5540 // API implementation.
5542 unsigned int drainCounter; // Tracks callback counts when draining
5543 bool internalDrain; // Indicates if stop is initiated from callback or not.
5547 UINT bufferPointer[2];
5548 DWORD dsBufferSize[2];
5549 DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by.
5553 :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; }
5556 // Declarations for utility functions, callbacks, and structures
5557 // specific to the DirectSound implementation.
5558 static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
5559 LPCTSTR description,
5563 static const char* getErrorString( int code );
5565 static unsigned __stdcall callbackHandler( void *ptr );
5574 : found(false) { validId[0] = false; validId[1] = false; }
5577 struct DsProbeData {
5579 std::vector<struct DsDevice>* dsDevices;
5582 RtApiDs :: RtApiDs()
5584 // Dsound will run both-threaded. If CoInitialize fails, then just
5585 // accept whatever the mainline chose for a threading model.
5586 coInitialized_ = false;
5587 HRESULT hr = CoInitialize( NULL );
5588 if ( !FAILED( hr ) ) coInitialized_ = true;
5591 RtApiDs :: ~RtApiDs()
5593 if ( stream_.state != STREAM_CLOSED ) closeStream();
5594 if ( coInitialized_ ) CoUninitialize(); // balanced call.
5597 // The DirectSound default output is always the first device.
5598 unsigned int RtApiDs :: getDefaultOutputDevice( void )
5603 // The DirectSound default input is always the first input device,
5604 // which is the first capture device enumerated.
5605 unsigned int RtApiDs :: getDefaultInputDevice( void )
5610 unsigned int RtApiDs :: getDeviceCount( void )
5612 // Set query flag for previously found devices to false, so that we
5613 // can check for any devices that have disappeared.
5614 for ( unsigned int i=0; i<dsDevices.size(); i++ )
5615 dsDevices[i].found = false;
5617 // Query DirectSound devices.
5618 struct DsProbeData probeInfo;
5619 probeInfo.isInput = false;
5620 probeInfo.dsDevices = &dsDevices;
5621 HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );
5622 if ( FAILED( result ) ) {
5623 errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!";
5624 errorText_ = errorStream_.str();
5625 error( RtAudioError::WARNING );
5628 // Query DirectSoundCapture devices.
5629 probeInfo.isInput = true;
5630 result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );
5631 if ( FAILED( result ) ) {
5632 errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!";
5633 errorText_ = errorStream_.str();
5634 error( RtAudioError::WARNING );
5637 // Clean out any devices that may have disappeared (code update submitted by Eli Zehngut).
5638 for ( unsigned int i=0; i<dsDevices.size(); ) {
5639 if ( dsDevices[i].found == false ) dsDevices.erase( dsDevices.begin() + i );
5643 return static_cast<unsigned int>(dsDevices.size());
5646 RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device )
5648 RtAudio::DeviceInfo info;
5649 info.probed = false;
5651 if ( dsDevices.size() == 0 ) {
5652 // Force a query of all devices
5654 if ( dsDevices.size() == 0 ) {
5655 errorText_ = "RtApiDs::getDeviceInfo: no devices found!";
5656 error( RtAudioError::INVALID_USE );
5661 if ( device >= dsDevices.size() ) {
5662 errorText_ = "RtApiDs::getDeviceInfo: device ID is invalid!";
5663 error( RtAudioError::INVALID_USE );
5668 if ( dsDevices[ device ].validId[0] == false ) goto probeInput;
5670 LPDIRECTSOUND output;
5672 result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );
5673 if ( FAILED( result ) ) {
5674 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!";
5675 errorText_ = errorStream_.str();
5676 error( RtAudioError::WARNING );
5680 outCaps.dwSize = sizeof( outCaps );
5681 result = output->GetCaps( &outCaps );
5682 if ( FAILED( result ) ) {
5684 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting capabilities!";
5685 errorText_ = errorStream_.str();
5686 error( RtAudioError::WARNING );
5690 // Get output channel information.
5691 info.outputChannels = ( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1;
5693 // Get sample rate information.
5694 info.sampleRates.clear();
5695 for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
5696 if ( SAMPLE_RATES[k] >= (unsigned int) outCaps.dwMinSecondarySampleRate &&
5697 SAMPLE_RATES[k] <= (unsigned int) outCaps.dwMaxSecondarySampleRate ) {
5698 info.sampleRates.push_back( SAMPLE_RATES[k] );
5700 if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
5701 info.preferredSampleRate = SAMPLE_RATES[k];
5705 // Get format information.
5706 if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT ) info.nativeFormats |= RTAUDIO_SINT16;
5707 if ( outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) info.nativeFormats |= RTAUDIO_SINT8;
5711 if ( getDefaultOutputDevice() == device )
5712 info.isDefaultOutput = true;
5714 if ( dsDevices[ device ].validId[1] == false ) {
5715 info.name = dsDevices[ device ].name;
5722 LPDIRECTSOUNDCAPTURE input;
5723 result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );
5724 if ( FAILED( result ) ) {
5725 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";
5726 errorText_ = errorStream_.str();
5727 error( RtAudioError::WARNING );
5732 inCaps.dwSize = sizeof( inCaps );
5733 result = input->GetCaps( &inCaps );
5734 if ( FAILED( result ) ) {
5736 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsDevices[ device ].name << ")!";
5737 errorText_ = errorStream_.str();
5738 error( RtAudioError::WARNING );
5742 // Get input channel information.
5743 info.inputChannels = inCaps.dwChannels;
5745 // Get sample rate and format information.
5746 std::vector<unsigned int> rates;
5747 if ( inCaps.dwChannels >= 2 ) {
5748 if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.nativeFormats |= RTAUDIO_SINT16;
5749 if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.nativeFormats |= RTAUDIO_SINT16;
5750 if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.nativeFormats |= RTAUDIO_SINT16;
5751 if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.nativeFormats |= RTAUDIO_SINT16;
5752 if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.nativeFormats |= RTAUDIO_SINT8;
5753 if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.nativeFormats |= RTAUDIO_SINT8;
5754 if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.nativeFormats |= RTAUDIO_SINT8;
5755 if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.nativeFormats |= RTAUDIO_SINT8;
5757 if ( info.nativeFormats & RTAUDIO_SINT16 ) {
5758 if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) rates.push_back( 11025 );
5759 if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) rates.push_back( 22050 );
5760 if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) rates.push_back( 44100 );
5761 if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) rates.push_back( 96000 );
5763 else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
5764 if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) rates.push_back( 11025 );
5765 if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) rates.push_back( 22050 );
5766 if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) rates.push_back( 44100 );
5767 if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) rates.push_back( 96000 );
5770 else if ( inCaps.dwChannels == 1 ) {
5771 if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.nativeFormats |= RTAUDIO_SINT16;
5772 if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.nativeFormats |= RTAUDIO_SINT16;
5773 if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.nativeFormats |= RTAUDIO_SINT16;
5774 if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.nativeFormats |= RTAUDIO_SINT16;
5775 if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.nativeFormats |= RTAUDIO_SINT8;
5776 if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.nativeFormats |= RTAUDIO_SINT8;
5777 if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.nativeFormats |= RTAUDIO_SINT8;
5778 if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.nativeFormats |= RTAUDIO_SINT8;
5780 if ( info.nativeFormats & RTAUDIO_SINT16 ) {
5781 if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) rates.push_back( 11025 );
5782 if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) rates.push_back( 22050 );
5783 if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) rates.push_back( 44100 );
5784 if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) rates.push_back( 96000 );
5786 else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
5787 if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) rates.push_back( 11025 );
5788 if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) rates.push_back( 22050 );
5789 if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) rates.push_back( 44100 );
5790 if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) rates.push_back( 96000 );
5793 else info.inputChannels = 0; // technically, this would be an error
5797 if ( info.inputChannels == 0 ) return info;
5799 // Copy the supported rates to the info structure but avoid duplication.
5801 for ( unsigned int i=0; i<rates.size(); i++ ) {
5803 for ( unsigned int j=0; j<info.sampleRates.size(); j++ ) {
5804 if ( rates[i] == info.sampleRates[j] ) {
5809 if ( found == false ) info.sampleRates.push_back( rates[i] );
5811 std::sort( info.sampleRates.begin(), info.sampleRates.end() );
5813 // If device opens for both playback and capture, we determine the channels.
5814 if ( info.outputChannels > 0 && info.inputChannels > 0 )
5815 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
5817 if ( device == 0 ) info.isDefaultInput = true;
5819 // Copy name and return.
5820 info.name = dsDevices[ device ].name;
5825 bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
5826 unsigned int firstChannel, unsigned int sampleRate,
5827 RtAudioFormat format, unsigned int *bufferSize,
5828 RtAudio::StreamOptions *options )
5830 if ( channels + firstChannel > 2 ) {
5831 errorText_ = "RtApiDs::probeDeviceOpen: DirectSound does not support more than 2 channels per device.";
5835 size_t nDevices = dsDevices.size();
5836 if ( nDevices == 0 ) {
5837 // This should not happen because a check is made before this function is called.
5838 errorText_ = "RtApiDs::probeDeviceOpen: no devices found!";
5842 if ( device >= nDevices ) {
5843 // This should not happen because a check is made before this function is called.
5844 errorText_ = "RtApiDs::probeDeviceOpen: device ID is invalid!";
5848 if ( mode == OUTPUT ) {
5849 if ( dsDevices[ device ].validId[0] == false ) {
5850 errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support output!";
5851 errorText_ = errorStream_.str();
5855 else { // mode == INPUT
5856 if ( dsDevices[ device ].validId[1] == false ) {
5857 errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support input!";
5858 errorText_ = errorStream_.str();
5863 // According to a note in PortAudio, using GetDesktopWindow()
5864 // instead of GetForegroundWindow() is supposed to avoid problems
5865 // that occur when the application's window is not the foreground
5866 // window. Also, if the application window closes before the
5867 // DirectSound buffer, DirectSound can crash. In the past, I had
5868 // problems when using GetDesktopWindow() but it seems fine now
5869 // (January 2010). I'll leave it commented here.
5870 // HWND hWnd = GetForegroundWindow();
5871 HWND hWnd = GetDesktopWindow();
5873 // Check the numberOfBuffers parameter and limit the lowest value to
5874 // two. This is a judgement call and a value of two is probably too
5875 // low for capture, but it should work for playback.
5877 if ( options ) nBuffers = options->numberOfBuffers;
5878 if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) nBuffers = 2;
5879 if ( nBuffers < 2 ) nBuffers = 3;
5881 // Check the lower range of the user-specified buffer size and set
5882 // (arbitrarily) to a lower bound of 32.
5883 if ( *bufferSize < 32 ) *bufferSize = 32;
5885 // Create the wave format structure. The data format setting will
5886 // be determined later.
5887 WAVEFORMATEX waveFormat;
5888 ZeroMemory( &waveFormat, sizeof(WAVEFORMATEX) );
5889 waveFormat.wFormatTag = WAVE_FORMAT_PCM;
5890 waveFormat.nChannels = channels + firstChannel;
5891 waveFormat.nSamplesPerSec = (unsigned long) sampleRate;
5893 // Determine the device buffer size. By default, we'll use the value
5894 // defined above (32K), but we will grow it to make allowances for
5895 // very large software buffer sizes.
5896 DWORD dsBufferSize = MINIMUM_DEVICE_BUFFER_SIZE;
5897 DWORD dsPointerLeadTime = 0;
5899 void *ohandle = 0, *bhandle = 0;
5901 if ( mode == OUTPUT ) {
5903 LPDIRECTSOUND output;
5904 result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );
5905 if ( FAILED( result ) ) {
5906 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!";
5907 errorText_ = errorStream_.str();
5912 outCaps.dwSize = sizeof( outCaps );
5913 result = output->GetCaps( &outCaps );
5914 if ( FAILED( result ) ) {
5916 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting capabilities (" << dsDevices[ device ].name << ")!";
5917 errorText_ = errorStream_.str();
5921 // Check channel information.
5922 if ( channels + firstChannel == 2 && !( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ) {
5923 errorStream_ << "RtApiDs::getDeviceInfo: the output device (" << dsDevices[ device ].name << ") does not support stereo playback.";
5924 errorText_ = errorStream_.str();
5928 // Check format information. Use 16-bit format unless not
5929 // supported or user requests 8-bit.
5930 if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT &&
5931 !( format == RTAUDIO_SINT8 && outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) ) {
5932 waveFormat.wBitsPerSample = 16;
5933 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
5936 waveFormat.wBitsPerSample = 8;
5937 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
5939 stream_.userFormat = format;
5941 // Update wave format structure and buffer information.
5942 waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
5943 waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
5944 dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
5946 // If the user wants an even bigger buffer, increase the device buffer size accordingly.
5947 while ( dsPointerLeadTime * 2U > dsBufferSize )
5950 // Set cooperative level to DSSCL_EXCLUSIVE ... sound stops when window focus changes.
5951 // result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE );
5952 // Set cooperative level to DSSCL_PRIORITY ... sound remains when window focus changes.
5953 result = output->SetCooperativeLevel( hWnd, DSSCL_PRIORITY );
5954 if ( FAILED( result ) ) {
5956 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting cooperative level (" << dsDevices[ device ].name << ")!";
5957 errorText_ = errorStream_.str();
5961 // Even though we will write to the secondary buffer, we need to
5962 // access the primary buffer to set the correct output format
5963 // (since the default is 8-bit, 22 kHz!). Setup the DS primary
5964 // buffer description.
5965 DSBUFFERDESC bufferDescription;
5966 ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
5967 bufferDescription.dwSize = sizeof( DSBUFFERDESC );
5968 bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER;
5970 // Obtain the primary buffer
5971 LPDIRECTSOUNDBUFFER buffer;
5972 result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
5973 if ( FAILED( result ) ) {
5975 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") accessing primary buffer (" << dsDevices[ device ].name << ")!";
5976 errorText_ = errorStream_.str();
5980 // Set the primary DS buffer sound format.
5981 result = buffer->SetFormat( &waveFormat );
5982 if ( FAILED( result ) ) {
5984 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting primary buffer format (" << dsDevices[ device ].name << ")!";
5985 errorText_ = errorStream_.str();
5989 // Setup the secondary DS buffer description.
5990 ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
5991 bufferDescription.dwSize = sizeof( DSBUFFERDESC );
5992 bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
5993 DSBCAPS_GLOBALFOCUS |
5994 DSBCAPS_GETCURRENTPOSITION2 |
5995 DSBCAPS_LOCHARDWARE ); // Force hardware mixing
5996 bufferDescription.dwBufferBytes = dsBufferSize;
5997 bufferDescription.lpwfxFormat = &waveFormat;
5999 // Try to create the secondary DS buffer. If that doesn't work,
6000 // try to use software mixing. Otherwise, there's a problem.
6001 result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
6002 if ( FAILED( result ) ) {
6003 bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
6004 DSBCAPS_GLOBALFOCUS |
6005 DSBCAPS_GETCURRENTPOSITION2 |
6006 DSBCAPS_LOCSOFTWARE ); // Force software mixing
6007 result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
6008 if ( FAILED( result ) ) {
6010 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating secondary buffer (" << dsDevices[ device ].name << ")!";
6011 errorText_ = errorStream_.str();
6016 // Get the buffer size ... might be different from what we specified.
6018 dsbcaps.dwSize = sizeof( DSBCAPS );
6019 result = buffer->GetCaps( &dsbcaps );
6020 if ( FAILED( result ) ) {
6023 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";
6024 errorText_ = errorStream_.str();
6028 dsBufferSize = dsbcaps.dwBufferBytes;
6030 // Lock the DS buffer
6033 result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );
6034 if ( FAILED( result ) ) {
6037 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking buffer (" << dsDevices[ device ].name << ")!";
6038 errorText_ = errorStream_.str();
6042 // Zero the DS buffer
6043 ZeroMemory( audioPtr, dataLen );
6045 // Unlock the DS buffer
6046 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
6047 if ( FAILED( result ) ) {
6050 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking buffer (" << dsDevices[ device ].name << ")!";
6051 errorText_ = errorStream_.str();
6055 ohandle = (void *) output;
6056 bhandle = (void *) buffer;
6059 if ( mode == INPUT ) {
6061 LPDIRECTSOUNDCAPTURE input;
6062 result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );
6063 if ( FAILED( result ) ) {
6064 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";
6065 errorText_ = errorStream_.str();
6070 inCaps.dwSize = sizeof( inCaps );
6071 result = input->GetCaps( &inCaps );
6072 if ( FAILED( result ) ) {
6074 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsDevices[ device ].name << ")!";
6075 errorText_ = errorStream_.str();
6079 // Check channel information.
6080 if ( inCaps.dwChannels < channels + firstChannel ) {
6081 errorText_ = "RtApiDs::getDeviceInfo: the input device does not support requested input channels.";
6085 // Check format information. Use 16-bit format unless user
6087 DWORD deviceFormats;
6088 if ( channels + firstChannel == 2 ) {
6089 deviceFormats = WAVE_FORMAT_1S08 | WAVE_FORMAT_2S08 | WAVE_FORMAT_4S08 | WAVE_FORMAT_96S08;
6090 if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
6091 waveFormat.wBitsPerSample = 8;
6092 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
6094 else { // assume 16-bit is supported
6095 waveFormat.wBitsPerSample = 16;
6096 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
6099 else { // channel == 1
6100 deviceFormats = WAVE_FORMAT_1M08 | WAVE_FORMAT_2M08 | WAVE_FORMAT_4M08 | WAVE_FORMAT_96M08;
6101 if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
6102 waveFormat.wBitsPerSample = 8;
6103 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
6105 else { // assume 16-bit is supported
6106 waveFormat.wBitsPerSample = 16;
6107 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
6110 stream_.userFormat = format;
6112 // Update wave format structure and buffer information.
6113 waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
6114 waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
6115 dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
6117 // If the user wants an even bigger buffer, increase the device buffer size accordingly.
6118 while ( dsPointerLeadTime * 2U > dsBufferSize )
6121 // Setup the secondary DS buffer description.
6122 DSCBUFFERDESC bufferDescription;
6123 ZeroMemory( &bufferDescription, sizeof( DSCBUFFERDESC ) );
6124 bufferDescription.dwSize = sizeof( DSCBUFFERDESC );
6125 bufferDescription.dwFlags = 0;
6126 bufferDescription.dwReserved = 0;
6127 bufferDescription.dwBufferBytes = dsBufferSize;
6128 bufferDescription.lpwfxFormat = &waveFormat;
6130 // Create the capture buffer.
6131 LPDIRECTSOUNDCAPTUREBUFFER buffer;
6132 result = input->CreateCaptureBuffer( &bufferDescription, &buffer, NULL );
6133 if ( FAILED( result ) ) {
6135 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating input buffer (" << dsDevices[ device ].name << ")!";
6136 errorText_ = errorStream_.str();
6140 // Get the buffer size ... might be different from what we specified.
6142 dscbcaps.dwSize = sizeof( DSCBCAPS );
6143 result = buffer->GetCaps( &dscbcaps );
6144 if ( FAILED( result ) ) {
6147 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";
6148 errorText_ = errorStream_.str();
6152 dsBufferSize = dscbcaps.dwBufferBytes;
6154 // NOTE: We could have a problem here if this is a duplex stream
6155 // and the play and capture hardware buffer sizes are different
6156 // (I'm actually not sure if that is a problem or not).
6157 // Currently, we are not verifying that.
6159 // Lock the capture buffer
6162 result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );
6163 if ( FAILED( result ) ) {
6166 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking input buffer (" << dsDevices[ device ].name << ")!";
6167 errorText_ = errorStream_.str();
6172 ZeroMemory( audioPtr, dataLen );
6174 // Unlock the buffer
6175 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
6176 if ( FAILED( result ) ) {
6179 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking input buffer (" << dsDevices[ device ].name << ")!";
6180 errorText_ = errorStream_.str();
6184 ohandle = (void *) input;
6185 bhandle = (void *) buffer;
6188 // Set various stream parameters
6189 DsHandle *handle = 0;
6190 stream_.nDeviceChannels[mode] = channels + firstChannel;
6191 stream_.nUserChannels[mode] = channels;
6192 stream_.bufferSize = *bufferSize;
6193 stream_.channelOffset[mode] = firstChannel;
6194 stream_.deviceInterleaved[mode] = true;
6195 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
6196 else stream_.userInterleaved = true;
6198 // Set flag for buffer conversion
6199 stream_.doConvertBuffer[mode] = false;
6200 if (stream_.nUserChannels[mode] != stream_.nDeviceChannels[mode])
6201 stream_.doConvertBuffer[mode] = true;
6202 if (stream_.userFormat != stream_.deviceFormat[mode])
6203 stream_.doConvertBuffer[mode] = true;
6204 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
6205 stream_.nUserChannels[mode] > 1 )
6206 stream_.doConvertBuffer[mode] = true;
6208 // Allocate necessary internal buffers
6209 long bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
6210 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
6211 if ( stream_.userBuffer[mode] == NULL ) {
6212 errorText_ = "RtApiDs::probeDeviceOpen: error allocating user buffer memory.";
6216 if ( stream_.doConvertBuffer[mode] ) {
6218 bool makeBuffer = true;
6219 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
6220 if ( mode == INPUT ) {
6221 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
6222 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
6223 if ( bufferBytes <= (long) bytesOut ) makeBuffer = false;
6228 bufferBytes *= *bufferSize;
6229 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
6230 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
6231 if ( stream_.deviceBuffer == NULL ) {
6232 errorText_ = "RtApiDs::probeDeviceOpen: error allocating device buffer memory.";
6238 // Allocate our DsHandle structures for the stream.
6239 if ( stream_.apiHandle == 0 ) {
6241 handle = new DsHandle;
6243 catch ( std::bad_alloc& ) {
6244 errorText_ = "RtApiDs::probeDeviceOpen: error allocating AsioHandle memory.";
6248 // Create a manual-reset event.
6249 handle->condition = CreateEvent( NULL, // no security
6250 TRUE, // manual-reset
6251 FALSE, // non-signaled initially
6253 stream_.apiHandle = (void *) handle;
6256 handle = (DsHandle *) stream_.apiHandle;
6257 handle->id[mode] = ohandle;
6258 handle->buffer[mode] = bhandle;
6259 handle->dsBufferSize[mode] = dsBufferSize;
6260 handle->dsPointerLeadTime[mode] = dsPointerLeadTime;
6262 stream_.device[mode] = device;
6263 stream_.state = STREAM_STOPPED;
6264 if ( stream_.mode == OUTPUT && mode == INPUT )
6265 // We had already set up an output stream.
6266 stream_.mode = DUPLEX;
6268 stream_.mode = mode;
6269 stream_.nBuffers = nBuffers;
6270 stream_.sampleRate = sampleRate;
6272 // Setup the buffer conversion information structure.
6273 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
6275 // Setup the callback thread.
6276 if ( stream_.callbackInfo.isRunning == false ) {
6278 stream_.callbackInfo.isRunning = true;
6279 stream_.callbackInfo.object = (void *) this;
6280 stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &callbackHandler,
6281 &stream_.callbackInfo, 0, &threadId );
6282 if ( stream_.callbackInfo.thread == 0 ) {
6283 errorText_ = "RtApiDs::probeDeviceOpen: error creating callback thread!";
6287 // Boost DS thread priority
6288 SetThreadPriority( (HANDLE) stream_.callbackInfo.thread, THREAD_PRIORITY_HIGHEST );
6294 if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
6295 LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
6296 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6297 if ( buffer ) buffer->Release();
6300 if ( handle->buffer[1] ) {
6301 LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
6302 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6303 if ( buffer ) buffer->Release();
6306 CloseHandle( handle->condition );
6308 stream_.apiHandle = 0;
6311 for ( int i=0; i<2; i++ ) {
6312 if ( stream_.userBuffer[i] ) {
6313 free( stream_.userBuffer[i] );
6314 stream_.userBuffer[i] = 0;
6318 if ( stream_.deviceBuffer ) {
6319 free( stream_.deviceBuffer );
6320 stream_.deviceBuffer = 0;
6323 stream_.state = STREAM_CLOSED;
6327 void RtApiDs :: closeStream()
6329 if ( stream_.state == STREAM_CLOSED ) {
6330 errorText_ = "RtApiDs::closeStream(): no open stream to close!";
6331 error( RtAudioError::WARNING );
6335 // Stop the callback thread.
6336 stream_.callbackInfo.isRunning = false;
6337 WaitForSingleObject( (HANDLE) stream_.callbackInfo.thread, INFINITE );
6338 CloseHandle( (HANDLE) stream_.callbackInfo.thread );
6340 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6342 if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
6343 LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
6344 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6351 if ( handle->buffer[1] ) {
6352 LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
6353 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6360 CloseHandle( handle->condition );
6362 stream_.apiHandle = 0;
6365 for ( int i=0; i<2; i++ ) {
6366 if ( stream_.userBuffer[i] ) {
6367 free( stream_.userBuffer[i] );
6368 stream_.userBuffer[i] = 0;
6372 if ( stream_.deviceBuffer ) {
6373 free( stream_.deviceBuffer );
6374 stream_.deviceBuffer = 0;
6377 stream_.mode = UNINITIALIZED;
6378 stream_.state = STREAM_CLOSED;
6381 void RtApiDs :: startStream()
6384 if ( stream_.state == STREAM_RUNNING ) {
6385 errorText_ = "RtApiDs::startStream(): the stream is already running!";
6386 error( RtAudioError::WARNING );
6390 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6392 // Increase scheduler frequency on lesser windows (a side-effect of
6393 // increasing timer accuracy). On greater windows (Win2K or later),
6394 // this is already in effect.
6395 timeBeginPeriod( 1 );
6397 buffersRolling = false;
6398 duplexPrerollBytes = 0;
6400 if ( stream_.mode == DUPLEX ) {
6401 // 0.5 seconds of silence in DUPLEX mode while the devices spin up and synchronize.
6402 duplexPrerollBytes = (int) ( 0.5 * stream_.sampleRate * formatBytes( stream_.deviceFormat[1] ) * stream_.nDeviceChannels[1] );
6406 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
6408 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6409 result = buffer->Play( 0, 0, DSBPLAY_LOOPING );
6410 if ( FAILED( result ) ) {
6411 errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting output buffer!";
6412 errorText_ = errorStream_.str();
6417 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
6419 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6420 result = buffer->Start( DSCBSTART_LOOPING );
6421 if ( FAILED( result ) ) {
6422 errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting input buffer!";
6423 errorText_ = errorStream_.str();
6428 handle->drainCounter = 0;
6429 handle->internalDrain = false;
6430 ResetEvent( handle->condition );
6431 stream_.state = STREAM_RUNNING;
6434 if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );
6437 void RtApiDs :: stopStream()
6440 if ( stream_.state == STREAM_STOPPED ) {
6441 errorText_ = "RtApiDs::stopStream(): the stream is already stopped!";
6442 error( RtAudioError::WARNING );
6449 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6450 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
6451 if ( handle->drainCounter == 0 ) {
6452 handle->drainCounter = 2;
6453 WaitForSingleObject( handle->condition, INFINITE ); // block until signaled
6456 stream_.state = STREAM_STOPPED;
6458 MUTEX_LOCK( &stream_.mutex );
6460 // Stop the buffer and clear memory
6461 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6462 result = buffer->Stop();
6463 if ( FAILED( result ) ) {
6464 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping output buffer!";
6465 errorText_ = errorStream_.str();
6469 // Lock the buffer and clear it so that if we start to play again,
6470 // we won't have old data playing.
6471 result = buffer->Lock( 0, handle->dsBufferSize[0], &audioPtr, &dataLen, NULL, NULL, 0 );
6472 if ( FAILED( result ) ) {
6473 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking output buffer!";
6474 errorText_ = errorStream_.str();
6478 // Zero the DS buffer
6479 ZeroMemory( audioPtr, dataLen );
6481 // Unlock the DS buffer
6482 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
6483 if ( FAILED( result ) ) {
6484 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking output buffer!";
6485 errorText_ = errorStream_.str();
6489 // If we start playing again, we must begin at beginning of buffer.
6490 handle->bufferPointer[0] = 0;
6493 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
6494 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6498 stream_.state = STREAM_STOPPED;
6500 if ( stream_.mode != DUPLEX )
6501 MUTEX_LOCK( &stream_.mutex );
6503 result = buffer->Stop();
6504 if ( FAILED( result ) ) {
6505 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping input buffer!";
6506 errorText_ = errorStream_.str();
6510 // Lock the buffer and clear it so that if we start to play again,
6511 // we won't have old data playing.
6512 result = buffer->Lock( 0, handle->dsBufferSize[1], &audioPtr, &dataLen, NULL, NULL, 0 );
6513 if ( FAILED( result ) ) {
6514 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking input buffer!";
6515 errorText_ = errorStream_.str();
6519 // Zero the DS buffer
6520 ZeroMemory( audioPtr, dataLen );
6522 // Unlock the DS buffer
6523 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
6524 if ( FAILED( result ) ) {
6525 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking input buffer!";
6526 errorText_ = errorStream_.str();
6530 // If we start recording again, we must begin at beginning of buffer.
6531 handle->bufferPointer[1] = 0;
6535 timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows.
6536 MUTEX_UNLOCK( &stream_.mutex );
6538 if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );
6541 void RtApiDs :: abortStream()
6544 if ( stream_.state == STREAM_STOPPED ) {
6545 errorText_ = "RtApiDs::abortStream(): the stream is already stopped!";
6546 error( RtAudioError::WARNING );
6550 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6551 handle->drainCounter = 2;
6556 void RtApiDs :: callbackEvent()
6558 if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) {
6559 Sleep( 50 ); // sleep 50 milliseconds
6563 if ( stream_.state == STREAM_CLOSED ) {
6564 errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!";
6565 error( RtAudioError::WARNING );
6569 CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
6570 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6572 // Check if we were draining the stream and signal is finished.
6573 if ( handle->drainCounter > stream_.nBuffers + 2 ) {
6575 stream_.state = STREAM_STOPPING;
6576 if ( handle->internalDrain == false )
6577 SetEvent( handle->condition );
6583 // Invoke user callback to get fresh output data UNLESS we are
6585 if ( handle->drainCounter == 0 ) {
6586 RtAudioCallback callback = (RtAudioCallback) info->callback;
6587 double streamTime = getStreamTime();
6588 RtAudioStreamStatus status = 0;
6589 if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
6590 status |= RTAUDIO_OUTPUT_UNDERFLOW;
6591 handle->xrun[0] = false;
6593 if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
6594 status |= RTAUDIO_INPUT_OVERFLOW;
6595 handle->xrun[1] = false;
6597 int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
6598 stream_.bufferSize, streamTime, status, info->userData );
6599 if ( cbReturnValue == 2 ) {
6600 stream_.state = STREAM_STOPPING;
6601 handle->drainCounter = 2;
6605 else if ( cbReturnValue == 1 ) {
6606 handle->drainCounter = 1;
6607 handle->internalDrain = true;
6612 DWORD currentWritePointer, safeWritePointer;
6613 DWORD currentReadPointer, safeReadPointer;
6614 UINT nextWritePointer;
6616 LPVOID buffer1 = NULL;
6617 LPVOID buffer2 = NULL;
6618 DWORD bufferSize1 = 0;
6619 DWORD bufferSize2 = 0;
6624 MUTEX_LOCK( &stream_.mutex );
6625 if ( stream_.state == STREAM_STOPPED ) {
6626 MUTEX_UNLOCK( &stream_.mutex );
6630 if ( buffersRolling == false ) {
6631 if ( stream_.mode == DUPLEX ) {
6632 //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
6634 // It takes a while for the devices to get rolling. As a result,
6635 // there's no guarantee that the capture and write device pointers
6636 // will move in lockstep. Wait here for both devices to start
6637 // rolling, and then set our buffer pointers accordingly.
6638 // e.g. Crystal Drivers: the capture buffer starts up 5700 to 9600
6639 // bytes later than the write buffer.
6641 // Stub: a serious risk of having a pre-emptive scheduling round
6642 // take place between the two GetCurrentPosition calls... but I'm
6643 // really not sure how to solve the problem. Temporarily boost to
6644 // Realtime priority, maybe; but I'm not sure what priority the
6645 // DirectSound service threads run at. We *should* be roughly
6646 // within a ms or so of correct.
6648 LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6649 LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6651 DWORD startSafeWritePointer, startSafeReadPointer;
6653 result = dsWriteBuffer->GetCurrentPosition( NULL, &startSafeWritePointer );
6654 if ( FAILED( result ) ) {
6655 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
6656 errorText_ = errorStream_.str();
6657 MUTEX_UNLOCK( &stream_.mutex );
6658 error( RtAudioError::SYSTEM_ERROR );
6661 result = dsCaptureBuffer->GetCurrentPosition( NULL, &startSafeReadPointer );
6662 if ( FAILED( result ) ) {
6663 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
6664 errorText_ = errorStream_.str();
6665 MUTEX_UNLOCK( &stream_.mutex );
6666 error( RtAudioError::SYSTEM_ERROR );
6670 result = dsWriteBuffer->GetCurrentPosition( NULL, &safeWritePointer );
6671 if ( FAILED( result ) ) {
6672 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
6673 errorText_ = errorStream_.str();
6674 MUTEX_UNLOCK( &stream_.mutex );
6675 error( RtAudioError::SYSTEM_ERROR );
6678 result = dsCaptureBuffer->GetCurrentPosition( NULL, &safeReadPointer );
6679 if ( FAILED( result ) ) {
6680 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
6681 errorText_ = errorStream_.str();
6682 MUTEX_UNLOCK( &stream_.mutex );
6683 error( RtAudioError::SYSTEM_ERROR );
6686 if ( safeWritePointer != startSafeWritePointer && safeReadPointer != startSafeReadPointer ) break;
6690 //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
6692 handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];
6693 if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];
6694 handle->bufferPointer[1] = safeReadPointer;
6696 else if ( stream_.mode == OUTPUT ) {
6698 // Set the proper nextWritePosition after initial startup.
6699 LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6700 result = dsWriteBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer );
6701 if ( FAILED( result ) ) {
6702 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
6703 errorText_ = errorStream_.str();
6704 MUTEX_UNLOCK( &stream_.mutex );
6705 error( RtAudioError::SYSTEM_ERROR );
6708 handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];
6709 if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];
6712 buffersRolling = true;
6715 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
6717 LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6719 if ( handle->drainCounter > 1 ) { // write zeros to the output stream
6720 bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
6721 bufferBytes *= formatBytes( stream_.userFormat );
6722 memset( stream_.userBuffer[0], 0, bufferBytes );
6725 // Setup parameters and do buffer conversion if necessary.
6726 if ( stream_.doConvertBuffer[0] ) {
6727 buffer = stream_.deviceBuffer;
6728 convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
6729 bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[0];
6730 bufferBytes *= formatBytes( stream_.deviceFormat[0] );
6733 buffer = stream_.userBuffer[0];
6734 bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
6735 bufferBytes *= formatBytes( stream_.userFormat );
6738 // No byte swapping necessary in DirectSound implementation.
6740 // Ahhh ... windoze. 16-bit data is signed but 8-bit data is
6741 // unsigned. So, we need to convert our signed 8-bit data here to
6743 if ( stream_.deviceFormat[0] == RTAUDIO_SINT8 )
6744 for ( int i=0; i<bufferBytes; i++ ) buffer[i] = (unsigned char) ( buffer[i] + 128 );
6746 DWORD dsBufferSize = handle->dsBufferSize[0];
6747 nextWritePointer = handle->bufferPointer[0];
6749 DWORD endWrite, leadPointer;
6751 // Find out where the read and "safe write" pointers are.
6752 result = dsBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer );
6753 if ( FAILED( result ) ) {
6754 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
6755 errorText_ = errorStream_.str();
6756 MUTEX_UNLOCK( &stream_.mutex );
6757 error( RtAudioError::SYSTEM_ERROR );
6761 // We will copy our output buffer into the region between
6762 // safeWritePointer and leadPointer. If leadPointer is not
6763 // beyond the next endWrite position, wait until it is.
6764 leadPointer = safeWritePointer + handle->dsPointerLeadTime[0];
6765 //std::cout << "safeWritePointer = " << safeWritePointer << ", leadPointer = " << leadPointer << ", nextWritePointer = " << nextWritePointer << std::endl;
6766 if ( leadPointer > dsBufferSize ) leadPointer -= dsBufferSize;
6767 if ( leadPointer < nextWritePointer ) leadPointer += dsBufferSize; // unwrap offset
6768 endWrite = nextWritePointer + bufferBytes;
6770 // Check whether the entire write region is behind the play pointer.
6771 if ( leadPointer >= endWrite ) break;
6773 // If we are here, then we must wait until the leadPointer advances
6774 // beyond the end of our next write region. We use the
6775 // Sleep() function to suspend operation until that happens.
6776 double millis = ( endWrite - leadPointer ) * 1000.0;
6777 millis /= ( formatBytes( stream_.deviceFormat[0]) * stream_.nDeviceChannels[0] * stream_.sampleRate);
6778 if ( millis < 1.0 ) millis = 1.0;
6779 Sleep( (DWORD) millis );
6782 if ( dsPointerBetween( nextWritePointer, safeWritePointer, currentWritePointer, dsBufferSize )
6783 || dsPointerBetween( endWrite, safeWritePointer, currentWritePointer, dsBufferSize ) ) {
6784 // We've strayed into the forbidden zone ... resync the read pointer.
6785 handle->xrun[0] = true;
6786 nextWritePointer = safeWritePointer + handle->dsPointerLeadTime[0] - bufferBytes;
6787 if ( nextWritePointer >= dsBufferSize ) nextWritePointer -= dsBufferSize;
6788 handle->bufferPointer[0] = nextWritePointer;
6789 endWrite = nextWritePointer + bufferBytes;
6792 // Lock free space in the buffer
6793 result = dsBuffer->Lock( nextWritePointer, bufferBytes, &buffer1,
6794 &bufferSize1, &buffer2, &bufferSize2, 0 );
6795 if ( FAILED( result ) ) {
6796 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking buffer during playback!";
6797 errorText_ = errorStream_.str();
6798 MUTEX_UNLOCK( &stream_.mutex );
6799 error( RtAudioError::SYSTEM_ERROR );
6803 // Copy our buffer into the DS buffer
6804 CopyMemory( buffer1, buffer, bufferSize1 );
6805 if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 );
6807 // Update our buffer offset and unlock sound buffer
6808 dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
6809 if ( FAILED( result ) ) {
6810 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!";
6811 errorText_ = errorStream_.str();
6812 MUTEX_UNLOCK( &stream_.mutex );
6813 error( RtAudioError::SYSTEM_ERROR );
6816 nextWritePointer = ( nextWritePointer + bufferSize1 + bufferSize2 ) % dsBufferSize;
6817 handle->bufferPointer[0] = nextWritePointer;
6820 // Don't bother draining input
6821 if ( handle->drainCounter ) {
6822 handle->drainCounter++;
6826 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
6828 // Setup parameters.
6829 if ( stream_.doConvertBuffer[1] ) {
6830 buffer = stream_.deviceBuffer;
6831 bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[1];
6832 bufferBytes *= formatBytes( stream_.deviceFormat[1] );
6835 buffer = stream_.userBuffer[1];
6836 bufferBytes = stream_.bufferSize * stream_.nUserChannels[1];
6837 bufferBytes *= formatBytes( stream_.userFormat );
6840 LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6841 long nextReadPointer = handle->bufferPointer[1];
6842 DWORD dsBufferSize = handle->dsBufferSize[1];
6844 // Find out where the write and "safe read" pointers are.
6845 result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer );
6846 if ( FAILED( result ) ) {
6847 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
6848 errorText_ = errorStream_.str();
6849 MUTEX_UNLOCK( &stream_.mutex );
6850 error( RtAudioError::SYSTEM_ERROR );
6854 if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset
6855 DWORD endRead = nextReadPointer + bufferBytes;
6857 // Handling depends on whether we are INPUT or DUPLEX.
6858 // If we're in INPUT mode then waiting is a good thing. If we're in DUPLEX mode,
6859 // then a wait here will drag the write pointers into the forbidden zone.
6861 // In DUPLEX mode, rather than wait, we will back off the read pointer until
6862 // it's in a safe position. This causes dropouts, but it seems to be the only
6863 // practical way to sync up the read and write pointers reliably, given the
6864 // the very complex relationship between phase and increment of the read and write
6867 // In order to minimize audible dropouts in DUPLEX mode, we will
6868 // provide a pre-roll period of 0.5 seconds in which we return
6869 // zeros from the read buffer while the pointers sync up.
6871 if ( stream_.mode == DUPLEX ) {
6872 if ( safeReadPointer < endRead ) {
6873 if ( duplexPrerollBytes <= 0 ) {
6874 // Pre-roll time over. Be more agressive.
6875 int adjustment = endRead-safeReadPointer;
6877 handle->xrun[1] = true;
6879 // - large adjustments: we've probably run out of CPU cycles, so just resync exactly,
6880 // and perform fine adjustments later.
6881 // - small adjustments: back off by twice as much.
6882 if ( adjustment >= 2*bufferBytes )
6883 nextReadPointer = safeReadPointer-2*bufferBytes;
6885 nextReadPointer = safeReadPointer-bufferBytes-adjustment;
6887 if ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;
6891 // In pre=roll time. Just do it.
6892 nextReadPointer = safeReadPointer - bufferBytes;
6893 while ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;
6895 endRead = nextReadPointer + bufferBytes;
6898 else { // mode == INPUT
6899 while ( safeReadPointer < endRead && stream_.callbackInfo.isRunning ) {
6900 // See comments for playback.
6901 double millis = (endRead - safeReadPointer) * 1000.0;
6902 millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate);
6903 if ( millis < 1.0 ) millis = 1.0;
6904 Sleep( (DWORD) millis );
6906 // Wake up and find out where we are now.
6907 result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer );
6908 if ( FAILED( result ) ) {
6909 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
6910 errorText_ = errorStream_.str();
6911 MUTEX_UNLOCK( &stream_.mutex );
6912 error( RtAudioError::SYSTEM_ERROR );
6916 if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset
6920 // Lock free space in the buffer
6921 result = dsBuffer->Lock( nextReadPointer, bufferBytes, &buffer1,
6922 &bufferSize1, &buffer2, &bufferSize2, 0 );
6923 if ( FAILED( result ) ) {
6924 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking capture buffer!";
6925 errorText_ = errorStream_.str();
6926 MUTEX_UNLOCK( &stream_.mutex );
6927 error( RtAudioError::SYSTEM_ERROR );
6931 if ( duplexPrerollBytes <= 0 ) {
6932 // Copy our buffer into the DS buffer
6933 CopyMemory( buffer, buffer1, bufferSize1 );
6934 if ( buffer2 != NULL ) CopyMemory( buffer+bufferSize1, buffer2, bufferSize2 );
6937 memset( buffer, 0, bufferSize1 );
6938 if ( buffer2 != NULL ) memset( buffer + bufferSize1, 0, bufferSize2 );
6939 duplexPrerollBytes -= bufferSize1 + bufferSize2;
6942 // Update our buffer offset and unlock sound buffer
6943 nextReadPointer = ( nextReadPointer + bufferSize1 + bufferSize2 ) % dsBufferSize;
6944 dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
6945 if ( FAILED( result ) ) {
6946 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking capture buffer!";
6947 errorText_ = errorStream_.str();
6948 MUTEX_UNLOCK( &stream_.mutex );
6949 error( RtAudioError::SYSTEM_ERROR );
6952 handle->bufferPointer[1] = nextReadPointer;
6954 // No byte swapping necessary in DirectSound implementation.
6956 // If necessary, convert 8-bit data from unsigned to signed.
6957 if ( stream_.deviceFormat[1] == RTAUDIO_SINT8 )
6958 for ( int j=0; j<bufferBytes; j++ ) buffer[j] = (signed char) ( buffer[j] - 128 );
6960 // Do buffer conversion if necessary.
6961 if ( stream_.doConvertBuffer[1] )
6962 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
6966 MUTEX_UNLOCK( &stream_.mutex );
6967 RtApi::tickStreamTime();
6970 // Definitions for utility functions and callbacks
6971 // specific to the DirectSound implementation.
6973 static unsigned __stdcall callbackHandler( void *ptr )
6975 CallbackInfo *info = (CallbackInfo *) ptr;
6976 RtApiDs *object = (RtApiDs *) info->object;
6977 bool* isRunning = &info->isRunning;
6979 while ( *isRunning == true ) {
6980 object->callbackEvent();
6987 static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
6988 LPCTSTR description,
6992 struct DsProbeData& probeInfo = *(struct DsProbeData*) lpContext;
6993 std::vector<struct DsDevice>& dsDevices = *probeInfo.dsDevices;
6996 bool validDevice = false;
6997 if ( probeInfo.isInput == true ) {
6999 LPDIRECTSOUNDCAPTURE object;
7001 hr = DirectSoundCaptureCreate( lpguid, &object, NULL );
7002 if ( hr != DS_OK ) return TRUE;
7004 caps.dwSize = sizeof(caps);
7005 hr = object->GetCaps( &caps );
7006 if ( hr == DS_OK ) {
7007 if ( caps.dwChannels > 0 && caps.dwFormats > 0 )
7014 LPDIRECTSOUND object;
7015 hr = DirectSoundCreate( lpguid, &object, NULL );
7016 if ( hr != DS_OK ) return TRUE;
7018 caps.dwSize = sizeof(caps);
7019 hr = object->GetCaps( &caps );
7020 if ( hr == DS_OK ) {
7021 if ( caps.dwFlags & DSCAPS_PRIMARYMONO || caps.dwFlags & DSCAPS_PRIMARYSTEREO )
7027 // If good device, then save its name and guid.
7028 std::string name = convertCharPointerToStdString( description );
7029 //if ( name == "Primary Sound Driver" || name == "Primary Sound Capture Driver" )
7030 if ( lpguid == NULL )
7031 name = "Default Device";
7032 if ( validDevice ) {
7033 for ( unsigned int i=0; i<dsDevices.size(); i++ ) {
7034 if ( dsDevices[i].name == name ) {
7035 dsDevices[i].found = true;
7036 if ( probeInfo.isInput ) {
7037 dsDevices[i].id[1] = lpguid;
7038 dsDevices[i].validId[1] = true;
7041 dsDevices[i].id[0] = lpguid;
7042 dsDevices[i].validId[0] = true;
7050 device.found = true;
7051 if ( probeInfo.isInput ) {
7052 device.id[1] = lpguid;
7053 device.validId[1] = true;
7056 device.id[0] = lpguid;
7057 device.validId[0] = true;
7059 dsDevices.push_back( device );
7065 static const char* getErrorString( int code )
7069 case DSERR_ALLOCATED:
7070 return "Already allocated";
7072 case DSERR_CONTROLUNAVAIL:
7073 return "Control unavailable";
7075 case DSERR_INVALIDPARAM:
7076 return "Invalid parameter";
7078 case DSERR_INVALIDCALL:
7079 return "Invalid call";
7082 return "Generic error";
7084 case DSERR_PRIOLEVELNEEDED:
7085 return "Priority level needed";
7087 case DSERR_OUTOFMEMORY:
7088 return "Out of memory";
7090 case DSERR_BADFORMAT:
7091 return "The sample rate or the channel format is not supported";
7093 case DSERR_UNSUPPORTED:
7094 return "Not supported";
7096 case DSERR_NODRIVER:
7099 case DSERR_ALREADYINITIALIZED:
7100 return "Already initialized";
7102 case DSERR_NOAGGREGATION:
7103 return "No aggregation";
7105 case DSERR_BUFFERLOST:
7106 return "Buffer lost";
7108 case DSERR_OTHERAPPHASPRIO:
7109 return "Another application already has priority";
7111 case DSERR_UNINITIALIZED:
7112 return "Uninitialized";
7115 return "DirectSound unknown error";
7118 //******************** End of __WINDOWS_DS__ *********************//
7122 #if defined(__LINUX_ALSA__)
7124 #include <alsa/asoundlib.h>
7127 // A structure to hold various information related to the ALSA API
7130 snd_pcm_t *handles[2];
7133 pthread_cond_t runnable_cv;
7137 :synchronized(false), runnable(false) { xrun[0] = false; xrun[1] = false; }
7140 static void *alsaCallbackHandler( void * ptr );
7142 RtApiAlsa :: RtApiAlsa()
7144 // Nothing to do here.
7147 RtApiAlsa :: ~RtApiAlsa()
7149 if ( stream_.state != STREAM_CLOSED ) closeStream();
7152 unsigned int RtApiAlsa :: getDeviceCount( void )
7154 unsigned nDevices = 0;
7155 int result, subdevice, card;
7159 // Count cards and devices
7161 snd_card_next( &card );
7162 while ( card >= 0 ) {
7163 sprintf( name, "hw:%d", card );
7164 result = snd_ctl_open( &handle, name, 0 );
7166 errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << ".";
7167 errorText_ = errorStream_.str();
7168 error( RtAudioError::WARNING );
7173 result = snd_ctl_pcm_next_device( handle, &subdevice );
7175 errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
7176 errorText_ = errorStream_.str();
7177 error( RtAudioError::WARNING );
7180 if ( subdevice < 0 )
7185 snd_ctl_close( handle );
7186 snd_card_next( &card );
7189 result = snd_ctl_open( &handle, "default", 0 );
7192 snd_ctl_close( handle );
7198 RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device )
7200 RtAudio::DeviceInfo info;
7201 info.probed = false;
7203 unsigned nDevices = 0;
7204 int result, subdevice, card;
7208 // Count cards and devices
7211 snd_card_next( &card );
7212 while ( card >= 0 ) {
7213 sprintf( name, "hw:%d", card );
7214 result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
7216 errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << ".";
7217 errorText_ = errorStream_.str();
7218 error( RtAudioError::WARNING );
7223 result = snd_ctl_pcm_next_device( chandle, &subdevice );
7225 errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
7226 errorText_ = errorStream_.str();
7227 error( RtAudioError::WARNING );
7230 if ( subdevice < 0 ) break;
7231 if ( nDevices == device ) {
7232 sprintf( name, "hw:%d,%d", card, subdevice );
7238 snd_ctl_close( chandle );
7239 snd_card_next( &card );
7242 result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
7243 if ( result == 0 ) {
7244 if ( nDevices == device ) {
7245 strcpy( name, "default" );
7251 if ( nDevices == 0 ) {
7252 errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!";
7253 error( RtAudioError::INVALID_USE );
7257 if ( device >= nDevices ) {
7258 errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!";
7259 error( RtAudioError::INVALID_USE );
7265 // If a stream is already open, we cannot probe the stream devices.
7266 // Thus, use the saved results.
7267 if ( stream_.state != STREAM_CLOSED &&
7268 ( stream_.device[0] == device || stream_.device[1] == device ) ) {
7269 snd_ctl_close( chandle );
7270 if ( device >= devices_.size() ) {
7271 errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened.";
7272 error( RtAudioError::WARNING );
7275 return devices_[ device ];
7278 int openMode = SND_PCM_ASYNC;
7279 snd_pcm_stream_t stream;
7280 snd_pcm_info_t *pcminfo;
7281 snd_pcm_info_alloca( &pcminfo );
7283 snd_pcm_hw_params_t *params;
7284 snd_pcm_hw_params_alloca( ¶ms );
7286 // First try for playback unless default device (which has subdev -1)
7287 stream = SND_PCM_STREAM_PLAYBACK;
7288 snd_pcm_info_set_stream( pcminfo, stream );
7289 if ( subdevice != -1 ) {
7290 snd_pcm_info_set_device( pcminfo, subdevice );
7291 snd_pcm_info_set_subdevice( pcminfo, 0 );
7293 result = snd_ctl_pcm_info( chandle, pcminfo );
7295 // Device probably doesn't support playback.
7300 result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK );
7302 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
7303 errorText_ = errorStream_.str();
7304 error( RtAudioError::WARNING );
7308 // The device is open ... fill the parameter structure.
7309 result = snd_pcm_hw_params_any( phandle, params );
7311 snd_pcm_close( phandle );
7312 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
7313 errorText_ = errorStream_.str();
7314 error( RtAudioError::WARNING );
7318 // Get output channel information.
7320 result = snd_pcm_hw_params_get_channels_max( params, &value );
7322 snd_pcm_close( phandle );
7323 errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << ".";
7324 errorText_ = errorStream_.str();
7325 error( RtAudioError::WARNING );
7328 info.outputChannels = value;
7329 snd_pcm_close( phandle );
7332 stream = SND_PCM_STREAM_CAPTURE;
7333 snd_pcm_info_set_stream( pcminfo, stream );
7335 // Now try for capture unless default device (with subdev = -1)
7336 if ( subdevice != -1 ) {
7337 result = snd_ctl_pcm_info( chandle, pcminfo );
7338 snd_ctl_close( chandle );
7340 // Device probably doesn't support capture.
7341 if ( info.outputChannels == 0 ) return info;
7342 goto probeParameters;
7346 snd_ctl_close( chandle );
7348 result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
7350 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
7351 errorText_ = errorStream_.str();
7352 error( RtAudioError::WARNING );
7353 if ( info.outputChannels == 0 ) return info;
7354 goto probeParameters;
7357 // The device is open ... fill the parameter structure.
7358 result = snd_pcm_hw_params_any( phandle, params );
7360 snd_pcm_close( phandle );
7361 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
7362 errorText_ = errorStream_.str();
7363 error( RtAudioError::WARNING );
7364 if ( info.outputChannels == 0 ) return info;
7365 goto probeParameters;
7368 result = snd_pcm_hw_params_get_channels_max( params, &value );
7370 snd_pcm_close( phandle );
7371 errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << ".";
7372 errorText_ = errorStream_.str();
7373 error( RtAudioError::WARNING );
7374 if ( info.outputChannels == 0 ) return info;
7375 goto probeParameters;
7377 info.inputChannels = value;
7378 snd_pcm_close( phandle );
7380 // If device opens for both playback and capture, we determine the channels.
7381 if ( info.outputChannels > 0 && info.inputChannels > 0 )
7382 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
7384 // ALSA doesn't provide default devices so we'll use the first available one.
7385 if ( device == 0 && info.outputChannels > 0 )
7386 info.isDefaultOutput = true;
7387 if ( device == 0 && info.inputChannels > 0 )
7388 info.isDefaultInput = true;
7391 // At this point, we just need to figure out the supported data
7392 // formats and sample rates. We'll proceed by opening the device in
7393 // the direction with the maximum number of channels, or playback if
7394 // they are equal. This might limit our sample rate options, but so
7397 if ( info.outputChannels >= info.inputChannels )
7398 stream = SND_PCM_STREAM_PLAYBACK;
7400 stream = SND_PCM_STREAM_CAPTURE;
7401 snd_pcm_info_set_stream( pcminfo, stream );
7403 result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
7405 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
7406 errorText_ = errorStream_.str();
7407 error( RtAudioError::WARNING );
7411 // The device is open ... fill the parameter structure.
7412 result = snd_pcm_hw_params_any( phandle, params );
7414 snd_pcm_close( phandle );
7415 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
7416 errorText_ = errorStream_.str();
7417 error( RtAudioError::WARNING );
7421 // Test our discrete set of sample rate values.
7422 info.sampleRates.clear();
7423 for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {
7424 if ( snd_pcm_hw_params_test_rate( phandle, params, SAMPLE_RATES[i], 0 ) == 0 ) {
7425 info.sampleRates.push_back( SAMPLE_RATES[i] );
7427 if ( !info.preferredSampleRate || ( SAMPLE_RATES[i] <= 48000 && SAMPLE_RATES[i] > info.preferredSampleRate ) )
7428 info.preferredSampleRate = SAMPLE_RATES[i];
7431 if ( info.sampleRates.size() == 0 ) {
7432 snd_pcm_close( phandle );
7433 errorStream_ << "RtApiAlsa::getDeviceInfo: no supported sample rates found for device (" << name << ").";
7434 errorText_ = errorStream_.str();
7435 error( RtAudioError::WARNING );
7439 // Probe the supported data formats ... we don't care about endian-ness just yet
7440 snd_pcm_format_t format;
7441 info.nativeFormats = 0;
7442 format = SND_PCM_FORMAT_S8;
7443 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7444 info.nativeFormats |= RTAUDIO_SINT8;
7445 format = SND_PCM_FORMAT_S16;
7446 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7447 info.nativeFormats |= RTAUDIO_SINT16;
7448 format = SND_PCM_FORMAT_S24;
7449 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7450 info.nativeFormats |= RTAUDIO_SINT24;
7451 format = SND_PCM_FORMAT_S32;
7452 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7453 info.nativeFormats |= RTAUDIO_SINT32;
7454 format = SND_PCM_FORMAT_FLOAT;
7455 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7456 info.nativeFormats |= RTAUDIO_FLOAT32;
7457 format = SND_PCM_FORMAT_FLOAT64;
7458 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7459 info.nativeFormats |= RTAUDIO_FLOAT64;
7461 // Check that we have at least one supported format
7462 if ( info.nativeFormats == 0 ) {
7463 snd_pcm_close( phandle );
7464 errorStream_ << "RtApiAlsa::getDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio.";
7465 errorText_ = errorStream_.str();
7466 error( RtAudioError::WARNING );
7470 // Get the device name
7472 result = snd_card_get_name( card, &cardname );
7473 if ( result >= 0 ) {
7474 sprintf( name, "hw:%s,%d", cardname, subdevice );
7479 // That's all ... close the device and return
7480 snd_pcm_close( phandle );
7485 void RtApiAlsa :: saveDeviceInfo( void )
7489 unsigned int nDevices = getDeviceCount();
7490 devices_.resize( nDevices );
7491 for ( unsigned int i=0; i<nDevices; i++ )
7492 devices_[i] = getDeviceInfo( i );
7495 bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
7496 unsigned int firstChannel, unsigned int sampleRate,
7497 RtAudioFormat format, unsigned int *bufferSize,
7498 RtAudio::StreamOptions *options )
7501 #if defined(__RTAUDIO_DEBUG__)
7503 snd_output_stdio_attach(&out, stderr, 0);
7506 // I'm not using the "plug" interface ... too much inconsistent behavior.
7508 unsigned nDevices = 0;
7509 int result, subdevice, card;
7513 if ( options && options->flags & RTAUDIO_ALSA_USE_DEFAULT )
7514 snprintf(name, sizeof(name), "%s", "default");
7516 // Count cards and devices
7518 snd_card_next( &card );
7519 while ( card >= 0 ) {
7520 sprintf( name, "hw:%d", card );
7521 result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
7523 errorStream_ << "RtApiAlsa::probeDeviceOpen: control open, card = " << card << ", " << snd_strerror( result ) << ".";
7524 errorText_ = errorStream_.str();
7529 result = snd_ctl_pcm_next_device( chandle, &subdevice );
7530 if ( result < 0 ) break;
7531 if ( subdevice < 0 ) break;
7532 if ( nDevices == device ) {
7533 sprintf( name, "hw:%d,%d", card, subdevice );
7534 snd_ctl_close( chandle );
7539 snd_ctl_close( chandle );
7540 snd_card_next( &card );
7543 result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
7544 if ( result == 0 ) {
7545 if ( nDevices == device ) {
7546 strcpy( name, "default" );
7547 snd_ctl_close( chandle );
7552 snd_ctl_close( chandle );
7554 if ( nDevices == 0 ) {
7555 // This should not happen because a check is made before this function is called.
7556 errorText_ = "RtApiAlsa::probeDeviceOpen: no devices found!";
7560 if ( device >= nDevices ) {
7561 // This should not happen because a check is made before this function is called.
7562 errorText_ = "RtApiAlsa::probeDeviceOpen: device ID is invalid!";
7569 // The getDeviceInfo() function will not work for a device that is
7570 // already open. Thus, we'll probe the system before opening a
7571 // stream and save the results for use by getDeviceInfo().
7572 if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) // only do once
7573 this->saveDeviceInfo();
7575 snd_pcm_stream_t stream;
7576 if ( mode == OUTPUT )
7577 stream = SND_PCM_STREAM_PLAYBACK;
7579 stream = SND_PCM_STREAM_CAPTURE;
7582 int openMode = SND_PCM_ASYNC;
7583 result = snd_pcm_open( &phandle, name, stream, openMode );
7585 if ( mode == OUTPUT )
7586 errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for output.";
7588 errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for input.";
7589 errorText_ = errorStream_.str();
7593 // Fill the parameter structure.
7594 snd_pcm_hw_params_t *hw_params;
7595 snd_pcm_hw_params_alloca( &hw_params );
7596 result = snd_pcm_hw_params_any( phandle, hw_params );
7598 snd_pcm_close( phandle );
7599 errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") parameters, " << snd_strerror( result ) << ".";
7600 errorText_ = errorStream_.str();
7604 #if defined(__RTAUDIO_DEBUG__)
7605 fprintf( stderr, "\nRtApiAlsa: dump hardware params just after device open:\n\n" );
7606 snd_pcm_hw_params_dump( hw_params, out );
7609 // Set access ... check user preference.
7610 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) {
7611 stream_.userInterleaved = false;
7612 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
7614 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
7615 stream_.deviceInterleaved[mode] = true;
7618 stream_.deviceInterleaved[mode] = false;
7621 stream_.userInterleaved = true;
7622 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
7624 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
7625 stream_.deviceInterleaved[mode] = false;
7628 stream_.deviceInterleaved[mode] = true;
7632 snd_pcm_close( phandle );
7633 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") access, " << snd_strerror( result ) << ".";
7634 errorText_ = errorStream_.str();
7638 // Determine how to set the device format.
7639 stream_.userFormat = format;
7640 snd_pcm_format_t deviceFormat = SND_PCM_FORMAT_UNKNOWN;
7642 if ( format == RTAUDIO_SINT8 )
7643 deviceFormat = SND_PCM_FORMAT_S8;
7644 else if ( format == RTAUDIO_SINT16 )
7645 deviceFormat = SND_PCM_FORMAT_S16;
7646 else if ( format == RTAUDIO_SINT24 )
7647 deviceFormat = SND_PCM_FORMAT_S24;
7648 else if ( format == RTAUDIO_SINT32 )
7649 deviceFormat = SND_PCM_FORMAT_S32;
7650 else if ( format == RTAUDIO_FLOAT32 )
7651 deviceFormat = SND_PCM_FORMAT_FLOAT;
7652 else if ( format == RTAUDIO_FLOAT64 )
7653 deviceFormat = SND_PCM_FORMAT_FLOAT64;
7655 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat) == 0) {
7656 stream_.deviceFormat[mode] = format;
7660 // The user requested format is not natively supported by the device.
7661 deviceFormat = SND_PCM_FORMAT_FLOAT64;
7662 if ( snd_pcm_hw_params_test_format( phandle, hw_params, deviceFormat ) == 0 ) {
7663 stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
7667 deviceFormat = SND_PCM_FORMAT_FLOAT;
7668 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7669 stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
7673 deviceFormat = SND_PCM_FORMAT_S32;
7674 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7675 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
7679 deviceFormat = SND_PCM_FORMAT_S24;
7680 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7681 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
7685 deviceFormat = SND_PCM_FORMAT_S16;
7686 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7687 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
7691 deviceFormat = SND_PCM_FORMAT_S8;
7692 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7693 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
7697 // If we get here, no supported format was found.
7698 snd_pcm_close( phandle );
7699 errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device " << device << " data format not supported by RtAudio.";
7700 errorText_ = errorStream_.str();
7704 result = snd_pcm_hw_params_set_format( phandle, hw_params, deviceFormat );
7706 snd_pcm_close( phandle );
7707 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") data format, " << snd_strerror( result ) << ".";
7708 errorText_ = errorStream_.str();
7712 // Determine whether byte-swaping is necessary.
7713 stream_.doByteSwap[mode] = false;
7714 if ( deviceFormat != SND_PCM_FORMAT_S8 ) {
7715 result = snd_pcm_format_cpu_endian( deviceFormat );
7717 stream_.doByteSwap[mode] = true;
7718 else if (result < 0) {
7719 snd_pcm_close( phandle );
7720 errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") endian-ness, " << snd_strerror( result ) << ".";
7721 errorText_ = errorStream_.str();
7726 // Set the sample rate.
7727 result = snd_pcm_hw_params_set_rate_near( phandle, hw_params, (unsigned int*) &sampleRate, 0 );
7729 snd_pcm_close( phandle );
7730 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting sample rate on device (" << name << "), " << snd_strerror( result ) << ".";
7731 errorText_ = errorStream_.str();
7735 // Determine the number of channels for this device. We support a possible
7736 // minimum device channel number > than the value requested by the user.
7737 stream_.nUserChannels[mode] = channels;
7739 result = snd_pcm_hw_params_get_channels_max( hw_params, &value );
7740 unsigned int deviceChannels = value;
7741 if ( result < 0 || deviceChannels < channels + firstChannel ) {
7742 snd_pcm_close( phandle );
7743 errorStream_ << "RtApiAlsa::probeDeviceOpen: requested channel parameters not supported by device (" << name << "), " << snd_strerror( result ) << ".";
7744 errorText_ = errorStream_.str();
7748 result = snd_pcm_hw_params_get_channels_min( hw_params, &value );
7750 snd_pcm_close( phandle );
7751 errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting minimum channels for device (" << name << "), " << snd_strerror( result ) << ".";
7752 errorText_ = errorStream_.str();
7755 deviceChannels = value;
7756 if ( deviceChannels < channels + firstChannel ) deviceChannels = channels + firstChannel;
7757 stream_.nDeviceChannels[mode] = deviceChannels;
7759 // Set the device channels.
7760 result = snd_pcm_hw_params_set_channels( phandle, hw_params, deviceChannels );
7762 snd_pcm_close( phandle );
7763 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting channels for device (" << name << "), " << snd_strerror( result ) << ".";
7764 errorText_ = errorStream_.str();
7768 // Set the buffer (or period) size.
7770 snd_pcm_uframes_t periodSize = *bufferSize;
7771 result = snd_pcm_hw_params_set_period_size_near( phandle, hw_params, &periodSize, &dir );
7773 snd_pcm_close( phandle );
7774 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting period size for device (" << name << "), " << snd_strerror( result ) << ".";
7775 errorText_ = errorStream_.str();
7778 *bufferSize = periodSize;
7780 // Set the buffer number, which in ALSA is referred to as the "period".
7781 unsigned int periods = 0;
7782 if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2;
7783 if ( options && options->numberOfBuffers > 0 ) periods = options->numberOfBuffers;
7784 if ( periods < 2 ) periods = 4; // a fairly safe default value
7785 result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir );
7787 snd_pcm_close( phandle );
7788 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << ".";
7789 errorText_ = errorStream_.str();
7793 // If attempting to setup a duplex stream, the bufferSize parameter
7794 // MUST be the same in both directions!
7795 if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {
7796 snd_pcm_close( phandle );
7797 errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ").";
7798 errorText_ = errorStream_.str();
7802 stream_.bufferSize = *bufferSize;
7804 // Install the hardware configuration
7805 result = snd_pcm_hw_params( phandle, hw_params );
7807 snd_pcm_close( phandle );
7808 errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing hardware configuration on device (" << name << "), " << snd_strerror( result ) << ".";
7809 errorText_ = errorStream_.str();
7813 #if defined(__RTAUDIO_DEBUG__)
7814 fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n");
7815 snd_pcm_hw_params_dump( hw_params, out );
7818 // Set the software configuration to fill buffers with zeros and prevent device stopping on xruns.
7819 snd_pcm_sw_params_t *sw_params = NULL;
7820 snd_pcm_sw_params_alloca( &sw_params );
7821 snd_pcm_sw_params_current( phandle, sw_params );
7822 snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize );
7823 snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, ULONG_MAX );
7824 snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 );
7826 // The following two settings were suggested by Theo Veenker
7827 //snd_pcm_sw_params_set_avail_min( phandle, sw_params, *bufferSize );
7828 //snd_pcm_sw_params_set_xfer_align( phandle, sw_params, 1 );
7830 // here are two options for a fix
7831 //snd_pcm_sw_params_set_silence_size( phandle, sw_params, ULONG_MAX );
7832 snd_pcm_uframes_t val;
7833 snd_pcm_sw_params_get_boundary( sw_params, &val );
7834 snd_pcm_sw_params_set_silence_size( phandle, sw_params, val );
7836 result = snd_pcm_sw_params( phandle, sw_params );
7838 snd_pcm_close( phandle );
7839 errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing software configuration on device (" << name << "), " << snd_strerror( result ) << ".";
7840 errorText_ = errorStream_.str();
7844 #if defined(__RTAUDIO_DEBUG__)
7845 fprintf(stderr, "\nRtApiAlsa: dump software params after installation:\n\n");
7846 snd_pcm_sw_params_dump( sw_params, out );
7849 // Set flags for buffer conversion
7850 stream_.doConvertBuffer[mode] = false;
7851 if ( stream_.userFormat != stream_.deviceFormat[mode] )
7852 stream_.doConvertBuffer[mode] = true;
7853 if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
7854 stream_.doConvertBuffer[mode] = true;
7855 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
7856 stream_.nUserChannels[mode] > 1 )
7857 stream_.doConvertBuffer[mode] = true;
7859 // Allocate the ApiHandle if necessary and then save.
7860 AlsaHandle *apiInfo = 0;
7861 if ( stream_.apiHandle == 0 ) {
7863 apiInfo = (AlsaHandle *) new AlsaHandle;
7865 catch ( std::bad_alloc& ) {
7866 errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating AlsaHandle memory.";
7870 if ( pthread_cond_init( &apiInfo->runnable_cv, NULL ) ) {
7871 errorText_ = "RtApiAlsa::probeDeviceOpen: error initializing pthread condition variable.";
7875 stream_.apiHandle = (void *) apiInfo;
7876 apiInfo->handles[0] = 0;
7877 apiInfo->handles[1] = 0;
7880 apiInfo = (AlsaHandle *) stream_.apiHandle;
7882 apiInfo->handles[mode] = phandle;
7885 // Allocate necessary internal buffers.
7886 unsigned long bufferBytes;
7887 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
7888 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
7889 if ( stream_.userBuffer[mode] == NULL ) {
7890 errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating user buffer memory.";
7894 if ( stream_.doConvertBuffer[mode] ) {
7896 bool makeBuffer = true;
7897 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
7898 if ( mode == INPUT ) {
7899 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
7900 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
7901 if ( bufferBytes <= bytesOut ) makeBuffer = false;
7906 bufferBytes *= *bufferSize;
7907 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
7908 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
7909 if ( stream_.deviceBuffer == NULL ) {
7910 errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating device buffer memory.";
7916 stream_.sampleRate = sampleRate;
7917 stream_.nBuffers = periods;
7918 stream_.device[mode] = device;
7919 stream_.state = STREAM_STOPPED;
7921 // Setup the buffer conversion information structure.
7922 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
7924 // Setup thread if necessary.
7925 if ( stream_.mode == OUTPUT && mode == INPUT ) {
7926 // We had already set up an output stream.
7927 stream_.mode = DUPLEX;
7928 // Link the streams if possible.
7929 apiInfo->synchronized = false;
7930 if ( snd_pcm_link( apiInfo->handles[0], apiInfo->handles[1] ) == 0 )
7931 apiInfo->synchronized = true;
7933 errorText_ = "RtApiAlsa::probeDeviceOpen: unable to synchronize input and output devices.";
7934 error( RtAudioError::WARNING );
7938 stream_.mode = mode;
7940 // Setup callback thread.
7941 stream_.callbackInfo.object = (void *) this;
7943 // Set the thread attributes for joinable and realtime scheduling
7944 // priority (optional). The higher priority will only take affect
7945 // if the program is run as root or suid. Note, under Linux
7946 // processes with CAP_SYS_NICE privilege, a user can change
7947 // scheduling policy and priority (thus need not be root). See
7948 // POSIX "capabilities".
7949 pthread_attr_t attr;
7950 pthread_attr_init( &attr );
7951 pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
7952 #ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
7953 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
7954 stream_.callbackInfo.doRealtime = true;
7955 struct sched_param param;
7956 int priority = options->priority;
7957 int min = sched_get_priority_min( SCHED_RR );
7958 int max = sched_get_priority_max( SCHED_RR );
7959 if ( priority < min ) priority = min;
7960 else if ( priority > max ) priority = max;
7961 param.sched_priority = priority;
7963 // Set the policy BEFORE the priority. Otherwise it fails.
7964 pthread_attr_setschedpolicy(&attr, SCHED_RR);
7965 pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM);
7966 // This is definitely required. Otherwise it fails.
7967 pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED);
7968 pthread_attr_setschedparam(&attr, ¶m);
7971 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
7973 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
7976 stream_.callbackInfo.isRunning = true;
7977 result = pthread_create( &stream_.callbackInfo.thread, &attr, alsaCallbackHandler, &stream_.callbackInfo );
7978 pthread_attr_destroy( &attr );
7980 // Failed. Try instead with default attributes.
7981 result = pthread_create( &stream_.callbackInfo.thread, NULL, alsaCallbackHandler, &stream_.callbackInfo );
7983 stream_.callbackInfo.isRunning = false;
7984 errorText_ = "RtApiAlsa::error creating callback thread!";
7994 pthread_cond_destroy( &apiInfo->runnable_cv );
7995 if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
7996 if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
7998 stream_.apiHandle = 0;
8001 if ( phandle) snd_pcm_close( phandle );
8003 for ( int i=0; i<2; i++ ) {
8004 if ( stream_.userBuffer[i] ) {
8005 free( stream_.userBuffer[i] );
8006 stream_.userBuffer[i] = 0;
8010 if ( stream_.deviceBuffer ) {
8011 free( stream_.deviceBuffer );
8012 stream_.deviceBuffer = 0;
8015 stream_.state = STREAM_CLOSED;
8019 void RtApiAlsa :: closeStream()
8021 if ( stream_.state == STREAM_CLOSED ) {
8022 errorText_ = "RtApiAlsa::closeStream(): no open stream to close!";
8023 error( RtAudioError::WARNING );
8027 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
8028 stream_.callbackInfo.isRunning = false;
8029 MUTEX_LOCK( &stream_.mutex );
8030 if ( stream_.state == STREAM_STOPPED ) {
8031 apiInfo->runnable = true;
8032 pthread_cond_signal( &apiInfo->runnable_cv );
8034 MUTEX_UNLOCK( &stream_.mutex );
8035 pthread_join( stream_.callbackInfo.thread, NULL );
8037 if ( stream_.state == STREAM_RUNNING ) {
8038 stream_.state = STREAM_STOPPED;
8039 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
8040 snd_pcm_drop( apiInfo->handles[0] );
8041 if ( stream_.mode == INPUT || stream_.mode == DUPLEX )
8042 snd_pcm_drop( apiInfo->handles[1] );
8046 pthread_cond_destroy( &apiInfo->runnable_cv );
8047 if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
8048 if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
8050 stream_.apiHandle = 0;
8053 for ( int i=0; i<2; i++ ) {
8054 if ( stream_.userBuffer[i] ) {
8055 free( stream_.userBuffer[i] );
8056 stream_.userBuffer[i] = 0;
8060 if ( stream_.deviceBuffer ) {
8061 free( stream_.deviceBuffer );
8062 stream_.deviceBuffer = 0;
8065 stream_.mode = UNINITIALIZED;
8066 stream_.state = STREAM_CLOSED;
8069 void RtApiAlsa :: startStream()
8071 // This method calls snd_pcm_prepare if the device isn't already in that state.
8074 if ( stream_.state == STREAM_RUNNING ) {
8075 errorText_ = "RtApiAlsa::startStream(): the stream is already running!";
8076 error( RtAudioError::WARNING );
8080 MUTEX_LOCK( &stream_.mutex );
8083 snd_pcm_state_t state;
8084 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
8085 snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
8086 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
8087 state = snd_pcm_state( handle[0] );
8088 if ( state != SND_PCM_STATE_PREPARED ) {
8089 result = snd_pcm_prepare( handle[0] );
8091 errorStream_ << "RtApiAlsa::startStream: error preparing output pcm device, " << snd_strerror( result ) << ".";
8092 errorText_ = errorStream_.str();
8098 if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
8099 result = snd_pcm_drop(handle[1]); // fix to remove stale data received since device has been open
8100 state = snd_pcm_state( handle[1] );
8101 if ( state != SND_PCM_STATE_PREPARED ) {
8102 result = snd_pcm_prepare( handle[1] );
8104 errorStream_ << "RtApiAlsa::startStream: error preparing input pcm device, " << snd_strerror( result ) << ".";
8105 errorText_ = errorStream_.str();
8111 stream_.state = STREAM_RUNNING;
8114 apiInfo->runnable = true;
8115 pthread_cond_signal( &apiInfo->runnable_cv );
8116 MUTEX_UNLOCK( &stream_.mutex );
8118 if ( result >= 0 ) return;
8119 error( RtAudioError::SYSTEM_ERROR );
8122 void RtApiAlsa :: stopStream()
8125 if ( stream_.state == STREAM_STOPPED ) {
8126 errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!";
8127 error( RtAudioError::WARNING );
8131 stream_.state = STREAM_STOPPED;
8132 MUTEX_LOCK( &stream_.mutex );
8135 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
8136 snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
8137 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
8138 if ( apiInfo->synchronized )
8139 result = snd_pcm_drop( handle[0] );
8141 result = snd_pcm_drain( handle[0] );
8143 errorStream_ << "RtApiAlsa::stopStream: error draining output pcm device, " << snd_strerror( result ) << ".";
8144 errorText_ = errorStream_.str();
8149 if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
8150 result = snd_pcm_drop( handle[1] );
8152 errorStream_ << "RtApiAlsa::stopStream: error stopping input pcm device, " << snd_strerror( result ) << ".";
8153 errorText_ = errorStream_.str();
8159 apiInfo->runnable = false; // fixes high CPU usage when stopped
8160 MUTEX_UNLOCK( &stream_.mutex );
8162 if ( result >= 0 ) return;
8163 error( RtAudioError::SYSTEM_ERROR );
8166 void RtApiAlsa :: abortStream()
8169 if ( stream_.state == STREAM_STOPPED ) {
8170 errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!";
8171 error( RtAudioError::WARNING );
8175 stream_.state = STREAM_STOPPED;
8176 MUTEX_LOCK( &stream_.mutex );
8179 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
8180 snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
8181 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
8182 result = snd_pcm_drop( handle[0] );
8184 errorStream_ << "RtApiAlsa::abortStream: error aborting output pcm device, " << snd_strerror( result ) << ".";
8185 errorText_ = errorStream_.str();
8190 if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
8191 result = snd_pcm_drop( handle[1] );
8193 errorStream_ << "RtApiAlsa::abortStream: error aborting input pcm device, " << snd_strerror( result ) << ".";
8194 errorText_ = errorStream_.str();
8200 apiInfo->runnable = false; // fixes high CPU usage when stopped
8201 MUTEX_UNLOCK( &stream_.mutex );
8203 if ( result >= 0 ) return;
8204 error( RtAudioError::SYSTEM_ERROR );
8207 void RtApiAlsa :: callbackEvent()
8209 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
8210 if ( stream_.state == STREAM_STOPPED ) {
8211 MUTEX_LOCK( &stream_.mutex );
8212 while ( !apiInfo->runnable )
8213 pthread_cond_wait( &apiInfo->runnable_cv, &stream_.mutex );
8215 if ( stream_.state != STREAM_RUNNING ) {
8216 MUTEX_UNLOCK( &stream_.mutex );
8219 MUTEX_UNLOCK( &stream_.mutex );
8222 if ( stream_.state == STREAM_CLOSED ) {
8223 errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!";
8224 error( RtAudioError::WARNING );
8228 int doStopStream = 0;
8229 RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
8230 double streamTime = getStreamTime();
8231 RtAudioStreamStatus status = 0;
8232 if ( stream_.mode != INPUT && apiInfo->xrun[0] == true ) {
8233 status |= RTAUDIO_OUTPUT_UNDERFLOW;
8234 apiInfo->xrun[0] = false;
8236 if ( stream_.mode != OUTPUT && apiInfo->xrun[1] == true ) {
8237 status |= RTAUDIO_INPUT_OVERFLOW;
8238 apiInfo->xrun[1] = false;
8240 doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
8241 stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
8243 if ( doStopStream == 2 ) {
8248 MUTEX_LOCK( &stream_.mutex );
8250 // The state might change while waiting on a mutex.
8251 if ( stream_.state == STREAM_STOPPED ) goto unlock;
8257 snd_pcm_sframes_t frames;
8258 RtAudioFormat format;
8259 handle = (snd_pcm_t **) apiInfo->handles;
8261 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
8263 // Setup parameters.
8264 if ( stream_.doConvertBuffer[1] ) {
8265 buffer = stream_.deviceBuffer;
8266 channels = stream_.nDeviceChannels[1];
8267 format = stream_.deviceFormat[1];
8270 buffer = stream_.userBuffer[1];
8271 channels = stream_.nUserChannels[1];
8272 format = stream_.userFormat;
8275 // Read samples from device in interleaved/non-interleaved format.
8276 if ( stream_.deviceInterleaved[1] )
8277 result = snd_pcm_readi( handle[1], buffer, stream_.bufferSize );
8279 void *bufs[channels];
8280 size_t offset = stream_.bufferSize * formatBytes( format );
8281 for ( int i=0; i<channels; i++ )
8282 bufs[i] = (void *) (buffer + (i * offset));
8283 result = snd_pcm_readn( handle[1], bufs, stream_.bufferSize );
8286 if ( result < (int) stream_.bufferSize ) {
8287 // Either an error or overrun occured.
8288 if ( result == -EPIPE ) {
8289 snd_pcm_state_t state = snd_pcm_state( handle[1] );
8290 if ( state == SND_PCM_STATE_XRUN ) {
8291 apiInfo->xrun[1] = true;
8292 result = snd_pcm_prepare( handle[1] );
8294 errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after overrun, " << snd_strerror( result ) << ".";
8295 errorText_ = errorStream_.str();
8299 errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
8300 errorText_ = errorStream_.str();
8304 errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << ".";
8305 errorText_ = errorStream_.str();
8307 error( RtAudioError::WARNING );
8311 // Do byte swapping if necessary.
8312 if ( stream_.doByteSwap[1] )
8313 byteSwapBuffer( buffer, stream_.bufferSize * channels, format );
8315 // Do buffer conversion if necessary.
8316 if ( stream_.doConvertBuffer[1] )
8317 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
8319 // Check stream latency
8320 result = snd_pcm_delay( handle[1], &frames );
8321 if ( result == 0 && frames > 0 ) stream_.latency[1] = frames;
8326 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
8328 // Setup parameters and do buffer conversion if necessary.
8329 if ( stream_.doConvertBuffer[0] ) {
8330 buffer = stream_.deviceBuffer;
8331 convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
8332 channels = stream_.nDeviceChannels[0];
8333 format = stream_.deviceFormat[0];
8336 buffer = stream_.userBuffer[0];
8337 channels = stream_.nUserChannels[0];
8338 format = stream_.userFormat;
8341 // Do byte swapping if necessary.
8342 if ( stream_.doByteSwap[0] )
8343 byteSwapBuffer(buffer, stream_.bufferSize * channels, format);
8345 // Write samples to device in interleaved/non-interleaved format.
8346 if ( stream_.deviceInterleaved[0] )
8347 result = snd_pcm_writei( handle[0], buffer, stream_.bufferSize );
8349 void *bufs[channels];
8350 size_t offset = stream_.bufferSize * formatBytes( format );
8351 for ( int i=0; i<channels; i++ )
8352 bufs[i] = (void *) (buffer + (i * offset));
8353 result = snd_pcm_writen( handle[0], bufs, stream_.bufferSize );
8356 if ( result < (int) stream_.bufferSize ) {
8357 // Either an error or underrun occured.
8358 if ( result == -EPIPE ) {
8359 snd_pcm_state_t state = snd_pcm_state( handle[0] );
8360 if ( state == SND_PCM_STATE_XRUN ) {
8361 apiInfo->xrun[0] = true;
8362 result = snd_pcm_prepare( handle[0] );
8364 errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after underrun, " << snd_strerror( result ) << ".";
8365 errorText_ = errorStream_.str();
8368 errorText_ = "RtApiAlsa::callbackEvent: audio write error, underrun.";
8371 errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
8372 errorText_ = errorStream_.str();
8376 errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << ".";
8377 errorText_ = errorStream_.str();
8379 error( RtAudioError::WARNING );
8383 // Check stream latency
8384 result = snd_pcm_delay( handle[0], &frames );
8385 if ( result == 0 && frames > 0 ) stream_.latency[0] = frames;
8389 MUTEX_UNLOCK( &stream_.mutex );
8391 RtApi::tickStreamTime();
8392 if ( doStopStream == 1 ) this->stopStream();
8395 static void *alsaCallbackHandler( void *ptr )
8397 CallbackInfo *info = (CallbackInfo *) ptr;
8398 RtApiAlsa *object = (RtApiAlsa *) info->object;
8399 bool *isRunning = &info->isRunning;
8401 #ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
8402 if ( info->doRealtime ) {
8403 std::cerr << "RtAudio alsa: " <<
8404 (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") <<
8405 "running realtime scheduling" << std::endl;
8409 while ( *isRunning == true ) {
8410 pthread_testcancel();
8411 object->callbackEvent();
8414 pthread_exit( NULL );
8417 //******************** End of __LINUX_ALSA__ *********************//
8420 #if defined(__LINUX_PULSE__)
8422 // Code written by Peter Meerwald, pmeerw@pmeerw.net
8423 // and Tristan Matthews.
8425 #include <pulse/error.h>
8426 #include <pulse/simple.h>
8429 static const unsigned int SUPPORTED_SAMPLERATES[] = { 8000, 16000, 22050, 32000,
8430 44100, 48000, 96000, 0};
8432 struct rtaudio_pa_format_mapping_t {
8433 RtAudioFormat rtaudio_format;
8434 pa_sample_format_t pa_format;
8437 static const rtaudio_pa_format_mapping_t supported_sampleformats[] = {
8438 {RTAUDIO_SINT16, PA_SAMPLE_S16LE},
8439 {RTAUDIO_SINT32, PA_SAMPLE_S32LE},
8440 {RTAUDIO_FLOAT32, PA_SAMPLE_FLOAT32LE},
8441 {0, PA_SAMPLE_INVALID}};
8443 struct PulseAudioHandle {
8447 pthread_cond_t runnable_cv;
8449 PulseAudioHandle() : s_play(0), s_rec(0), runnable(false) { }
8452 RtApiPulse::~RtApiPulse()
8454 if ( stream_.state != STREAM_CLOSED )
8458 unsigned int RtApiPulse::getDeviceCount( void )
8463 RtAudio::DeviceInfo RtApiPulse::getDeviceInfo( unsigned int /*device*/ )
8465 RtAudio::DeviceInfo info;
8467 info.name = "PulseAudio";
8468 info.outputChannels = 2;
8469 info.inputChannels = 2;
8470 info.duplexChannels = 2;
8471 info.isDefaultOutput = true;
8472 info.isDefaultInput = true;
8474 for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr )
8475 info.sampleRates.push_back( *sr );
8477 info.preferredSampleRate = 48000;
8478 info.nativeFormats = RTAUDIO_SINT16 | RTAUDIO_SINT32 | RTAUDIO_FLOAT32;
8483 static void *pulseaudio_callback( void * user )
8485 CallbackInfo *cbi = static_cast<CallbackInfo *>( user );
8486 RtApiPulse *context = static_cast<RtApiPulse *>( cbi->object );
8487 volatile bool *isRunning = &cbi->isRunning;
8489 #ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
8490 if (cbi->doRealtime) {
8491 std::cerr << "RtAudio pulse: " <<
8492 (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") <<
8493 "running realtime scheduling" << std::endl;
8497 while ( *isRunning ) {
8498 pthread_testcancel();
8499 context->callbackEvent();
8502 pthread_exit( NULL );
8505 void RtApiPulse::closeStream( void )
8507 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8509 stream_.callbackInfo.isRunning = false;
8511 MUTEX_LOCK( &stream_.mutex );
8512 if ( stream_.state == STREAM_STOPPED ) {
8513 pah->runnable = true;
8514 pthread_cond_signal( &pah->runnable_cv );
8516 MUTEX_UNLOCK( &stream_.mutex );
8518 pthread_join( pah->thread, 0 );
8519 if ( pah->s_play ) {
8520 pa_simple_flush( pah->s_play, NULL );
8521 pa_simple_free( pah->s_play );
8524 pa_simple_free( pah->s_rec );
8526 pthread_cond_destroy( &pah->runnable_cv );
8528 stream_.apiHandle = 0;
8531 if ( stream_.userBuffer[0] ) {
8532 free( stream_.userBuffer[0] );
8533 stream_.userBuffer[0] = 0;
8535 if ( stream_.userBuffer[1] ) {
8536 free( stream_.userBuffer[1] );
8537 stream_.userBuffer[1] = 0;
8540 stream_.state = STREAM_CLOSED;
8541 stream_.mode = UNINITIALIZED;
8544 void RtApiPulse::callbackEvent( void )
8546 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8548 if ( stream_.state == STREAM_STOPPED ) {
8549 MUTEX_LOCK( &stream_.mutex );
8550 while ( !pah->runnable )
8551 pthread_cond_wait( &pah->runnable_cv, &stream_.mutex );
8553 if ( stream_.state != STREAM_RUNNING ) {
8554 MUTEX_UNLOCK( &stream_.mutex );
8557 MUTEX_UNLOCK( &stream_.mutex );
8560 if ( stream_.state == STREAM_CLOSED ) {
8561 errorText_ = "RtApiPulse::callbackEvent(): the stream is closed ... "
8562 "this shouldn't happen!";
8563 error( RtAudioError::WARNING );
8567 RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
8568 double streamTime = getStreamTime();
8569 RtAudioStreamStatus status = 0;
8570 int doStopStream = callback( stream_.userBuffer[OUTPUT], stream_.userBuffer[INPUT],
8571 stream_.bufferSize, streamTime, status,
8572 stream_.callbackInfo.userData );
8574 if ( doStopStream == 2 ) {
8579 MUTEX_LOCK( &stream_.mutex );
8580 void *pulse_in = stream_.doConvertBuffer[INPUT] ? stream_.deviceBuffer : stream_.userBuffer[INPUT];
8581 void *pulse_out = stream_.doConvertBuffer[OUTPUT] ? stream_.deviceBuffer : stream_.userBuffer[OUTPUT];
8583 if ( stream_.state != STREAM_RUNNING )
8588 if (stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
8589 if ( stream_.doConvertBuffer[OUTPUT] ) {
8590 convertBuffer( stream_.deviceBuffer,
8591 stream_.userBuffer[OUTPUT],
8592 stream_.convertInfo[OUTPUT] );
8593 bytes = stream_.nDeviceChannels[OUTPUT] * stream_.bufferSize *
8594 formatBytes( stream_.deviceFormat[OUTPUT] );
8596 bytes = stream_.nUserChannels[OUTPUT] * stream_.bufferSize *
8597 formatBytes( stream_.userFormat );
8599 if ( pa_simple_write( pah->s_play, pulse_out, bytes, &pa_error ) < 0 ) {
8600 errorStream_ << "RtApiPulse::callbackEvent: audio write error, " <<
8601 pa_strerror( pa_error ) << ".";
8602 errorText_ = errorStream_.str();
8603 error( RtAudioError::WARNING );
8607 if ( stream_.mode == INPUT || stream_.mode == DUPLEX) {
8608 if ( stream_.doConvertBuffer[INPUT] )
8609 bytes = stream_.nDeviceChannels[INPUT] * stream_.bufferSize *
8610 formatBytes( stream_.deviceFormat[INPUT] );
8612 bytes = stream_.nUserChannels[INPUT] * stream_.bufferSize *
8613 formatBytes( stream_.userFormat );
8615 if ( pa_simple_read( pah->s_rec, pulse_in, bytes, &pa_error ) < 0 ) {
8616 errorStream_ << "RtApiPulse::callbackEvent: audio read error, " <<
8617 pa_strerror( pa_error ) << ".";
8618 errorText_ = errorStream_.str();
8619 error( RtAudioError::WARNING );
8621 if ( stream_.doConvertBuffer[INPUT] ) {
8622 convertBuffer( stream_.userBuffer[INPUT],
8623 stream_.deviceBuffer,
8624 stream_.convertInfo[INPUT] );
8629 MUTEX_UNLOCK( &stream_.mutex );
8630 RtApi::tickStreamTime();
8632 if ( doStopStream == 1 )
8636 void RtApiPulse::startStream( void )
8638 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8640 if ( stream_.state == STREAM_CLOSED ) {
8641 errorText_ = "RtApiPulse::startStream(): the stream is not open!";
8642 error( RtAudioError::INVALID_USE );
8645 if ( stream_.state == STREAM_RUNNING ) {
8646 errorText_ = "RtApiPulse::startStream(): the stream is already running!";
8647 error( RtAudioError::WARNING );
8651 MUTEX_LOCK( &stream_.mutex );
8653 stream_.state = STREAM_RUNNING;
8655 pah->runnable = true;
8656 pthread_cond_signal( &pah->runnable_cv );
8657 MUTEX_UNLOCK( &stream_.mutex );
8660 void RtApiPulse::stopStream( void )
8662 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8664 if ( stream_.state == STREAM_CLOSED ) {
8665 errorText_ = "RtApiPulse::stopStream(): the stream is not open!";
8666 error( RtAudioError::INVALID_USE );
8669 if ( stream_.state == STREAM_STOPPED ) {
8670 errorText_ = "RtApiPulse::stopStream(): the stream is already stopped!";
8671 error( RtAudioError::WARNING );
8675 stream_.state = STREAM_STOPPED;
8676 MUTEX_LOCK( &stream_.mutex );
8678 if ( pah && pah->s_play ) {
8680 if ( pa_simple_drain( pah->s_play, &pa_error ) < 0 ) {
8681 errorStream_ << "RtApiPulse::stopStream: error draining output device, " <<
8682 pa_strerror( pa_error ) << ".";
8683 errorText_ = errorStream_.str();
8684 MUTEX_UNLOCK( &stream_.mutex );
8685 error( RtAudioError::SYSTEM_ERROR );
8690 stream_.state = STREAM_STOPPED;
8691 MUTEX_UNLOCK( &stream_.mutex );
8694 void RtApiPulse::abortStream( void )
8696 PulseAudioHandle *pah = static_cast<PulseAudioHandle*>( stream_.apiHandle );
8698 if ( stream_.state == STREAM_CLOSED ) {
8699 errorText_ = "RtApiPulse::abortStream(): the stream is not open!";
8700 error( RtAudioError::INVALID_USE );
8703 if ( stream_.state == STREAM_STOPPED ) {
8704 errorText_ = "RtApiPulse::abortStream(): the stream is already stopped!";
8705 error( RtAudioError::WARNING );
8709 stream_.state = STREAM_STOPPED;
8710 MUTEX_LOCK( &stream_.mutex );
8712 if ( pah && pah->s_play ) {
8714 if ( pa_simple_flush( pah->s_play, &pa_error ) < 0 ) {
8715 errorStream_ << "RtApiPulse::abortStream: error flushing output device, " <<
8716 pa_strerror( pa_error ) << ".";
8717 errorText_ = errorStream_.str();
8718 MUTEX_UNLOCK( &stream_.mutex );
8719 error( RtAudioError::SYSTEM_ERROR );
8724 stream_.state = STREAM_STOPPED;
8725 MUTEX_UNLOCK( &stream_.mutex );
8728 bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode,
8729 unsigned int channels, unsigned int firstChannel,
8730 unsigned int sampleRate, RtAudioFormat format,
8731 unsigned int *bufferSize, RtAudio::StreamOptions *options )
8733 PulseAudioHandle *pah = 0;
8734 unsigned long bufferBytes = 0;
8737 if ( device != 0 ) return false;
8738 if ( mode != INPUT && mode != OUTPUT ) return false;
8739 if ( channels != 1 && channels != 2 ) {
8740 errorText_ = "RtApiPulse::probeDeviceOpen: unsupported number of channels.";
8743 ss.channels = channels;
8745 if ( firstChannel != 0 ) return false;
8747 bool sr_found = false;
8748 for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr ) {
8749 if ( sampleRate == *sr ) {
8751 stream_.sampleRate = sampleRate;
8752 ss.rate = sampleRate;
8757 errorText_ = "RtApiPulse::probeDeviceOpen: unsupported sample rate.";
8762 for ( const rtaudio_pa_format_mapping_t *sf = supported_sampleformats;
8763 sf->rtaudio_format && sf->pa_format != PA_SAMPLE_INVALID; ++sf ) {
8764 if ( format == sf->rtaudio_format ) {
8766 stream_.userFormat = sf->rtaudio_format;
8767 stream_.deviceFormat[mode] = stream_.userFormat;
8768 ss.format = sf->pa_format;
8772 if ( !sf_found ) { // Use internal data format conversion.
8773 stream_.userFormat = format;
8774 stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
8775 ss.format = PA_SAMPLE_FLOAT32LE;
8778 // Set other stream parameters.
8779 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
8780 else stream_.userInterleaved = true;
8781 stream_.deviceInterleaved[mode] = true;
8782 stream_.nBuffers = 1;
8783 stream_.doByteSwap[mode] = false;
8784 stream_.nUserChannels[mode] = channels;
8785 stream_.nDeviceChannels[mode] = channels + firstChannel;
8786 stream_.channelOffset[mode] = 0;
8787 std::string streamName = "RtAudio";
8789 // Set flags for buffer conversion.
8790 stream_.doConvertBuffer[mode] = false;
8791 if ( stream_.userFormat != stream_.deviceFormat[mode] )
8792 stream_.doConvertBuffer[mode] = true;
8793 if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
8794 stream_.doConvertBuffer[mode] = true;
8796 // Allocate necessary internal buffers.
8797 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
8798 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
8799 if ( stream_.userBuffer[mode] == NULL ) {
8800 errorText_ = "RtApiPulse::probeDeviceOpen: error allocating user buffer memory.";
8803 stream_.bufferSize = *bufferSize;
8805 if ( stream_.doConvertBuffer[mode] ) {
8807 bool makeBuffer = true;
8808 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
8809 if ( mode == INPUT ) {
8810 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
8811 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
8812 if ( bufferBytes <= bytesOut ) makeBuffer = false;
8817 bufferBytes *= *bufferSize;
8818 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
8819 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
8820 if ( stream_.deviceBuffer == NULL ) {
8821 errorText_ = "RtApiPulse::probeDeviceOpen: error allocating device buffer memory.";
8827 stream_.device[mode] = device;
8829 // Setup the buffer conversion information structure.
8830 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
8832 if ( !stream_.apiHandle ) {
8833 PulseAudioHandle *pah = new PulseAudioHandle;
8835 errorText_ = "RtApiPulse::probeDeviceOpen: error allocating memory for handle.";
8839 stream_.apiHandle = pah;
8840 if ( pthread_cond_init( &pah->runnable_cv, NULL ) != 0 ) {
8841 errorText_ = "RtApiPulse::probeDeviceOpen: error creating condition variable.";
8845 pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8848 if ( options && !options->streamName.empty() ) streamName = options->streamName;
8851 pa_buffer_attr buffer_attr;
8852 buffer_attr.fragsize = bufferBytes;
8853 buffer_attr.maxlength = -1;
8855 pah->s_rec = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_RECORD, NULL, "Record", &ss, NULL, &buffer_attr, &error );
8856 if ( !pah->s_rec ) {
8857 errorText_ = "RtApiPulse::probeDeviceOpen: error connecting input to PulseAudio server.";
8862 pah->s_play = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_PLAYBACK, NULL, "Playback", &ss, NULL, NULL, &error );
8863 if ( !pah->s_play ) {
8864 errorText_ = "RtApiPulse::probeDeviceOpen: error connecting output to PulseAudio server.";
8872 if ( stream_.mode == UNINITIALIZED )
8873 stream_.mode = mode;
8874 else if ( stream_.mode == mode )
8877 stream_.mode = DUPLEX;
8879 if ( !stream_.callbackInfo.isRunning ) {
8880 stream_.callbackInfo.object = this;
8882 stream_.state = STREAM_STOPPED;
8883 // Set the thread attributes for joinable and realtime scheduling
8884 // priority (optional). The higher priority will only take affect
8885 // if the program is run as root or suid. Note, under Linux
8886 // processes with CAP_SYS_NICE privilege, a user can change
8887 // scheduling policy and priority (thus need not be root). See
8888 // POSIX "capabilities".
8889 pthread_attr_t attr;
8890 pthread_attr_init( &attr );
8891 pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
8892 #ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
8893 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
8894 stream_.callbackInfo.doRealtime = true;
8895 struct sched_param param;
8896 int priority = options->priority;
8897 int min = sched_get_priority_min( SCHED_RR );
8898 int max = sched_get_priority_max( SCHED_RR );
8899 if ( priority < min ) priority = min;
8900 else if ( priority > max ) priority = max;
8901 param.sched_priority = priority;
8903 // Set the policy BEFORE the priority. Otherwise it fails.
8904 pthread_attr_setschedpolicy(&attr, SCHED_RR);
8905 pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM);
8906 // This is definitely required. Otherwise it fails.
8907 pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED);
8908 pthread_attr_setschedparam(&attr, ¶m);
8911 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
8913 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
8916 stream_.callbackInfo.isRunning = true;
8917 int result = pthread_create( &pah->thread, &attr, pulseaudio_callback, (void *)&stream_.callbackInfo);
8918 pthread_attr_destroy(&attr);
8920 // Failed. Try instead with default attributes.
8921 result = pthread_create( &pah->thread, NULL, pulseaudio_callback, (void *)&stream_.callbackInfo);
8923 stream_.callbackInfo.isRunning = false;
8924 errorText_ = "RtApiPulse::probeDeviceOpen: error creating thread.";
8933 if ( pah && stream_.callbackInfo.isRunning ) {
8934 pthread_cond_destroy( &pah->runnable_cv );
8936 stream_.apiHandle = 0;
8939 for ( int i=0; i<2; i++ ) {
8940 if ( stream_.userBuffer[i] ) {
8941 free( stream_.userBuffer[i] );
8942 stream_.userBuffer[i] = 0;
8946 if ( stream_.deviceBuffer ) {
8947 free( stream_.deviceBuffer );
8948 stream_.deviceBuffer = 0;
8951 stream_.state = STREAM_CLOSED;
8955 //******************** End of __LINUX_PULSE__ *********************//
8958 #if defined(__LINUX_OSS__)
8961 #include <sys/ioctl.h>
8964 #include <sys/soundcard.h>
8968 static void *ossCallbackHandler(void * ptr);
8970 // A structure to hold various information related to the OSS API
8973 int id[2]; // device ids
8976 pthread_cond_t runnable;
8979 :triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
8982 RtApiOss :: RtApiOss()
8984 // Nothing to do here.
8987 RtApiOss :: ~RtApiOss()
8989 if ( stream_.state != STREAM_CLOSED ) closeStream();
8992 unsigned int RtApiOss :: getDeviceCount( void )
8994 int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
8995 if ( mixerfd == -1 ) {
8996 errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'.";
8997 error( RtAudioError::WARNING );
9001 oss_sysinfo sysinfo;
9002 if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) {
9004 errorText_ = "RtApiOss::getDeviceCount: error getting sysinfo, OSS version >= 4.0 is required.";
9005 error( RtAudioError::WARNING );
9010 return sysinfo.numaudios;
9013 RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device )
9015 RtAudio::DeviceInfo info;
9016 info.probed = false;
9018 int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
9019 if ( mixerfd == -1 ) {
9020 errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'.";
9021 error( RtAudioError::WARNING );
9025 oss_sysinfo sysinfo;
9026 int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
9027 if ( result == -1 ) {
9029 errorText_ = "RtApiOss::getDeviceInfo: error getting sysinfo, OSS version >= 4.0 is required.";
9030 error( RtAudioError::WARNING );
9034 unsigned nDevices = sysinfo.numaudios;
9035 if ( nDevices == 0 ) {
9037 errorText_ = "RtApiOss::getDeviceInfo: no devices found!";
9038 error( RtAudioError::INVALID_USE );
9042 if ( device >= nDevices ) {
9044 errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!";
9045 error( RtAudioError::INVALID_USE );
9049 oss_audioinfo ainfo;
9051 result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
9053 if ( result == -1 ) {
9054 errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
9055 errorText_ = errorStream_.str();
9056 error( RtAudioError::WARNING );
9061 if ( ainfo.caps & PCM_CAP_OUTPUT ) info.outputChannels = ainfo.max_channels;
9062 if ( ainfo.caps & PCM_CAP_INPUT ) info.inputChannels = ainfo.max_channels;
9063 if ( ainfo.caps & PCM_CAP_DUPLEX ) {
9064 if ( info.outputChannels > 0 && info.inputChannels > 0 && ainfo.caps & PCM_CAP_DUPLEX )
9065 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
9068 // Probe data formats ... do for input
9069 unsigned long mask = ainfo.iformats;
9070 if ( mask & AFMT_S16_LE || mask & AFMT_S16_BE )
9071 info.nativeFormats |= RTAUDIO_SINT16;
9072 if ( mask & AFMT_S8 )
9073 info.nativeFormats |= RTAUDIO_SINT8;
9074 if ( mask & AFMT_S32_LE || mask & AFMT_S32_BE )
9075 info.nativeFormats |= RTAUDIO_SINT32;
9077 if ( mask & AFMT_FLOAT )
9078 info.nativeFormats |= RTAUDIO_FLOAT32;
9080 if ( mask & AFMT_S24_LE || mask & AFMT_S24_BE )
9081 info.nativeFormats |= RTAUDIO_SINT24;
9083 // Check that we have at least one supported format
9084 if ( info.nativeFormats == 0 ) {
9085 errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio.";
9086 errorText_ = errorStream_.str();
9087 error( RtAudioError::WARNING );
9091 // Probe the supported sample rates.
9092 info.sampleRates.clear();
9093 if ( ainfo.nrates ) {
9094 for ( unsigned int i=0; i<ainfo.nrates; i++ ) {
9095 for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
9096 if ( ainfo.rates[i] == SAMPLE_RATES[k] ) {
9097 info.sampleRates.push_back( SAMPLE_RATES[k] );
9099 if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
9100 info.preferredSampleRate = SAMPLE_RATES[k];
9108 // Check min and max rate values;
9109 for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
9110 if ( ainfo.min_rate <= (int) SAMPLE_RATES[k] && ainfo.max_rate >= (int) SAMPLE_RATES[k] ) {
9111 info.sampleRates.push_back( SAMPLE_RATES[k] );
9113 if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
9114 info.preferredSampleRate = SAMPLE_RATES[k];
9119 if ( info.sampleRates.size() == 0 ) {
9120 errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ").";
9121 errorText_ = errorStream_.str();
9122 error( RtAudioError::WARNING );
9126 info.name = ainfo.name;
9133 bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
9134 unsigned int firstChannel, unsigned int sampleRate,
9135 RtAudioFormat format, unsigned int *bufferSize,
9136 RtAudio::StreamOptions *options )
9138 int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
9139 if ( mixerfd == -1 ) {
9140 errorText_ = "RtApiOss::probeDeviceOpen: error opening '/dev/mixer'.";
9144 oss_sysinfo sysinfo;
9145 int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
9146 if ( result == -1 ) {
9148 errorText_ = "RtApiOss::probeDeviceOpen: error getting sysinfo, OSS version >= 4.0 is required.";
9152 unsigned nDevices = sysinfo.numaudios;
9153 if ( nDevices == 0 ) {
9154 // This should not happen because a check is made before this function is called.
9156 errorText_ = "RtApiOss::probeDeviceOpen: no devices found!";
9160 if ( device >= nDevices ) {
9161 // This should not happen because a check is made before this function is called.
9163 errorText_ = "RtApiOss::probeDeviceOpen: device ID is invalid!";
9167 oss_audioinfo ainfo;
9169 result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
9171 if ( result == -1 ) {
9172 errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
9173 errorText_ = errorStream_.str();
9177 // Check if device supports input or output
9178 if ( ( mode == OUTPUT && !( ainfo.caps & PCM_CAP_OUTPUT ) ) ||
9179 ( mode == INPUT && !( ainfo.caps & PCM_CAP_INPUT ) ) ) {
9180 if ( mode == OUTPUT )
9181 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support output.";
9183 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support input.";
9184 errorText_ = errorStream_.str();
9189 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9190 if ( mode == OUTPUT )
9192 else { // mode == INPUT
9193 if (stream_.mode == OUTPUT && stream_.device[0] == device) {
9194 // We just set the same device for playback ... close and reopen for duplex (OSS only).
9195 close( handle->id[0] );
9197 if ( !( ainfo.caps & PCM_CAP_DUPLEX ) ) {
9198 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support duplex mode.";
9199 errorText_ = errorStream_.str();
9202 // Check that the number previously set channels is the same.
9203 if ( stream_.nUserChannels[0] != channels ) {
9204 errorStream_ << "RtApiOss::probeDeviceOpen: input/output channels must be equal for OSS duplex device (" << ainfo.name << ").";
9205 errorText_ = errorStream_.str();
9214 // Set exclusive access if specified.
9215 if ( options && options->flags & RTAUDIO_HOG_DEVICE ) flags |= O_EXCL;
9217 // Try to open the device.
9219 fd = open( ainfo.devnode, flags, 0 );
9221 if ( errno == EBUSY )
9222 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") is busy.";
9224 errorStream_ << "RtApiOss::probeDeviceOpen: error opening device (" << ainfo.name << ").";
9225 errorText_ = errorStream_.str();
9229 // For duplex operation, specifically set this mode (this doesn't seem to work).
9231 if ( flags | O_RDWR ) {
9232 result = ioctl( fd, SNDCTL_DSP_SETDUPLEX, NULL );
9233 if ( result == -1) {
9234 errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ").";
9235 errorText_ = errorStream_.str();
9241 // Check the device channel support.
9242 stream_.nUserChannels[mode] = channels;
9243 if ( ainfo.max_channels < (int)(channels + firstChannel) ) {
9245 errorStream_ << "RtApiOss::probeDeviceOpen: the device (" << ainfo.name << ") does not support requested channel parameters.";
9246 errorText_ = errorStream_.str();
9250 // Set the number of channels.
9251 int deviceChannels = channels + firstChannel;
9252 result = ioctl( fd, SNDCTL_DSP_CHANNELS, &deviceChannels );
9253 if ( result == -1 || deviceChannels < (int)(channels + firstChannel) ) {
9255 errorStream_ << "RtApiOss::probeDeviceOpen: error setting channel parameters on device (" << ainfo.name << ").";
9256 errorText_ = errorStream_.str();
9259 stream_.nDeviceChannels[mode] = deviceChannels;
9261 // Get the data format mask
9263 result = ioctl( fd, SNDCTL_DSP_GETFMTS, &mask );
9264 if ( result == -1 ) {
9266 errorStream_ << "RtApiOss::probeDeviceOpen: error getting device (" << ainfo.name << ") data formats.";
9267 errorText_ = errorStream_.str();
9271 // Determine how to set the device format.
9272 stream_.userFormat = format;
9273 int deviceFormat = -1;
9274 stream_.doByteSwap[mode] = false;
9275 if ( format == RTAUDIO_SINT8 ) {
9276 if ( mask & AFMT_S8 ) {
9277 deviceFormat = AFMT_S8;
9278 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
9281 else if ( format == RTAUDIO_SINT16 ) {
9282 if ( mask & AFMT_S16_NE ) {
9283 deviceFormat = AFMT_S16_NE;
9284 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
9286 else if ( mask & AFMT_S16_OE ) {
9287 deviceFormat = AFMT_S16_OE;
9288 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
9289 stream_.doByteSwap[mode] = true;
9292 else if ( format == RTAUDIO_SINT24 ) {
9293 if ( mask & AFMT_S24_NE ) {
9294 deviceFormat = AFMT_S24_NE;
9295 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
9297 else if ( mask & AFMT_S24_OE ) {
9298 deviceFormat = AFMT_S24_OE;
9299 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
9300 stream_.doByteSwap[mode] = true;
9303 else if ( format == RTAUDIO_SINT32 ) {
9304 if ( mask & AFMT_S32_NE ) {
9305 deviceFormat = AFMT_S32_NE;
9306 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
9308 else if ( mask & AFMT_S32_OE ) {
9309 deviceFormat = AFMT_S32_OE;
9310 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
9311 stream_.doByteSwap[mode] = true;
9315 if ( deviceFormat == -1 ) {
9316 // The user requested format is not natively supported by the device.
9317 if ( mask & AFMT_S16_NE ) {
9318 deviceFormat = AFMT_S16_NE;
9319 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
9321 else if ( mask & AFMT_S32_NE ) {
9322 deviceFormat = AFMT_S32_NE;
9323 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
9325 else if ( mask & AFMT_S24_NE ) {
9326 deviceFormat = AFMT_S24_NE;
9327 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
9329 else if ( mask & AFMT_S16_OE ) {
9330 deviceFormat = AFMT_S16_OE;
9331 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
9332 stream_.doByteSwap[mode] = true;
9334 else if ( mask & AFMT_S32_OE ) {
9335 deviceFormat = AFMT_S32_OE;
9336 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
9337 stream_.doByteSwap[mode] = true;
9339 else if ( mask & AFMT_S24_OE ) {
9340 deviceFormat = AFMT_S24_OE;
9341 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
9342 stream_.doByteSwap[mode] = true;
9344 else if ( mask & AFMT_S8) {
9345 deviceFormat = AFMT_S8;
9346 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
9350 if ( stream_.deviceFormat[mode] == 0 ) {
9351 // This really shouldn't happen ...
9353 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") data format not supported by RtAudio.";
9354 errorText_ = errorStream_.str();
9358 // Set the data format.
9359 int temp = deviceFormat;
9360 result = ioctl( fd, SNDCTL_DSP_SETFMT, &deviceFormat );
9361 if ( result == -1 || deviceFormat != temp ) {
9363 errorStream_ << "RtApiOss::probeDeviceOpen: error setting data format on device (" << ainfo.name << ").";
9364 errorText_ = errorStream_.str();
9368 // Attempt to set the buffer size. According to OSS, the minimum
9369 // number of buffers is two. The supposed minimum buffer size is 16
9370 // bytes, so that will be our lower bound. The argument to this
9371 // call is in the form 0xMMMMSSSS (hex), where the buffer size (in
9372 // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM.
9373 // We'll check the actual value used near the end of the setup
9375 int ossBufferBytes = *bufferSize * formatBytes( stream_.deviceFormat[mode] ) * deviceChannels;
9376 if ( ossBufferBytes < 16 ) ossBufferBytes = 16;
9378 if ( options ) buffers = options->numberOfBuffers;
9379 if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) buffers = 2;
9380 if ( buffers < 2 ) buffers = 3;
9381 temp = ((int) buffers << 16) + (int)( log10( (double)ossBufferBytes ) / log10( 2.0 ) );
9382 result = ioctl( fd, SNDCTL_DSP_SETFRAGMENT, &temp );
9383 if ( result == -1 ) {
9385 errorStream_ << "RtApiOss::probeDeviceOpen: error setting buffer size on device (" << ainfo.name << ").";
9386 errorText_ = errorStream_.str();
9389 stream_.nBuffers = buffers;
9391 // Save buffer size (in sample frames).
9392 *bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels );
9393 stream_.bufferSize = *bufferSize;
9395 // Set the sample rate.
9396 int srate = sampleRate;
9397 result = ioctl( fd, SNDCTL_DSP_SPEED, &srate );
9398 if ( result == -1 ) {
9400 errorStream_ << "RtApiOss::probeDeviceOpen: error setting sample rate (" << sampleRate << ") on device (" << ainfo.name << ").";
9401 errorText_ = errorStream_.str();
9405 // Verify the sample rate setup worked.
9406 if ( abs( srate - (int)sampleRate ) > 100 ) {
9408 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ").";
9409 errorText_ = errorStream_.str();
9412 stream_.sampleRate = sampleRate;
9414 if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device) {
9415 // We're doing duplex setup here.
9416 stream_.deviceFormat[0] = stream_.deviceFormat[1];
9417 stream_.nDeviceChannels[0] = deviceChannels;
9420 // Set interleaving parameters.
9421 stream_.userInterleaved = true;
9422 stream_.deviceInterleaved[mode] = true;
9423 if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
9424 stream_.userInterleaved = false;
9426 // Set flags for buffer conversion
9427 stream_.doConvertBuffer[mode] = false;
9428 if ( stream_.userFormat != stream_.deviceFormat[mode] )
9429 stream_.doConvertBuffer[mode] = true;
9430 if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
9431 stream_.doConvertBuffer[mode] = true;
9432 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
9433 stream_.nUserChannels[mode] > 1 )
9434 stream_.doConvertBuffer[mode] = true;
9436 // Allocate the stream handles if necessary and then save.
9437 if ( stream_.apiHandle == 0 ) {
9439 handle = new OssHandle;
9441 catch ( std::bad_alloc& ) {
9442 errorText_ = "RtApiOss::probeDeviceOpen: error allocating OssHandle memory.";
9446 if ( pthread_cond_init( &handle->runnable, NULL ) ) {
9447 errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable.";
9451 stream_.apiHandle = (void *) handle;
9454 handle = (OssHandle *) stream_.apiHandle;
9456 handle->id[mode] = fd;
9458 // Allocate necessary internal buffers.
9459 unsigned long bufferBytes;
9460 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
9461 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
9462 if ( stream_.userBuffer[mode] == NULL ) {
9463 errorText_ = "RtApiOss::probeDeviceOpen: error allocating user buffer memory.";
9467 if ( stream_.doConvertBuffer[mode] ) {
9469 bool makeBuffer = true;
9470 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
9471 if ( mode == INPUT ) {
9472 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
9473 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
9474 if ( bufferBytes <= bytesOut ) makeBuffer = false;
9479 bufferBytes *= *bufferSize;
9480 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
9481 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
9482 if ( stream_.deviceBuffer == NULL ) {
9483 errorText_ = "RtApiOss::probeDeviceOpen: error allocating device buffer memory.";
9489 stream_.device[mode] = device;
9490 stream_.state = STREAM_STOPPED;
9492 // Setup the buffer conversion information structure.
9493 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
9495 // Setup thread if necessary.
9496 if ( stream_.mode == OUTPUT && mode == INPUT ) {
9497 // We had already set up an output stream.
9498 stream_.mode = DUPLEX;
9499 if ( stream_.device[0] == device ) handle->id[0] = fd;
9502 stream_.mode = mode;
9504 // Setup callback thread.
9505 stream_.callbackInfo.object = (void *) this;
9507 // Set the thread attributes for joinable and realtime scheduling
9508 // priority. The higher priority will only take affect if the
9509 // program is run as root or suid.
9510 pthread_attr_t attr;
9511 pthread_attr_init( &attr );
9512 pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
9513 #ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
9514 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
9515 stream_.callbackInfo.doRealtime = true;
9516 struct sched_param param;
9517 int priority = options->priority;
9518 int min = sched_get_priority_min( SCHED_RR );
9519 int max = sched_get_priority_max( SCHED_RR );
9520 if ( priority < min ) priority = min;
9521 else if ( priority > max ) priority = max;
9522 param.sched_priority = priority;
9524 // Set the policy BEFORE the priority. Otherwise it fails.
9525 pthread_attr_setschedpolicy(&attr, SCHED_RR);
9526 pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM);
9527 // This is definitely required. Otherwise it fails.
9528 pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED);
9529 pthread_attr_setschedparam(&attr, ¶m);
9532 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
9534 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
9537 stream_.callbackInfo.isRunning = true;
9538 result = pthread_create( &stream_.callbackInfo.thread, &attr, ossCallbackHandler, &stream_.callbackInfo );
9539 pthread_attr_destroy( &attr );
9541 // Failed. Try instead with default attributes.
9542 result = pthread_create( &stream_.callbackInfo.thread, NULL, ossCallbackHandler, &stream_.callbackInfo );
9544 stream_.callbackInfo.isRunning = false;
9545 errorText_ = "RtApiOss::error creating callback thread!";
9555 pthread_cond_destroy( &handle->runnable );
9556 if ( handle->id[0] ) close( handle->id[0] );
9557 if ( handle->id[1] ) close( handle->id[1] );
9559 stream_.apiHandle = 0;
9562 for ( int i=0; i<2; i++ ) {
9563 if ( stream_.userBuffer[i] ) {
9564 free( stream_.userBuffer[i] );
9565 stream_.userBuffer[i] = 0;
9569 if ( stream_.deviceBuffer ) {
9570 free( stream_.deviceBuffer );
9571 stream_.deviceBuffer = 0;
9574 stream_.state = STREAM_CLOSED;
9578 void RtApiOss :: closeStream()
9580 if ( stream_.state == STREAM_CLOSED ) {
9581 errorText_ = "RtApiOss::closeStream(): no open stream to close!";
9582 error( RtAudioError::WARNING );
9586 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9587 stream_.callbackInfo.isRunning = false;
9588 MUTEX_LOCK( &stream_.mutex );
9589 if ( stream_.state == STREAM_STOPPED )
9590 pthread_cond_signal( &handle->runnable );
9591 MUTEX_UNLOCK( &stream_.mutex );
9592 pthread_join( stream_.callbackInfo.thread, NULL );
9594 if ( stream_.state == STREAM_RUNNING ) {
9595 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
9596 ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
9598 ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
9599 stream_.state = STREAM_STOPPED;
9603 pthread_cond_destroy( &handle->runnable );
9604 if ( handle->id[0] ) close( handle->id[0] );
9605 if ( handle->id[1] ) close( handle->id[1] );
9607 stream_.apiHandle = 0;
9610 for ( int i=0; i<2; i++ ) {
9611 if ( stream_.userBuffer[i] ) {
9612 free( stream_.userBuffer[i] );
9613 stream_.userBuffer[i] = 0;
9617 if ( stream_.deviceBuffer ) {
9618 free( stream_.deviceBuffer );
9619 stream_.deviceBuffer = 0;
9622 stream_.mode = UNINITIALIZED;
9623 stream_.state = STREAM_CLOSED;
9626 void RtApiOss :: startStream()
9629 if ( stream_.state == STREAM_RUNNING ) {
9630 errorText_ = "RtApiOss::startStream(): the stream is already running!";
9631 error( RtAudioError::WARNING );
9635 MUTEX_LOCK( &stream_.mutex );
9637 stream_.state = STREAM_RUNNING;
9639 // No need to do anything else here ... OSS automatically starts
9640 // when fed samples.
9642 MUTEX_UNLOCK( &stream_.mutex );
9644 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9645 pthread_cond_signal( &handle->runnable );
9648 void RtApiOss :: stopStream()
9651 if ( stream_.state == STREAM_STOPPED ) {
9652 errorText_ = "RtApiOss::stopStream(): the stream is already stopped!";
9653 error( RtAudioError::WARNING );
9657 MUTEX_LOCK( &stream_.mutex );
9659 // The state might change while waiting on a mutex.
9660 if ( stream_.state == STREAM_STOPPED ) {
9661 MUTEX_UNLOCK( &stream_.mutex );
9666 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9667 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
9669 // Flush the output with zeros a few times.
9672 RtAudioFormat format;
9674 if ( stream_.doConvertBuffer[0] ) {
9675 buffer = stream_.deviceBuffer;
9676 samples = stream_.bufferSize * stream_.nDeviceChannels[0];
9677 format = stream_.deviceFormat[0];
9680 buffer = stream_.userBuffer[0];
9681 samples = stream_.bufferSize * stream_.nUserChannels[0];
9682 format = stream_.userFormat;
9685 memset( buffer, 0, samples * formatBytes(format) );
9686 for ( unsigned int i=0; i<stream_.nBuffers+1; i++ ) {
9687 result = write( handle->id[0], buffer, samples * formatBytes(format) );
9688 if ( result == -1 ) {
9689 errorText_ = "RtApiOss::stopStream: audio write error.";
9690 error( RtAudioError::WARNING );
9694 result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
9695 if ( result == -1 ) {
9696 errorStream_ << "RtApiOss::stopStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
9697 errorText_ = errorStream_.str();
9700 handle->triggered = false;
9703 if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
9704 result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
9705 if ( result == -1 ) {
9706 errorStream_ << "RtApiOss::stopStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
9707 errorText_ = errorStream_.str();
9713 stream_.state = STREAM_STOPPED;
9714 MUTEX_UNLOCK( &stream_.mutex );
9716 if ( result != -1 ) return;
9717 error( RtAudioError::SYSTEM_ERROR );
9720 void RtApiOss :: abortStream()
9723 if ( stream_.state == STREAM_STOPPED ) {
9724 errorText_ = "RtApiOss::abortStream(): the stream is already stopped!";
9725 error( RtAudioError::WARNING );
9729 MUTEX_LOCK( &stream_.mutex );
9731 // The state might change while waiting on a mutex.
9732 if ( stream_.state == STREAM_STOPPED ) {
9733 MUTEX_UNLOCK( &stream_.mutex );
9738 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9739 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
9740 result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
9741 if ( result == -1 ) {
9742 errorStream_ << "RtApiOss::abortStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
9743 errorText_ = errorStream_.str();
9746 handle->triggered = false;
9749 if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
9750 result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
9751 if ( result == -1 ) {
9752 errorStream_ << "RtApiOss::abortStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
9753 errorText_ = errorStream_.str();
9759 stream_.state = STREAM_STOPPED;
9760 MUTEX_UNLOCK( &stream_.mutex );
9762 if ( result != -1 ) return;
9763 error( RtAudioError::SYSTEM_ERROR );
9766 void RtApiOss :: callbackEvent()
9768 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9769 if ( stream_.state == STREAM_STOPPED ) {
9770 MUTEX_LOCK( &stream_.mutex );
9771 pthread_cond_wait( &handle->runnable, &stream_.mutex );
9772 if ( stream_.state != STREAM_RUNNING ) {
9773 MUTEX_UNLOCK( &stream_.mutex );
9776 MUTEX_UNLOCK( &stream_.mutex );
9779 if ( stream_.state == STREAM_CLOSED ) {
9780 errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!";
9781 error( RtAudioError::WARNING );
9785 // Invoke user callback to get fresh output data.
9786 int doStopStream = 0;
9787 RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
9788 double streamTime = getStreamTime();
9789 RtAudioStreamStatus status = 0;
9790 if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
9791 status |= RTAUDIO_OUTPUT_UNDERFLOW;
9792 handle->xrun[0] = false;
9794 if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
9795 status |= RTAUDIO_INPUT_OVERFLOW;
9796 handle->xrun[1] = false;
9798 doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
9799 stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
9800 if ( doStopStream == 2 ) {
9801 this->abortStream();
9805 MUTEX_LOCK( &stream_.mutex );
9807 // The state might change while waiting on a mutex.
9808 if ( stream_.state == STREAM_STOPPED ) goto unlock;
9813 RtAudioFormat format;
9815 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
9817 // Setup parameters and do buffer conversion if necessary.
9818 if ( stream_.doConvertBuffer[0] ) {
9819 buffer = stream_.deviceBuffer;
9820 convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
9821 samples = stream_.bufferSize * stream_.nDeviceChannels[0];
9822 format = stream_.deviceFormat[0];
9825 buffer = stream_.userBuffer[0];
9826 samples = stream_.bufferSize * stream_.nUserChannels[0];
9827 format = stream_.userFormat;
9830 // Do byte swapping if necessary.
9831 if ( stream_.doByteSwap[0] )
9832 byteSwapBuffer( buffer, samples, format );
9834 if ( stream_.mode == DUPLEX && handle->triggered == false ) {
9836 ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
9837 result = write( handle->id[0], buffer, samples * formatBytes(format) );
9838 trig = PCM_ENABLE_INPUT|PCM_ENABLE_OUTPUT;
9839 ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
9840 handle->triggered = true;
9843 // Write samples to device.
9844 result = write( handle->id[0], buffer, samples * formatBytes(format) );
9846 if ( result == -1 ) {
9847 // We'll assume this is an underrun, though there isn't a
9848 // specific means for determining that.
9849 handle->xrun[0] = true;
9850 errorText_ = "RtApiOss::callbackEvent: audio write error.";
9851 error( RtAudioError::WARNING );
9852 // Continue on to input section.
9856 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
9858 // Setup parameters.
9859 if ( stream_.doConvertBuffer[1] ) {
9860 buffer = stream_.deviceBuffer;
9861 samples = stream_.bufferSize * stream_.nDeviceChannels[1];
9862 format = stream_.deviceFormat[1];
9865 buffer = stream_.userBuffer[1];
9866 samples = stream_.bufferSize * stream_.nUserChannels[1];
9867 format = stream_.userFormat;
9870 // Read samples from device.
9871 result = read( handle->id[1], buffer, samples * formatBytes(format) );
9873 if ( result == -1 ) {
9874 // We'll assume this is an overrun, though there isn't a
9875 // specific means for determining that.
9876 handle->xrun[1] = true;
9877 errorText_ = "RtApiOss::callbackEvent: audio read error.";
9878 error( RtAudioError::WARNING );
9882 // Do byte swapping if necessary.
9883 if ( stream_.doByteSwap[1] )
9884 byteSwapBuffer( buffer, samples, format );
9886 // Do buffer conversion if necessary.
9887 if ( stream_.doConvertBuffer[1] )
9888 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
9892 MUTEX_UNLOCK( &stream_.mutex );
9894 RtApi::tickStreamTime();
9895 if ( doStopStream == 1 ) this->stopStream();
9898 static void *ossCallbackHandler( void *ptr )
9900 CallbackInfo *info = (CallbackInfo *) ptr;
9901 RtApiOss *object = (RtApiOss *) info->object;
9902 bool *isRunning = &info->isRunning;
9904 #ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
9905 if (info->doRealtime) {
9906 std::cerr << "RtAudio oss: " <<
9907 (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") <<
9908 "running realtime scheduling" << std::endl;
9912 while ( *isRunning == true ) {
9913 pthread_testcancel();
9914 object->callbackEvent();
9917 pthread_exit( NULL );
9920 //******************** End of __LINUX_OSS__ *********************//
9924 // *************************************************** //
9926 // Protected common (OS-independent) RtAudio methods.
9928 // *************************************************** //
9930 // This method can be modified to control the behavior of error
9931 // message printing.
9932 void RtApi :: error( RtAudioError::Type type )
9934 errorStream_.str(""); // clear the ostringstream
9936 RtAudioErrorCallback errorCallback = (RtAudioErrorCallback) stream_.callbackInfo.errorCallback;
9937 if ( errorCallback ) {
9938 // abortStream() can generate new error messages. Ignore them. Just keep original one.
9940 if ( firstErrorOccurred_ )
9943 firstErrorOccurred_ = true;
9944 const std::string errorMessage = errorText_;
9946 if ( type != RtAudioError::WARNING && stream_.state != STREAM_STOPPED) {
9947 stream_.callbackInfo.isRunning = false; // exit from the thread
9951 errorCallback( type, errorMessage );
9952 firstErrorOccurred_ = false;
9956 if ( type == RtAudioError::WARNING && showWarnings_ == true )
9957 std::cerr << '\n' << errorText_ << "\n\n";
9958 else if ( type != RtAudioError::WARNING )
9959 throw( RtAudioError( errorText_, type ) );
9962 void RtApi :: verifyStream()
9964 if ( stream_.state == STREAM_CLOSED ) {
9965 errorText_ = "RtApi:: a stream is not open!";
9966 error( RtAudioError::INVALID_USE );
9970 void RtApi :: clearStreamInfo()
9972 stream_.mode = UNINITIALIZED;
9973 stream_.state = STREAM_CLOSED;
9974 stream_.sampleRate = 0;
9975 stream_.bufferSize = 0;
9976 stream_.nBuffers = 0;
9977 stream_.userFormat = 0;
9978 stream_.userInterleaved = true;
9979 stream_.streamTime = 0.0;
9980 stream_.apiHandle = 0;
9981 stream_.deviceBuffer = 0;
9982 stream_.callbackInfo.callback = 0;
9983 stream_.callbackInfo.userData = 0;
9984 stream_.callbackInfo.isRunning = false;
9985 stream_.callbackInfo.errorCallback = 0;
9986 for ( int i=0; i<2; i++ ) {
9987 stream_.device[i] = 11111;
9988 stream_.doConvertBuffer[i] = false;
9989 stream_.deviceInterleaved[i] = true;
9990 stream_.doByteSwap[i] = false;
9991 stream_.nUserChannels[i] = 0;
9992 stream_.nDeviceChannels[i] = 0;
9993 stream_.channelOffset[i] = 0;
9994 stream_.deviceFormat[i] = 0;
9995 stream_.latency[i] = 0;
9996 stream_.userBuffer[i] = 0;
9997 stream_.convertInfo[i].channels = 0;
9998 stream_.convertInfo[i].inJump = 0;
9999 stream_.convertInfo[i].outJump = 0;
10000 stream_.convertInfo[i].inFormat = 0;
10001 stream_.convertInfo[i].outFormat = 0;
10002 stream_.convertInfo[i].inOffset.clear();
10003 stream_.convertInfo[i].outOffset.clear();
10007 unsigned int RtApi :: formatBytes( RtAudioFormat format )
10009 if ( format == RTAUDIO_SINT16 )
10011 else if ( format == RTAUDIO_SINT32 || format == RTAUDIO_FLOAT32 )
10013 else if ( format == RTAUDIO_FLOAT64 )
10015 else if ( format == RTAUDIO_SINT24 )
10017 else if ( format == RTAUDIO_SINT8 )
10020 errorText_ = "RtApi::formatBytes: undefined format.";
10021 error( RtAudioError::WARNING );
10026 void RtApi :: setConvertInfo( StreamMode mode, unsigned int firstChannel )
10028 if ( mode == INPUT ) { // convert device to user buffer
10029 stream_.convertInfo[mode].inJump = stream_.nDeviceChannels[1];
10030 stream_.convertInfo[mode].outJump = stream_.nUserChannels[1];
10031 stream_.convertInfo[mode].inFormat = stream_.deviceFormat[1];
10032 stream_.convertInfo[mode].outFormat = stream_.userFormat;
10034 else { // convert user to device buffer
10035 stream_.convertInfo[mode].inJump = stream_.nUserChannels[0];
10036 stream_.convertInfo[mode].outJump = stream_.nDeviceChannels[0];
10037 stream_.convertInfo[mode].inFormat = stream_.userFormat;
10038 stream_.convertInfo[mode].outFormat = stream_.deviceFormat[0];
10041 if ( stream_.convertInfo[mode].inJump < stream_.convertInfo[mode].outJump )
10042 stream_.convertInfo[mode].channels = stream_.convertInfo[mode].inJump;
10044 stream_.convertInfo[mode].channels = stream_.convertInfo[mode].outJump;
10046 // Set up the interleave/deinterleave offsets.
10047 if ( stream_.deviceInterleaved[mode] != stream_.userInterleaved ) {
10048 if ( ( mode == OUTPUT && stream_.deviceInterleaved[mode] ) ||
10049 ( mode == INPUT && stream_.userInterleaved ) ) {
10050 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
10051 stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
10052 stream_.convertInfo[mode].outOffset.push_back( k );
10053 stream_.convertInfo[mode].inJump = 1;
10057 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
10058 stream_.convertInfo[mode].inOffset.push_back( k );
10059 stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
10060 stream_.convertInfo[mode].outJump = 1;
10064 else { // no (de)interleaving
10065 if ( stream_.userInterleaved ) {
10066 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
10067 stream_.convertInfo[mode].inOffset.push_back( k );
10068 stream_.convertInfo[mode].outOffset.push_back( k );
10072 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
10073 stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
10074 stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
10075 stream_.convertInfo[mode].inJump = 1;
10076 stream_.convertInfo[mode].outJump = 1;
10081 // Add channel offset.
10082 if ( firstChannel > 0 ) {
10083 if ( stream_.deviceInterleaved[mode] ) {
10084 if ( mode == OUTPUT ) {
10085 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
10086 stream_.convertInfo[mode].outOffset[k] += firstChannel;
10089 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
10090 stream_.convertInfo[mode].inOffset[k] += firstChannel;
10094 if ( mode == OUTPUT ) {
10095 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
10096 stream_.convertInfo[mode].outOffset[k] += ( firstChannel * stream_.bufferSize );
10099 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
10100 stream_.convertInfo[mode].inOffset[k] += ( firstChannel * stream_.bufferSize );
10106 void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info )
10108 // This function does format conversion, input/output channel compensation, and
10109 // data interleaving/deinterleaving. 24-bit integers are assumed to occupy
10110 // the lower three bytes of a 32-bit integer.
10112 // Clear our device buffer when in/out duplex device channels are different
10113 if ( outBuffer == stream_.deviceBuffer && stream_.mode == DUPLEX &&
10114 ( stream_.nDeviceChannels[0] < stream_.nDeviceChannels[1] ) )
10115 memset( outBuffer, 0, stream_.bufferSize * info.outJump * formatBytes( info.outFormat ) );
10118 if (info.outFormat == RTAUDIO_FLOAT64) {
10120 Float64 *out = (Float64 *)outBuffer;
10122 if (info.inFormat == RTAUDIO_SINT8) {
10123 signed char *in = (signed char *)inBuffer;
10124 scale = 1.0 / 127.5;
10125 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10126 for (j=0; j<info.channels; j++) {
10127 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
10128 out[info.outOffset[j]] += 0.5;
10129 out[info.outOffset[j]] *= scale;
10132 out += info.outJump;
10135 else if (info.inFormat == RTAUDIO_SINT16) {
10136 Int16 *in = (Int16 *)inBuffer;
10137 scale = 1.0 / 32767.5;
10138 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10139 for (j=0; j<info.channels; j++) {
10140 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
10141 out[info.outOffset[j]] += 0.5;
10142 out[info.outOffset[j]] *= scale;
10145 out += info.outJump;
10148 else if (info.inFormat == RTAUDIO_SINT24) {
10149 Int24 *in = (Int24 *)inBuffer;
10150 scale = 1.0 / 8388607.5;
10151 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10152 for (j=0; j<info.channels; j++) {
10153 out[info.outOffset[j]] = (Float64) (in[info.inOffset[j]].asInt());
10154 out[info.outOffset[j]] += 0.5;
10155 out[info.outOffset[j]] *= scale;
10158 out += info.outJump;
10161 else if (info.inFormat == RTAUDIO_SINT32) {
10162 Int32 *in = (Int32 *)inBuffer;
10163 scale = 1.0 / 2147483647.5;
10164 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10165 for (j=0; j<info.channels; j++) {
10166 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
10167 out[info.outOffset[j]] += 0.5;
10168 out[info.outOffset[j]] *= scale;
10171 out += info.outJump;
10174 else if (info.inFormat == RTAUDIO_FLOAT32) {
10175 Float32 *in = (Float32 *)inBuffer;
10176 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10177 for (j=0; j<info.channels; j++) {
10178 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
10181 out += info.outJump;
10184 else if (info.inFormat == RTAUDIO_FLOAT64) {
10185 // Channel compensation and/or (de)interleaving only.
10186 Float64 *in = (Float64 *)inBuffer;
10187 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10188 for (j=0; j<info.channels; j++) {
10189 out[info.outOffset[j]] = in[info.inOffset[j]];
10192 out += info.outJump;
10196 else if (info.outFormat == RTAUDIO_FLOAT32) {
10198 Float32 *out = (Float32 *)outBuffer;
10200 if (info.inFormat == RTAUDIO_SINT8) {
10201 signed char *in = (signed char *)inBuffer;
10202 scale = (Float32) ( 1.0 / 127.5 );
10203 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10204 for (j=0; j<info.channels; j++) {
10205 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
10206 out[info.outOffset[j]] += 0.5;
10207 out[info.outOffset[j]] *= scale;
10210 out += info.outJump;
10213 else if (info.inFormat == RTAUDIO_SINT16) {
10214 Int16 *in = (Int16 *)inBuffer;
10215 scale = (Float32) ( 1.0 / 32767.5 );
10216 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10217 for (j=0; j<info.channels; j++) {
10218 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
10219 out[info.outOffset[j]] += 0.5;
10220 out[info.outOffset[j]] *= scale;
10223 out += info.outJump;
10226 else if (info.inFormat == RTAUDIO_SINT24) {
10227 Int24 *in = (Int24 *)inBuffer;
10228 scale = (Float32) ( 1.0 / 8388607.5 );
10229 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10230 for (j=0; j<info.channels; j++) {
10231 out[info.outOffset[j]] = (Float32) (in[info.inOffset[j]].asInt());
10232 out[info.outOffset[j]] += 0.5;
10233 out[info.outOffset[j]] *= scale;
10236 out += info.outJump;
10239 else if (info.inFormat == RTAUDIO_SINT32) {
10240 Int32 *in = (Int32 *)inBuffer;
10241 scale = (Float32) ( 1.0 / 2147483647.5 );
10242 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10243 for (j=0; j<info.channels; j++) {
10244 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
10245 out[info.outOffset[j]] += 0.5;
10246 out[info.outOffset[j]] *= scale;
10249 out += info.outJump;
10252 else if (info.inFormat == RTAUDIO_FLOAT32) {
10253 // Channel compensation and/or (de)interleaving only.
10254 Float32 *in = (Float32 *)inBuffer;
10255 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10256 for (j=0; j<info.channels; j++) {
10257 out[info.outOffset[j]] = in[info.inOffset[j]];
10260 out += info.outJump;
10263 else if (info.inFormat == RTAUDIO_FLOAT64) {
10264 Float64 *in = (Float64 *)inBuffer;
10265 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10266 for (j=0; j<info.channels; j++) {
10267 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
10270 out += info.outJump;
10274 else if (info.outFormat == RTAUDIO_SINT32) {
10275 Int32 *out = (Int32 *)outBuffer;
10276 if (info.inFormat == RTAUDIO_SINT8) {
10277 signed char *in = (signed char *)inBuffer;
10278 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10279 for (j=0; j<info.channels; j++) {
10280 out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
10281 out[info.outOffset[j]] <<= 24;
10284 out += info.outJump;
10287 else if (info.inFormat == RTAUDIO_SINT16) {
10288 Int16 *in = (Int16 *)inBuffer;
10289 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10290 for (j=0; j<info.channels; j++) {
10291 out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
10292 out[info.outOffset[j]] <<= 16;
10295 out += info.outJump;
10298 else if (info.inFormat == RTAUDIO_SINT24) {
10299 Int24 *in = (Int24 *)inBuffer;
10300 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10301 for (j=0; j<info.channels; j++) {
10302 out[info.outOffset[j]] = (Int32) in[info.inOffset[j]].asInt();
10303 out[info.outOffset[j]] <<= 8;
10306 out += info.outJump;
10309 else if (info.inFormat == RTAUDIO_SINT32) {
10310 // Channel compensation and/or (de)interleaving only.
10311 Int32 *in = (Int32 *)inBuffer;
10312 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10313 for (j=0; j<info.channels; j++) {
10314 out[info.outOffset[j]] = in[info.inOffset[j]];
10317 out += info.outJump;
10320 else if (info.inFormat == RTAUDIO_FLOAT32) {
10321 Float32 *in = (Float32 *)inBuffer;
10322 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10323 for (j=0; j<info.channels; j++) {
10324 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
10327 out += info.outJump;
10330 else if (info.inFormat == RTAUDIO_FLOAT64) {
10331 Float64 *in = (Float64 *)inBuffer;
10332 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10333 for (j=0; j<info.channels; j++) {
10334 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
10337 out += info.outJump;
10341 else if (info.outFormat == RTAUDIO_SINT24) {
10342 Int24 *out = (Int24 *)outBuffer;
10343 if (info.inFormat == RTAUDIO_SINT8) {
10344 signed char *in = (signed char *)inBuffer;
10345 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10346 for (j=0; j<info.channels; j++) {
10347 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] << 16);
10348 //out[info.outOffset[j]] <<= 16;
10351 out += info.outJump;
10354 else if (info.inFormat == RTAUDIO_SINT16) {
10355 Int16 *in = (Int16 *)inBuffer;
10356 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10357 for (j=0; j<info.channels; j++) {
10358 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] << 8);
10359 //out[info.outOffset[j]] <<= 8;
10362 out += info.outJump;
10365 else if (info.inFormat == RTAUDIO_SINT24) {
10366 // Channel compensation and/or (de)interleaving only.
10367 Int24 *in = (Int24 *)inBuffer;
10368 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10369 for (j=0; j<info.channels; j++) {
10370 out[info.outOffset[j]] = in[info.inOffset[j]];
10373 out += info.outJump;
10376 else if (info.inFormat == RTAUDIO_SINT32) {
10377 Int32 *in = (Int32 *)inBuffer;
10378 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10379 for (j=0; j<info.channels; j++) {
10380 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] >> 8);
10381 //out[info.outOffset[j]] >>= 8;
10384 out += info.outJump;
10387 else if (info.inFormat == RTAUDIO_FLOAT32) {
10388 Float32 *in = (Float32 *)inBuffer;
10389 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10390 for (j=0; j<info.channels; j++) {
10391 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
10394 out += info.outJump;
10397 else if (info.inFormat == RTAUDIO_FLOAT64) {
10398 Float64 *in = (Float64 *)inBuffer;
10399 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10400 for (j=0; j<info.channels; j++) {
10401 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
10404 out += info.outJump;
10408 else if (info.outFormat == RTAUDIO_SINT16) {
10409 Int16 *out = (Int16 *)outBuffer;
10410 if (info.inFormat == RTAUDIO_SINT8) {
10411 signed char *in = (signed char *)inBuffer;
10412 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10413 for (j=0; j<info.channels; j++) {
10414 out[info.outOffset[j]] = (Int16) in[info.inOffset[j]];
10415 out[info.outOffset[j]] <<= 8;
10418 out += info.outJump;
10421 else if (info.inFormat == RTAUDIO_SINT16) {
10422 // Channel compensation and/or (de)interleaving only.
10423 Int16 *in = (Int16 *)inBuffer;
10424 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10425 for (j=0; j<info.channels; j++) {
10426 out[info.outOffset[j]] = in[info.inOffset[j]];
10429 out += info.outJump;
10432 else if (info.inFormat == RTAUDIO_SINT24) {
10433 Int24 *in = (Int24 *)inBuffer;
10434 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10435 for (j=0; j<info.channels; j++) {
10436 out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]].asInt() >> 8);
10439 out += info.outJump;
10442 else if (info.inFormat == RTAUDIO_SINT32) {
10443 Int32 *in = (Int32 *)inBuffer;
10444 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10445 for (j=0; j<info.channels; j++) {
10446 out[info.outOffset[j]] = (Int16) ((in[info.inOffset[j]] >> 16) & 0x0000ffff);
10449 out += info.outJump;
10452 else if (info.inFormat == RTAUDIO_FLOAT32) {
10453 Float32 *in = (Float32 *)inBuffer;
10454 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10455 for (j=0; j<info.channels; j++) {
10456 out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
10459 out += info.outJump;
10462 else if (info.inFormat == RTAUDIO_FLOAT64) {
10463 Float64 *in = (Float64 *)inBuffer;
10464 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10465 for (j=0; j<info.channels; j++) {
10466 out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
10469 out += info.outJump;
10473 else if (info.outFormat == RTAUDIO_SINT8) {
10474 signed char *out = (signed char *)outBuffer;
10475 if (info.inFormat == RTAUDIO_SINT8) {
10476 // Channel compensation and/or (de)interleaving only.
10477 signed char *in = (signed char *)inBuffer;
10478 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10479 for (j=0; j<info.channels; j++) {
10480 out[info.outOffset[j]] = in[info.inOffset[j]];
10483 out += info.outJump;
10486 if (info.inFormat == RTAUDIO_SINT16) {
10487 Int16 *in = (Int16 *)inBuffer;
10488 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10489 for (j=0; j<info.channels; j++) {
10490 out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 8) & 0x00ff);
10493 out += info.outJump;
10496 else if (info.inFormat == RTAUDIO_SINT24) {
10497 Int24 *in = (Int24 *)inBuffer;
10498 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10499 for (j=0; j<info.channels; j++) {
10500 out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]].asInt() >> 16);
10503 out += info.outJump;
10506 else if (info.inFormat == RTAUDIO_SINT32) {
10507 Int32 *in = (Int32 *)inBuffer;
10508 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10509 for (j=0; j<info.channels; j++) {
10510 out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 24) & 0x000000ff);
10513 out += info.outJump;
10516 else if (info.inFormat == RTAUDIO_FLOAT32) {
10517 Float32 *in = (Float32 *)inBuffer;
10518 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10519 for (j=0; j<info.channels; j++) {
10520 out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
10523 out += info.outJump;
10526 else if (info.inFormat == RTAUDIO_FLOAT64) {
10527 Float64 *in = (Float64 *)inBuffer;
10528 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10529 for (j=0; j<info.channels; j++) {
10530 out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
10533 out += info.outJump;
10539 //static inline uint16_t bswap_16(uint16_t x) { return (x>>8) | (x<<8); }
10540 //static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); }
10541 //static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); }
10543 void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format )
10549 if ( format == RTAUDIO_SINT16 ) {
10550 for ( unsigned int i=0; i<samples; i++ ) {
10551 // Swap 1st and 2nd bytes.
10556 // Increment 2 bytes.
10560 else if ( format == RTAUDIO_SINT32 ||
10561 format == RTAUDIO_FLOAT32 ) {
10562 for ( unsigned int i=0; i<samples; i++ ) {
10563 // Swap 1st and 4th bytes.
10568 // Swap 2nd and 3rd bytes.
10574 // Increment 3 more bytes.
10578 else if ( format == RTAUDIO_SINT24 ) {
10579 for ( unsigned int i=0; i<samples; i++ ) {
10580 // Swap 1st and 3rd bytes.
10585 // Increment 2 more bytes.
10589 else if ( format == RTAUDIO_FLOAT64 ) {
10590 for ( unsigned int i=0; i<samples; i++ ) {
10591 // Swap 1st and 8th bytes
10596 // Swap 2nd and 7th bytes
10602 // Swap 3rd and 6th bytes
10608 // Swap 4th and 5th bytes
10614 // Increment 5 more bytes.
10620 // Indentation settings for Vim and Emacs
10622 // Local Variables:
10623 // c-basic-offset: 2
10624 // indent-tabs-mode: nil
10627 // vim: et sts=2 sw=2