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 )
4810 hr = renderDevices->Item( device, &devicePtr );
4811 if ( FAILED( hr ) ) {
4812 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device handle.";
4816 hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
4817 NULL, ( void** ) &renderAudioClient );
4818 if ( FAILED( hr ) ) {
4819 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device audio client.";
4823 hr = renderAudioClient->GetMixFormat( &deviceFormat );
4824 if ( FAILED( hr ) ) {
4825 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device mix format.";
4829 stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
4830 renderAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
4834 if ( ( stream_.mode == OUTPUT && mode == INPUT ) ||
4835 ( stream_.mode == INPUT && mode == OUTPUT ) ) {
4836 stream_.mode = DUPLEX;
4839 stream_.mode = mode;
4842 stream_.device[mode] = device;
4843 stream_.doByteSwap[mode] = false;
4844 stream_.sampleRate = sampleRate;
4845 stream_.bufferSize = *bufferSize;
4846 stream_.nBuffers = 1;
4847 stream_.nUserChannels[mode] = channels;
4848 stream_.channelOffset[mode] = firstChannel;
4849 stream_.userFormat = format;
4850 stream_.deviceFormat[mode] = getDeviceInfo( device ).nativeFormats;
4852 if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
4853 stream_.userInterleaved = false;
4855 stream_.userInterleaved = true;
4856 stream_.deviceInterleaved[mode] = true;
4858 // Set flags for buffer conversion.
4859 stream_.doConvertBuffer[mode] = false;
4860 if ( stream_.userFormat != stream_.deviceFormat[mode] ||
4861 stream_.nUserChannels[0] != stream_.nDeviceChannels[0] ||
4862 stream_.nUserChannels[1] != stream_.nDeviceChannels[1] )
4863 stream_.doConvertBuffer[mode] = true;
4864 else if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
4865 stream_.nUserChannels[mode] > 1 )
4866 stream_.doConvertBuffer[mode] = true;
4868 if ( stream_.doConvertBuffer[mode] )
4869 setConvertInfo( mode, 0 );
4871 // Allocate necessary internal buffers
4872 bufferBytes = stream_.nUserChannels[mode] * stream_.bufferSize * formatBytes( stream_.userFormat );
4874 stream_.userBuffer[mode] = ( char* ) calloc( bufferBytes, 1 );
4875 if ( !stream_.userBuffer[mode] ) {
4876 errorType = RtAudioError::MEMORY_ERROR;
4877 errorText_ = "RtApiWasapi::probeDeviceOpen: Error allocating user buffer memory.";
4881 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME )
4882 stream_.callbackInfo.priority = 15;
4884 stream_.callbackInfo.priority = 0;
4886 ///! TODO: RTAUDIO_MINIMIZE_LATENCY // Provide stream buffers directly to callback
4887 ///! TODO: RTAUDIO_HOG_DEVICE // Exclusive mode
4889 methodResult = SUCCESS;
4893 SAFE_RELEASE( captureDevices );
4894 SAFE_RELEASE( renderDevices );
4895 SAFE_RELEASE( devicePtr );
4896 CoTaskMemFree( deviceFormat );
4898 // if method failed, close the stream
4899 if ( methodResult == FAILURE )
4902 if ( !errorText_.empty() )
4904 return methodResult;
4907 //=============================================================================
4909 DWORD WINAPI RtApiWasapi::runWasapiThread( void* wasapiPtr )
4912 ( ( RtApiWasapi* ) wasapiPtr )->wasapiThread();
4917 DWORD WINAPI RtApiWasapi::stopWasapiThread( void* wasapiPtr )
4920 ( ( RtApiWasapi* ) wasapiPtr )->stopStream();
4925 DWORD WINAPI RtApiWasapi::abortWasapiThread( void* wasapiPtr )
4928 ( ( RtApiWasapi* ) wasapiPtr )->abortStream();
4933 //-----------------------------------------------------------------------------
4935 void RtApiWasapi::wasapiThread()
4937 // as this is a new thread, we must CoInitialize it
4938 CoInitialize( NULL );
4942 IAudioClient* captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
4943 IAudioClient* renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
4944 IAudioCaptureClient* captureClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureClient;
4945 IAudioRenderClient* renderClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderClient;
4946 HANDLE captureEvent = ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent;
4947 HANDLE renderEvent = ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent;
4949 WAVEFORMATEX* captureFormat = NULL;
4950 WAVEFORMATEX* renderFormat = NULL;
4951 float captureSrRatio = 0.0f;
4952 float renderSrRatio = 0.0f;
4953 WasapiBuffer captureBuffer;
4954 WasapiBuffer renderBuffer;
4955 WasapiResampler* captureResampler = NULL;
4956 WasapiResampler* renderResampler = NULL;
4958 // declare local stream variables
4959 RtAudioCallback callback = ( RtAudioCallback ) stream_.callbackInfo.callback;
4960 BYTE* streamBuffer = NULL;
4961 unsigned long captureFlags = 0;
4962 unsigned int bufferFrameCount = 0;
4963 unsigned int numFramesPadding = 0;
4964 unsigned int convBufferSize = 0;
4965 bool loopbackEnabled = stream_.device[INPUT] == stream_.device[OUTPUT];
4966 bool callbackPushed = true;
4967 bool callbackPulled = false;
4968 bool callbackStopped = false;
4969 int callbackResult = 0;
4971 // convBuffer is used to store converted buffers between WASAPI and the user
4972 char* convBuffer = NULL;
4973 unsigned int convBuffSize = 0;
4974 unsigned int deviceBuffSize = 0;
4977 RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
4979 // Attempt to assign "Pro Audio" characteristic to thread
4980 HMODULE AvrtDll = LoadLibrary( (LPCTSTR) "AVRT.dll" );
4982 DWORD taskIndex = 0;
4983 TAvSetMmThreadCharacteristicsPtr AvSetMmThreadCharacteristicsPtr = ( TAvSetMmThreadCharacteristicsPtr ) GetProcAddress( AvrtDll, "AvSetMmThreadCharacteristicsW" );
4984 AvSetMmThreadCharacteristicsPtr( L"Pro Audio", &taskIndex );
4985 FreeLibrary( AvrtDll );
4988 // start capture stream if applicable
4989 if ( captureAudioClient ) {
4990 hr = captureAudioClient->GetMixFormat( &captureFormat );
4991 if ( FAILED( hr ) ) {
4992 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
4996 // init captureResampler
4997 captureResampler = new WasapiResampler( stream_.deviceFormat[INPUT] == RTAUDIO_FLOAT32 || stream_.deviceFormat[INPUT] == RTAUDIO_FLOAT64,
4998 formatBytes( stream_.deviceFormat[INPUT] ) * 8, stream_.nDeviceChannels[INPUT],
4999 captureFormat->nSamplesPerSec, stream_.sampleRate );
5001 captureSrRatio = ( ( float ) captureFormat->nSamplesPerSec / stream_.sampleRate );
5003 if ( !captureClient ) {
5004 hr = captureAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
5005 loopbackEnabled ? AUDCLNT_STREAMFLAGS_LOOPBACK : AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
5010 if ( FAILED( hr ) ) {
5011 errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize capture audio client.";
5015 hr = captureAudioClient->GetService( __uuidof( IAudioCaptureClient ),
5016 ( void** ) &captureClient );
5017 if ( FAILED( hr ) ) {
5018 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture client handle.";
5022 // don't configure captureEvent if in loopback mode
5023 if ( !loopbackEnabled )
5025 // configure captureEvent to trigger on every available capture buffer
5026 captureEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
5027 if ( !captureEvent ) {
5028 errorType = RtAudioError::SYSTEM_ERROR;
5029 errorText_ = "RtApiWasapi::wasapiThread: Unable to create capture event.";
5033 hr = captureAudioClient->SetEventHandle( captureEvent );
5034 if ( FAILED( hr ) ) {
5035 errorText_ = "RtApiWasapi::wasapiThread: Unable to set capture event handle.";
5039 ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent = captureEvent;
5042 ( ( WasapiHandle* ) stream_.apiHandle )->captureClient = captureClient;
5045 unsigned int inBufferSize = 0;
5046 hr = captureAudioClient->GetBufferSize( &inBufferSize );
5047 if ( FAILED( hr ) ) {
5048 errorText_ = "RtApiWasapi::wasapiThread: Unable to get capture buffer size.";
5052 // scale outBufferSize according to stream->user sample rate ratio
5053 unsigned int outBufferSize = ( unsigned int ) ceilf( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT];
5054 inBufferSize *= stream_.nDeviceChannels[INPUT];
5056 // set captureBuffer size
5057 captureBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[INPUT] ) );
5059 // reset the capture stream
5060 hr = captureAudioClient->Reset();
5061 if ( FAILED( hr ) ) {
5062 errorText_ = "RtApiWasapi::wasapiThread: Unable to reset capture stream.";
5066 // start the capture stream
5067 hr = captureAudioClient->Start();
5068 if ( FAILED( hr ) ) {
5069 errorText_ = "RtApiWasapi::wasapiThread: Unable to start capture stream.";
5074 // start render stream if applicable
5075 if ( renderAudioClient ) {
5076 hr = renderAudioClient->GetMixFormat( &renderFormat );
5077 if ( FAILED( hr ) ) {
5078 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
5082 // init renderResampler
5083 renderResampler = new WasapiResampler( stream_.deviceFormat[OUTPUT] == RTAUDIO_FLOAT32 || stream_.deviceFormat[OUTPUT] == RTAUDIO_FLOAT64,
5084 formatBytes( stream_.deviceFormat[OUTPUT] ) * 8, stream_.nDeviceChannels[OUTPUT],
5085 stream_.sampleRate, renderFormat->nSamplesPerSec );
5087 renderSrRatio = ( ( float ) renderFormat->nSamplesPerSec / stream_.sampleRate );
5089 if ( !renderClient ) {
5090 hr = renderAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
5091 AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
5096 if ( FAILED( hr ) ) {
5097 errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize render audio client.";
5101 hr = renderAudioClient->GetService( __uuidof( IAudioRenderClient ),
5102 ( void** ) &renderClient );
5103 if ( FAILED( hr ) ) {
5104 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render client handle.";
5108 // configure renderEvent to trigger on every available render buffer
5109 renderEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
5110 if ( !renderEvent ) {
5111 errorType = RtAudioError::SYSTEM_ERROR;
5112 errorText_ = "RtApiWasapi::wasapiThread: Unable to create render event.";
5116 hr = renderAudioClient->SetEventHandle( renderEvent );
5117 if ( FAILED( hr ) ) {
5118 errorText_ = "RtApiWasapi::wasapiThread: Unable to set render event handle.";
5122 ( ( WasapiHandle* ) stream_.apiHandle )->renderClient = renderClient;
5123 ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent = renderEvent;
5126 unsigned int outBufferSize = 0;
5127 hr = renderAudioClient->GetBufferSize( &outBufferSize );
5128 if ( FAILED( hr ) ) {
5129 errorText_ = "RtApiWasapi::wasapiThread: Unable to get render buffer size.";
5133 // scale inBufferSize according to user->stream sample rate ratio
5134 unsigned int inBufferSize = ( unsigned int ) ceilf( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT];
5135 outBufferSize *= stream_.nDeviceChannels[OUTPUT];
5137 // set renderBuffer size
5138 renderBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[OUTPUT] ) );
5140 // reset the render stream
5141 hr = renderAudioClient->Reset();
5142 if ( FAILED( hr ) ) {
5143 errorText_ = "RtApiWasapi::wasapiThread: Unable to reset render stream.";
5147 // start the render stream
5148 hr = renderAudioClient->Start();
5149 if ( FAILED( hr ) ) {
5150 errorText_ = "RtApiWasapi::wasapiThread: Unable to start render stream.";
5155 // malloc buffer memory
5156 if ( stream_.mode == INPUT )
5158 using namespace std; // for ceilf
5159 convBuffSize = ( size_t ) ( ceilf( stream_.bufferSize * captureSrRatio ) ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
5160 deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
5162 else if ( stream_.mode == OUTPUT )
5164 convBuffSize = ( size_t ) ( ceilf( stream_.bufferSize * renderSrRatio ) ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
5165 deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
5167 else if ( stream_.mode == DUPLEX )
5169 convBuffSize = std::max( ( size_t ) ( ceilf( stream_.bufferSize * captureSrRatio ) ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
5170 ( size_t ) ( ceilf( stream_.bufferSize * renderSrRatio ) ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
5171 deviceBuffSize = std::max( stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
5172 stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
5175 convBuffSize *= 2; // allow overflow for *SrRatio remainders
5176 convBuffer = ( char* ) malloc( convBuffSize );
5177 stream_.deviceBuffer = ( char* ) malloc( deviceBuffSize );
5178 if ( !convBuffer || !stream_.deviceBuffer ) {
5179 errorType = RtAudioError::MEMORY_ERROR;
5180 errorText_ = "RtApiWasapi::wasapiThread: Error allocating device buffer memory.";
5184 // stream process loop
5185 while ( stream_.state != STREAM_STOPPING ) {
5186 if ( !callbackPulled ) {
5189 // 1. Pull callback buffer from inputBuffer
5190 // 2. If 1. was successful: Convert callback buffer to user sample rate and channel count
5191 // Convert callback buffer to user format
5193 if ( captureAudioClient )
5195 int samplesToPull = ( unsigned int ) floorf( stream_.bufferSize * captureSrRatio );
5196 if ( captureSrRatio != 1 )
5198 // account for remainders
5203 while ( convBufferSize < stream_.bufferSize )
5205 // Pull callback buffer from inputBuffer
5206 callbackPulled = captureBuffer.pullBuffer( convBuffer,
5207 samplesToPull * stream_.nDeviceChannels[INPUT],
5208 stream_.deviceFormat[INPUT] );
5210 if ( !callbackPulled )
5215 // Convert callback buffer to user sample rate
5216 unsigned int deviceBufferOffset = convBufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
5217 unsigned int convSamples = 0;
5219 captureResampler->Convert( stream_.deviceBuffer + deviceBufferOffset,
5224 convBufferSize += convSamples;
5225 samplesToPull = 1; // now pull one sample at a time until we have stream_.bufferSize samples
5228 if ( callbackPulled )
5230 if ( stream_.doConvertBuffer[INPUT] ) {
5231 // Convert callback buffer to user format
5232 convertBuffer( stream_.userBuffer[INPUT],
5233 stream_.deviceBuffer,
5234 stream_.convertInfo[INPUT] );
5237 // no further conversion, simple copy deviceBuffer to userBuffer
5238 memcpy( stream_.userBuffer[INPUT],
5239 stream_.deviceBuffer,
5240 stream_.bufferSize * stream_.nUserChannels[INPUT] * formatBytes( stream_.userFormat ) );
5245 // if there is no capture stream, set callbackPulled flag
5246 callbackPulled = true;
5251 // 1. Execute user callback method
5252 // 2. Handle return value from callback
5254 // if callback has not requested the stream to stop
5255 if ( callbackPulled && !callbackStopped ) {
5256 // Execute user callback method
5257 callbackResult = callback( stream_.userBuffer[OUTPUT],
5258 stream_.userBuffer[INPUT],
5261 captureFlags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY ? RTAUDIO_INPUT_OVERFLOW : 0,
5262 stream_.callbackInfo.userData );
5264 // Handle return value from callback
5265 if ( callbackResult == 1 ) {
5266 // instantiate a thread to stop this thread
5267 HANDLE threadHandle = CreateThread( NULL, 0, stopWasapiThread, this, 0, NULL );
5268 if ( !threadHandle ) {
5269 errorType = RtAudioError::THREAD_ERROR;
5270 errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream stop thread.";
5273 else if ( !CloseHandle( threadHandle ) ) {
5274 errorType = RtAudioError::THREAD_ERROR;
5275 errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream stop thread handle.";
5279 callbackStopped = true;
5281 else if ( callbackResult == 2 ) {
5282 // instantiate a thread to stop this thread
5283 HANDLE threadHandle = CreateThread( NULL, 0, abortWasapiThread, this, 0, NULL );
5284 if ( !threadHandle ) {
5285 errorType = RtAudioError::THREAD_ERROR;
5286 errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream abort thread.";
5289 else if ( !CloseHandle( threadHandle ) ) {
5290 errorType = RtAudioError::THREAD_ERROR;
5291 errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream abort thread handle.";
5295 callbackStopped = true;
5302 // 1. Convert callback buffer to stream format
5303 // 2. Convert callback buffer to stream sample rate and channel count
5304 // 3. Push callback buffer into outputBuffer
5306 if ( renderAudioClient && callbackPulled )
5308 // if the last call to renderBuffer.PushBuffer() was successful
5309 if ( callbackPushed || convBufferSize == 0 )
5311 if ( stream_.doConvertBuffer[OUTPUT] )
5313 // Convert callback buffer to stream format
5314 convertBuffer( stream_.deviceBuffer,
5315 stream_.userBuffer[OUTPUT],
5316 stream_.convertInfo[OUTPUT] );
5320 // Convert callback buffer to stream sample rate
5321 renderResampler->Convert( convBuffer,
5322 stream_.deviceBuffer,
5327 // Push callback buffer into outputBuffer
5328 callbackPushed = renderBuffer.pushBuffer( convBuffer,
5329 convBufferSize * stream_.nDeviceChannels[OUTPUT],
5330 stream_.deviceFormat[OUTPUT] );
5333 // if there is no render stream, set callbackPushed flag
5334 callbackPushed = true;
5339 // 1. Get capture buffer from stream
5340 // 2. Push capture buffer into inputBuffer
5341 // 3. If 2. was successful: Release capture buffer
5343 if ( captureAudioClient ) {
5344 // if the callback input buffer was not pulled from captureBuffer, wait for next capture event
5345 if ( !callbackPulled ) {
5346 WaitForSingleObject( loopbackEnabled ? renderEvent : captureEvent, INFINITE );
5349 // Get capture buffer from stream
5350 hr = captureClient->GetBuffer( &streamBuffer,
5352 &captureFlags, NULL, NULL );
5353 if ( FAILED( hr ) ) {
5354 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture buffer.";
5358 if ( bufferFrameCount != 0 ) {
5359 // Push capture buffer into inputBuffer
5360 if ( captureBuffer.pushBuffer( ( char* ) streamBuffer,
5361 bufferFrameCount * stream_.nDeviceChannels[INPUT],
5362 stream_.deviceFormat[INPUT] ) )
5364 // Release capture buffer
5365 hr = captureClient->ReleaseBuffer( bufferFrameCount );
5366 if ( FAILED( hr ) ) {
5367 errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
5373 // Inform WASAPI that capture was unsuccessful
5374 hr = captureClient->ReleaseBuffer( 0 );
5375 if ( FAILED( hr ) ) {
5376 errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
5383 // Inform WASAPI that capture was unsuccessful
5384 hr = captureClient->ReleaseBuffer( 0 );
5385 if ( FAILED( hr ) ) {
5386 errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
5394 // 1. Get render buffer from stream
5395 // 2. Pull next buffer from outputBuffer
5396 // 3. If 2. was successful: Fill render buffer with next buffer
5397 // Release render buffer
5399 if ( renderAudioClient ) {
5400 // if the callback output buffer was not pushed to renderBuffer, wait for next render event
5401 if ( callbackPulled && !callbackPushed ) {
5402 WaitForSingleObject( renderEvent, INFINITE );
5405 // Get render buffer from stream
5406 hr = renderAudioClient->GetBufferSize( &bufferFrameCount );
5407 if ( FAILED( hr ) ) {
5408 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer size.";
5412 hr = renderAudioClient->GetCurrentPadding( &numFramesPadding );
5413 if ( FAILED( hr ) ) {
5414 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer padding.";
5418 bufferFrameCount -= numFramesPadding;
5420 if ( bufferFrameCount != 0 ) {
5421 hr = renderClient->GetBuffer( bufferFrameCount, &streamBuffer );
5422 if ( FAILED( hr ) ) {
5423 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer.";
5427 // Pull next buffer from outputBuffer
5428 // Fill render buffer with next buffer
5429 if ( renderBuffer.pullBuffer( ( char* ) streamBuffer,
5430 bufferFrameCount * stream_.nDeviceChannels[OUTPUT],
5431 stream_.deviceFormat[OUTPUT] ) )
5433 // Release render buffer
5434 hr = renderClient->ReleaseBuffer( bufferFrameCount, 0 );
5435 if ( FAILED( hr ) ) {
5436 errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
5442 // Inform WASAPI that render was unsuccessful
5443 hr = renderClient->ReleaseBuffer( 0, 0 );
5444 if ( FAILED( hr ) ) {
5445 errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
5452 // Inform WASAPI that render was unsuccessful
5453 hr = renderClient->ReleaseBuffer( 0, 0 );
5454 if ( FAILED( hr ) ) {
5455 errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
5461 // if the callback buffer was pushed renderBuffer reset callbackPulled flag
5462 if ( callbackPushed ) {
5463 // unsetting the callbackPulled flag lets the stream know that
5464 // the audio device is ready for another callback output buffer.
5465 callbackPulled = false;
5468 RtApi::tickStreamTime();
5475 CoTaskMemFree( captureFormat );
5476 CoTaskMemFree( renderFormat );
5478 free ( convBuffer );
5479 delete renderResampler;
5480 delete captureResampler;
5484 if ( !errorText_.empty() )
5487 // update stream state
5488 stream_.state = STREAM_STOPPED;
5491 //******************** End of __WINDOWS_WASAPI__ *********************//
5495 #if defined(__WINDOWS_DS__) // Windows DirectSound API
5497 // Modified by Robin Davies, October 2005
5498 // - Improvements to DirectX pointer chasing.
5499 // - Bug fix for non-power-of-two Asio granularity used by Edirol PCR-A30.
5500 // - Auto-call CoInitialize for DSOUND and ASIO platforms.
5501 // Various revisions for RtAudio 4.0 by Gary Scavone, April 2007
5502 // Changed device query structure for RtAudio 4.0.7, January 2010
5504 #include <windows.h>
5505 #include <process.h>
5506 #include <mmsystem.h>
5510 #include <algorithm>
5512 #if defined(__MINGW32__)
5513 // missing from latest mingw winapi
5514 #define WAVE_FORMAT_96M08 0x00010000 /* 96 kHz, Mono, 8-bit */
5515 #define WAVE_FORMAT_96S08 0x00020000 /* 96 kHz, Stereo, 8-bit */
5516 #define WAVE_FORMAT_96M16 0x00040000 /* 96 kHz, Mono, 16-bit */
5517 #define WAVE_FORMAT_96S16 0x00080000 /* 96 kHz, Stereo, 16-bit */
5520 #define MINIMUM_DEVICE_BUFFER_SIZE 32768
5522 #ifdef _MSC_VER // if Microsoft Visual C++
5523 #pragma comment( lib, "winmm.lib" ) // then, auto-link winmm.lib. Otherwise, it has to be added manually.
5526 static inline DWORD dsPointerBetween( DWORD pointer, DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize )
5528 if ( pointer > bufferSize ) pointer -= bufferSize;
5529 if ( laterPointer < earlierPointer ) laterPointer += bufferSize;
5530 if ( pointer < earlierPointer ) pointer += bufferSize;
5531 return pointer >= earlierPointer && pointer < laterPointer;
5534 // A structure to hold various information related to the DirectSound
5535 // API implementation.
5537 unsigned int drainCounter; // Tracks callback counts when draining
5538 bool internalDrain; // Indicates if stop is initiated from callback or not.
5542 UINT bufferPointer[2];
5543 DWORD dsBufferSize[2];
5544 DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by.
5548 :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; }
5551 // Declarations for utility functions, callbacks, and structures
5552 // specific to the DirectSound implementation.
5553 static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
5554 LPCTSTR description,
5558 static const char* getErrorString( int code );
5560 static unsigned __stdcall callbackHandler( void *ptr );
5569 : found(false) { validId[0] = false; validId[1] = false; }
5572 struct DsProbeData {
5574 std::vector<struct DsDevice>* dsDevices;
5577 RtApiDs :: RtApiDs()
5579 // Dsound will run both-threaded. If CoInitialize fails, then just
5580 // accept whatever the mainline chose for a threading model.
5581 coInitialized_ = false;
5582 HRESULT hr = CoInitialize( NULL );
5583 if ( !FAILED( hr ) ) coInitialized_ = true;
5586 RtApiDs :: ~RtApiDs()
5588 if ( stream_.state != STREAM_CLOSED ) closeStream();
5589 if ( coInitialized_ ) CoUninitialize(); // balanced call.
5592 // The DirectSound default output is always the first device.
5593 unsigned int RtApiDs :: getDefaultOutputDevice( void )
5598 // The DirectSound default input is always the first input device,
5599 // which is the first capture device enumerated.
5600 unsigned int RtApiDs :: getDefaultInputDevice( void )
5605 unsigned int RtApiDs :: getDeviceCount( void )
5607 // Set query flag for previously found devices to false, so that we
5608 // can check for any devices that have disappeared.
5609 for ( unsigned int i=0; i<dsDevices.size(); i++ )
5610 dsDevices[i].found = false;
5612 // Query DirectSound devices.
5613 struct DsProbeData probeInfo;
5614 probeInfo.isInput = false;
5615 probeInfo.dsDevices = &dsDevices;
5616 HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );
5617 if ( FAILED( result ) ) {
5618 errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!";
5619 errorText_ = errorStream_.str();
5620 error( RtAudioError::WARNING );
5623 // Query DirectSoundCapture devices.
5624 probeInfo.isInput = true;
5625 result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );
5626 if ( FAILED( result ) ) {
5627 errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!";
5628 errorText_ = errorStream_.str();
5629 error( RtAudioError::WARNING );
5632 // Clean out any devices that may have disappeared (code update submitted by Eli Zehngut).
5633 for ( unsigned int i=0; i<dsDevices.size(); ) {
5634 if ( dsDevices[i].found == false ) dsDevices.erase( dsDevices.begin() + i );
5638 return static_cast<unsigned int>(dsDevices.size());
5641 RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device )
5643 RtAudio::DeviceInfo info;
5644 info.probed = false;
5646 if ( dsDevices.size() == 0 ) {
5647 // Force a query of all devices
5649 if ( dsDevices.size() == 0 ) {
5650 errorText_ = "RtApiDs::getDeviceInfo: no devices found!";
5651 error( RtAudioError::INVALID_USE );
5656 if ( device >= dsDevices.size() ) {
5657 errorText_ = "RtApiDs::getDeviceInfo: device ID is invalid!";
5658 error( RtAudioError::INVALID_USE );
5663 if ( dsDevices[ device ].validId[0] == false ) goto probeInput;
5665 LPDIRECTSOUND output;
5667 result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );
5668 if ( FAILED( result ) ) {
5669 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!";
5670 errorText_ = errorStream_.str();
5671 error( RtAudioError::WARNING );
5675 outCaps.dwSize = sizeof( outCaps );
5676 result = output->GetCaps( &outCaps );
5677 if ( FAILED( result ) ) {
5679 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting capabilities!";
5680 errorText_ = errorStream_.str();
5681 error( RtAudioError::WARNING );
5685 // Get output channel information.
5686 info.outputChannels = ( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1;
5688 // Get sample rate information.
5689 info.sampleRates.clear();
5690 for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
5691 if ( SAMPLE_RATES[k] >= (unsigned int) outCaps.dwMinSecondarySampleRate &&
5692 SAMPLE_RATES[k] <= (unsigned int) outCaps.dwMaxSecondarySampleRate ) {
5693 info.sampleRates.push_back( SAMPLE_RATES[k] );
5695 if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
5696 info.preferredSampleRate = SAMPLE_RATES[k];
5700 // Get format information.
5701 if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT ) info.nativeFormats |= RTAUDIO_SINT16;
5702 if ( outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) info.nativeFormats |= RTAUDIO_SINT8;
5706 if ( getDefaultOutputDevice() == device )
5707 info.isDefaultOutput = true;
5709 if ( dsDevices[ device ].validId[1] == false ) {
5710 info.name = dsDevices[ device ].name;
5717 LPDIRECTSOUNDCAPTURE input;
5718 result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );
5719 if ( FAILED( result ) ) {
5720 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";
5721 errorText_ = errorStream_.str();
5722 error( RtAudioError::WARNING );
5727 inCaps.dwSize = sizeof( inCaps );
5728 result = input->GetCaps( &inCaps );
5729 if ( FAILED( result ) ) {
5731 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsDevices[ device ].name << ")!";
5732 errorText_ = errorStream_.str();
5733 error( RtAudioError::WARNING );
5737 // Get input channel information.
5738 info.inputChannels = inCaps.dwChannels;
5740 // Get sample rate and format information.
5741 std::vector<unsigned int> rates;
5742 if ( inCaps.dwChannels >= 2 ) {
5743 if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.nativeFormats |= RTAUDIO_SINT16;
5744 if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.nativeFormats |= RTAUDIO_SINT16;
5745 if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.nativeFormats |= RTAUDIO_SINT16;
5746 if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.nativeFormats |= RTAUDIO_SINT16;
5747 if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.nativeFormats |= RTAUDIO_SINT8;
5748 if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.nativeFormats |= RTAUDIO_SINT8;
5749 if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.nativeFormats |= RTAUDIO_SINT8;
5750 if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.nativeFormats |= RTAUDIO_SINT8;
5752 if ( info.nativeFormats & RTAUDIO_SINT16 ) {
5753 if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) rates.push_back( 11025 );
5754 if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) rates.push_back( 22050 );
5755 if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) rates.push_back( 44100 );
5756 if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) rates.push_back( 96000 );
5758 else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
5759 if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) rates.push_back( 11025 );
5760 if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) rates.push_back( 22050 );
5761 if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) rates.push_back( 44100 );
5762 if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) rates.push_back( 96000 );
5765 else if ( inCaps.dwChannels == 1 ) {
5766 if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.nativeFormats |= RTAUDIO_SINT16;
5767 if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.nativeFormats |= RTAUDIO_SINT16;
5768 if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.nativeFormats |= RTAUDIO_SINT16;
5769 if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.nativeFormats |= RTAUDIO_SINT16;
5770 if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.nativeFormats |= RTAUDIO_SINT8;
5771 if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.nativeFormats |= RTAUDIO_SINT8;
5772 if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.nativeFormats |= RTAUDIO_SINT8;
5773 if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.nativeFormats |= RTAUDIO_SINT8;
5775 if ( info.nativeFormats & RTAUDIO_SINT16 ) {
5776 if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) rates.push_back( 11025 );
5777 if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) rates.push_back( 22050 );
5778 if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) rates.push_back( 44100 );
5779 if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) rates.push_back( 96000 );
5781 else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
5782 if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) rates.push_back( 11025 );
5783 if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) rates.push_back( 22050 );
5784 if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) rates.push_back( 44100 );
5785 if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) rates.push_back( 96000 );
5788 else info.inputChannels = 0; // technically, this would be an error
5792 if ( info.inputChannels == 0 ) return info;
5794 // Copy the supported rates to the info structure but avoid duplication.
5796 for ( unsigned int i=0; i<rates.size(); i++ ) {
5798 for ( unsigned int j=0; j<info.sampleRates.size(); j++ ) {
5799 if ( rates[i] == info.sampleRates[j] ) {
5804 if ( found == false ) info.sampleRates.push_back( rates[i] );
5806 std::sort( info.sampleRates.begin(), info.sampleRates.end() );
5808 // If device opens for both playback and capture, we determine the channels.
5809 if ( info.outputChannels > 0 && info.inputChannels > 0 )
5810 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
5812 if ( device == 0 ) info.isDefaultInput = true;
5814 // Copy name and return.
5815 info.name = dsDevices[ device ].name;
5820 bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
5821 unsigned int firstChannel, unsigned int sampleRate,
5822 RtAudioFormat format, unsigned int *bufferSize,
5823 RtAudio::StreamOptions *options )
5825 if ( channels + firstChannel > 2 ) {
5826 errorText_ = "RtApiDs::probeDeviceOpen: DirectSound does not support more than 2 channels per device.";
5830 size_t nDevices = dsDevices.size();
5831 if ( nDevices == 0 ) {
5832 // This should not happen because a check is made before this function is called.
5833 errorText_ = "RtApiDs::probeDeviceOpen: no devices found!";
5837 if ( device >= nDevices ) {
5838 // This should not happen because a check is made before this function is called.
5839 errorText_ = "RtApiDs::probeDeviceOpen: device ID is invalid!";
5843 if ( mode == OUTPUT ) {
5844 if ( dsDevices[ device ].validId[0] == false ) {
5845 errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support output!";
5846 errorText_ = errorStream_.str();
5850 else { // mode == INPUT
5851 if ( dsDevices[ device ].validId[1] == false ) {
5852 errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support input!";
5853 errorText_ = errorStream_.str();
5858 // According to a note in PortAudio, using GetDesktopWindow()
5859 // instead of GetForegroundWindow() is supposed to avoid problems
5860 // that occur when the application's window is not the foreground
5861 // window. Also, if the application window closes before the
5862 // DirectSound buffer, DirectSound can crash. In the past, I had
5863 // problems when using GetDesktopWindow() but it seems fine now
5864 // (January 2010). I'll leave it commented here.
5865 // HWND hWnd = GetForegroundWindow();
5866 HWND hWnd = GetDesktopWindow();
5868 // Check the numberOfBuffers parameter and limit the lowest value to
5869 // two. This is a judgement call and a value of two is probably too
5870 // low for capture, but it should work for playback.
5872 if ( options ) nBuffers = options->numberOfBuffers;
5873 if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) nBuffers = 2;
5874 if ( nBuffers < 2 ) nBuffers = 3;
5876 // Check the lower range of the user-specified buffer size and set
5877 // (arbitrarily) to a lower bound of 32.
5878 if ( *bufferSize < 32 ) *bufferSize = 32;
5880 // Create the wave format structure. The data format setting will
5881 // be determined later.
5882 WAVEFORMATEX waveFormat;
5883 ZeroMemory( &waveFormat, sizeof(WAVEFORMATEX) );
5884 waveFormat.wFormatTag = WAVE_FORMAT_PCM;
5885 waveFormat.nChannels = channels + firstChannel;
5886 waveFormat.nSamplesPerSec = (unsigned long) sampleRate;
5888 // Determine the device buffer size. By default, we'll use the value
5889 // defined above (32K), but we will grow it to make allowances for
5890 // very large software buffer sizes.
5891 DWORD dsBufferSize = MINIMUM_DEVICE_BUFFER_SIZE;
5892 DWORD dsPointerLeadTime = 0;
5894 void *ohandle = 0, *bhandle = 0;
5896 if ( mode == OUTPUT ) {
5898 LPDIRECTSOUND output;
5899 result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );
5900 if ( FAILED( result ) ) {
5901 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!";
5902 errorText_ = errorStream_.str();
5907 outCaps.dwSize = sizeof( outCaps );
5908 result = output->GetCaps( &outCaps );
5909 if ( FAILED( result ) ) {
5911 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting capabilities (" << dsDevices[ device ].name << ")!";
5912 errorText_ = errorStream_.str();
5916 // Check channel information.
5917 if ( channels + firstChannel == 2 && !( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ) {
5918 errorStream_ << "RtApiDs::getDeviceInfo: the output device (" << dsDevices[ device ].name << ") does not support stereo playback.";
5919 errorText_ = errorStream_.str();
5923 // Check format information. Use 16-bit format unless not
5924 // supported or user requests 8-bit.
5925 if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT &&
5926 !( format == RTAUDIO_SINT8 && outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) ) {
5927 waveFormat.wBitsPerSample = 16;
5928 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
5931 waveFormat.wBitsPerSample = 8;
5932 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
5934 stream_.userFormat = format;
5936 // Update wave format structure and buffer information.
5937 waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
5938 waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
5939 dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
5941 // If the user wants an even bigger buffer, increase the device buffer size accordingly.
5942 while ( dsPointerLeadTime * 2U > dsBufferSize )
5945 // Set cooperative level to DSSCL_EXCLUSIVE ... sound stops when window focus changes.
5946 // result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE );
5947 // Set cooperative level to DSSCL_PRIORITY ... sound remains when window focus changes.
5948 result = output->SetCooperativeLevel( hWnd, DSSCL_PRIORITY );
5949 if ( FAILED( result ) ) {
5951 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting cooperative level (" << dsDevices[ device ].name << ")!";
5952 errorText_ = errorStream_.str();
5956 // Even though we will write to the secondary buffer, we need to
5957 // access the primary buffer to set the correct output format
5958 // (since the default is 8-bit, 22 kHz!). Setup the DS primary
5959 // buffer description.
5960 DSBUFFERDESC bufferDescription;
5961 ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
5962 bufferDescription.dwSize = sizeof( DSBUFFERDESC );
5963 bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER;
5965 // Obtain the primary buffer
5966 LPDIRECTSOUNDBUFFER buffer;
5967 result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
5968 if ( FAILED( result ) ) {
5970 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") accessing primary buffer (" << dsDevices[ device ].name << ")!";
5971 errorText_ = errorStream_.str();
5975 // Set the primary DS buffer sound format.
5976 result = buffer->SetFormat( &waveFormat );
5977 if ( FAILED( result ) ) {
5979 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting primary buffer format (" << dsDevices[ device ].name << ")!";
5980 errorText_ = errorStream_.str();
5984 // Setup the secondary DS buffer description.
5985 ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
5986 bufferDescription.dwSize = sizeof( DSBUFFERDESC );
5987 bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
5988 DSBCAPS_GLOBALFOCUS |
5989 DSBCAPS_GETCURRENTPOSITION2 |
5990 DSBCAPS_LOCHARDWARE ); // Force hardware mixing
5991 bufferDescription.dwBufferBytes = dsBufferSize;
5992 bufferDescription.lpwfxFormat = &waveFormat;
5994 // Try to create the secondary DS buffer. If that doesn't work,
5995 // try to use software mixing. Otherwise, there's a problem.
5996 result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
5997 if ( FAILED( result ) ) {
5998 bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
5999 DSBCAPS_GLOBALFOCUS |
6000 DSBCAPS_GETCURRENTPOSITION2 |
6001 DSBCAPS_LOCSOFTWARE ); // Force software mixing
6002 result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
6003 if ( FAILED( result ) ) {
6005 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating secondary buffer (" << dsDevices[ device ].name << ")!";
6006 errorText_ = errorStream_.str();
6011 // Get the buffer size ... might be different from what we specified.
6013 dsbcaps.dwSize = sizeof( DSBCAPS );
6014 result = buffer->GetCaps( &dsbcaps );
6015 if ( FAILED( result ) ) {
6018 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";
6019 errorText_ = errorStream_.str();
6023 dsBufferSize = dsbcaps.dwBufferBytes;
6025 // Lock the DS buffer
6028 result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );
6029 if ( FAILED( result ) ) {
6032 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking buffer (" << dsDevices[ device ].name << ")!";
6033 errorText_ = errorStream_.str();
6037 // Zero the DS buffer
6038 ZeroMemory( audioPtr, dataLen );
6040 // Unlock the DS buffer
6041 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
6042 if ( FAILED( result ) ) {
6045 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking buffer (" << dsDevices[ device ].name << ")!";
6046 errorText_ = errorStream_.str();
6050 ohandle = (void *) output;
6051 bhandle = (void *) buffer;
6054 if ( mode == INPUT ) {
6056 LPDIRECTSOUNDCAPTURE input;
6057 result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );
6058 if ( FAILED( result ) ) {
6059 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";
6060 errorText_ = errorStream_.str();
6065 inCaps.dwSize = sizeof( inCaps );
6066 result = input->GetCaps( &inCaps );
6067 if ( FAILED( result ) ) {
6069 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsDevices[ device ].name << ")!";
6070 errorText_ = errorStream_.str();
6074 // Check channel information.
6075 if ( inCaps.dwChannels < channels + firstChannel ) {
6076 errorText_ = "RtApiDs::getDeviceInfo: the input device does not support requested input channels.";
6080 // Check format information. Use 16-bit format unless user
6082 DWORD deviceFormats;
6083 if ( channels + firstChannel == 2 ) {
6084 deviceFormats = WAVE_FORMAT_1S08 | WAVE_FORMAT_2S08 | WAVE_FORMAT_4S08 | WAVE_FORMAT_96S08;
6085 if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
6086 waveFormat.wBitsPerSample = 8;
6087 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
6089 else { // assume 16-bit is supported
6090 waveFormat.wBitsPerSample = 16;
6091 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
6094 else { // channel == 1
6095 deviceFormats = WAVE_FORMAT_1M08 | WAVE_FORMAT_2M08 | WAVE_FORMAT_4M08 | WAVE_FORMAT_96M08;
6096 if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
6097 waveFormat.wBitsPerSample = 8;
6098 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
6100 else { // assume 16-bit is supported
6101 waveFormat.wBitsPerSample = 16;
6102 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
6105 stream_.userFormat = format;
6107 // Update wave format structure and buffer information.
6108 waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
6109 waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
6110 dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
6112 // If the user wants an even bigger buffer, increase the device buffer size accordingly.
6113 while ( dsPointerLeadTime * 2U > dsBufferSize )
6116 // Setup the secondary DS buffer description.
6117 DSCBUFFERDESC bufferDescription;
6118 ZeroMemory( &bufferDescription, sizeof( DSCBUFFERDESC ) );
6119 bufferDescription.dwSize = sizeof( DSCBUFFERDESC );
6120 bufferDescription.dwFlags = 0;
6121 bufferDescription.dwReserved = 0;
6122 bufferDescription.dwBufferBytes = dsBufferSize;
6123 bufferDescription.lpwfxFormat = &waveFormat;
6125 // Create the capture buffer.
6126 LPDIRECTSOUNDCAPTUREBUFFER buffer;
6127 result = input->CreateCaptureBuffer( &bufferDescription, &buffer, NULL );
6128 if ( FAILED( result ) ) {
6130 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating input buffer (" << dsDevices[ device ].name << ")!";
6131 errorText_ = errorStream_.str();
6135 // Get the buffer size ... might be different from what we specified.
6137 dscbcaps.dwSize = sizeof( DSCBCAPS );
6138 result = buffer->GetCaps( &dscbcaps );
6139 if ( FAILED( result ) ) {
6142 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";
6143 errorText_ = errorStream_.str();
6147 dsBufferSize = dscbcaps.dwBufferBytes;
6149 // NOTE: We could have a problem here if this is a duplex stream
6150 // and the play and capture hardware buffer sizes are different
6151 // (I'm actually not sure if that is a problem or not).
6152 // Currently, we are not verifying that.
6154 // Lock the capture buffer
6157 result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );
6158 if ( FAILED( result ) ) {
6161 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking input buffer (" << dsDevices[ device ].name << ")!";
6162 errorText_ = errorStream_.str();
6167 ZeroMemory( audioPtr, dataLen );
6169 // Unlock the buffer
6170 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
6171 if ( FAILED( result ) ) {
6174 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking input buffer (" << dsDevices[ device ].name << ")!";
6175 errorText_ = errorStream_.str();
6179 ohandle = (void *) input;
6180 bhandle = (void *) buffer;
6183 // Set various stream parameters
6184 DsHandle *handle = 0;
6185 stream_.nDeviceChannels[mode] = channels + firstChannel;
6186 stream_.nUserChannels[mode] = channels;
6187 stream_.bufferSize = *bufferSize;
6188 stream_.channelOffset[mode] = firstChannel;
6189 stream_.deviceInterleaved[mode] = true;
6190 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
6191 else stream_.userInterleaved = true;
6193 // Set flag for buffer conversion
6194 stream_.doConvertBuffer[mode] = false;
6195 if (stream_.nUserChannels[mode] != stream_.nDeviceChannels[mode])
6196 stream_.doConvertBuffer[mode] = true;
6197 if (stream_.userFormat != stream_.deviceFormat[mode])
6198 stream_.doConvertBuffer[mode] = true;
6199 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
6200 stream_.nUserChannels[mode] > 1 )
6201 stream_.doConvertBuffer[mode] = true;
6203 // Allocate necessary internal buffers
6204 long bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
6205 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
6206 if ( stream_.userBuffer[mode] == NULL ) {
6207 errorText_ = "RtApiDs::probeDeviceOpen: error allocating user buffer memory.";
6211 if ( stream_.doConvertBuffer[mode] ) {
6213 bool makeBuffer = true;
6214 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
6215 if ( mode == INPUT ) {
6216 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
6217 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
6218 if ( bufferBytes <= (long) bytesOut ) makeBuffer = false;
6223 bufferBytes *= *bufferSize;
6224 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
6225 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
6226 if ( stream_.deviceBuffer == NULL ) {
6227 errorText_ = "RtApiDs::probeDeviceOpen: error allocating device buffer memory.";
6233 // Allocate our DsHandle structures for the stream.
6234 if ( stream_.apiHandle == 0 ) {
6236 handle = new DsHandle;
6238 catch ( std::bad_alloc& ) {
6239 errorText_ = "RtApiDs::probeDeviceOpen: error allocating AsioHandle memory.";
6243 // Create a manual-reset event.
6244 handle->condition = CreateEvent( NULL, // no security
6245 TRUE, // manual-reset
6246 FALSE, // non-signaled initially
6248 stream_.apiHandle = (void *) handle;
6251 handle = (DsHandle *) stream_.apiHandle;
6252 handle->id[mode] = ohandle;
6253 handle->buffer[mode] = bhandle;
6254 handle->dsBufferSize[mode] = dsBufferSize;
6255 handle->dsPointerLeadTime[mode] = dsPointerLeadTime;
6257 stream_.device[mode] = device;
6258 stream_.state = STREAM_STOPPED;
6259 if ( stream_.mode == OUTPUT && mode == INPUT )
6260 // We had already set up an output stream.
6261 stream_.mode = DUPLEX;
6263 stream_.mode = mode;
6264 stream_.nBuffers = nBuffers;
6265 stream_.sampleRate = sampleRate;
6267 // Setup the buffer conversion information structure.
6268 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
6270 // Setup the callback thread.
6271 if ( stream_.callbackInfo.isRunning == false ) {
6273 stream_.callbackInfo.isRunning = true;
6274 stream_.callbackInfo.object = (void *) this;
6275 stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &callbackHandler,
6276 &stream_.callbackInfo, 0, &threadId );
6277 if ( stream_.callbackInfo.thread == 0 ) {
6278 errorText_ = "RtApiDs::probeDeviceOpen: error creating callback thread!";
6282 // Boost DS thread priority
6283 SetThreadPriority( (HANDLE) stream_.callbackInfo.thread, THREAD_PRIORITY_HIGHEST );
6289 if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
6290 LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
6291 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6292 if ( buffer ) buffer->Release();
6295 if ( handle->buffer[1] ) {
6296 LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
6297 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6298 if ( buffer ) buffer->Release();
6301 CloseHandle( handle->condition );
6303 stream_.apiHandle = 0;
6306 for ( int i=0; i<2; i++ ) {
6307 if ( stream_.userBuffer[i] ) {
6308 free( stream_.userBuffer[i] );
6309 stream_.userBuffer[i] = 0;
6313 if ( stream_.deviceBuffer ) {
6314 free( stream_.deviceBuffer );
6315 stream_.deviceBuffer = 0;
6318 stream_.state = STREAM_CLOSED;
6322 void RtApiDs :: closeStream()
6324 if ( stream_.state == STREAM_CLOSED ) {
6325 errorText_ = "RtApiDs::closeStream(): no open stream to close!";
6326 error( RtAudioError::WARNING );
6330 // Stop the callback thread.
6331 stream_.callbackInfo.isRunning = false;
6332 WaitForSingleObject( (HANDLE) stream_.callbackInfo.thread, INFINITE );
6333 CloseHandle( (HANDLE) stream_.callbackInfo.thread );
6335 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6337 if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
6338 LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
6339 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6346 if ( handle->buffer[1] ) {
6347 LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
6348 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6355 CloseHandle( handle->condition );
6357 stream_.apiHandle = 0;
6360 for ( int i=0; i<2; i++ ) {
6361 if ( stream_.userBuffer[i] ) {
6362 free( stream_.userBuffer[i] );
6363 stream_.userBuffer[i] = 0;
6367 if ( stream_.deviceBuffer ) {
6368 free( stream_.deviceBuffer );
6369 stream_.deviceBuffer = 0;
6372 stream_.mode = UNINITIALIZED;
6373 stream_.state = STREAM_CLOSED;
6376 void RtApiDs :: startStream()
6379 if ( stream_.state == STREAM_RUNNING ) {
6380 errorText_ = "RtApiDs::startStream(): the stream is already running!";
6381 error( RtAudioError::WARNING );
6385 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6387 // Increase scheduler frequency on lesser windows (a side-effect of
6388 // increasing timer accuracy). On greater windows (Win2K or later),
6389 // this is already in effect.
6390 timeBeginPeriod( 1 );
6392 buffersRolling = false;
6393 duplexPrerollBytes = 0;
6395 if ( stream_.mode == DUPLEX ) {
6396 // 0.5 seconds of silence in DUPLEX mode while the devices spin up and synchronize.
6397 duplexPrerollBytes = (int) ( 0.5 * stream_.sampleRate * formatBytes( stream_.deviceFormat[1] ) * stream_.nDeviceChannels[1] );
6401 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
6403 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6404 result = buffer->Play( 0, 0, DSBPLAY_LOOPING );
6405 if ( FAILED( result ) ) {
6406 errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting output buffer!";
6407 errorText_ = errorStream_.str();
6412 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
6414 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6415 result = buffer->Start( DSCBSTART_LOOPING );
6416 if ( FAILED( result ) ) {
6417 errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting input buffer!";
6418 errorText_ = errorStream_.str();
6423 handle->drainCounter = 0;
6424 handle->internalDrain = false;
6425 ResetEvent( handle->condition );
6426 stream_.state = STREAM_RUNNING;
6429 if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );
6432 void RtApiDs :: stopStream()
6435 if ( stream_.state == STREAM_STOPPED ) {
6436 errorText_ = "RtApiDs::stopStream(): the stream is already stopped!";
6437 error( RtAudioError::WARNING );
6444 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6445 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
6446 if ( handle->drainCounter == 0 ) {
6447 handle->drainCounter = 2;
6448 WaitForSingleObject( handle->condition, INFINITE ); // block until signaled
6451 stream_.state = STREAM_STOPPED;
6453 MUTEX_LOCK( &stream_.mutex );
6455 // Stop the buffer and clear memory
6456 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6457 result = buffer->Stop();
6458 if ( FAILED( result ) ) {
6459 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping output buffer!";
6460 errorText_ = errorStream_.str();
6464 // Lock the buffer and clear it so that if we start to play again,
6465 // we won't have old data playing.
6466 result = buffer->Lock( 0, handle->dsBufferSize[0], &audioPtr, &dataLen, NULL, NULL, 0 );
6467 if ( FAILED( result ) ) {
6468 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking output buffer!";
6469 errorText_ = errorStream_.str();
6473 // Zero the DS buffer
6474 ZeroMemory( audioPtr, dataLen );
6476 // Unlock the DS buffer
6477 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
6478 if ( FAILED( result ) ) {
6479 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking output buffer!";
6480 errorText_ = errorStream_.str();
6484 // If we start playing again, we must begin at beginning of buffer.
6485 handle->bufferPointer[0] = 0;
6488 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
6489 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6493 stream_.state = STREAM_STOPPED;
6495 if ( stream_.mode != DUPLEX )
6496 MUTEX_LOCK( &stream_.mutex );
6498 result = buffer->Stop();
6499 if ( FAILED( result ) ) {
6500 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping input buffer!";
6501 errorText_ = errorStream_.str();
6505 // Lock the buffer and clear it so that if we start to play again,
6506 // we won't have old data playing.
6507 result = buffer->Lock( 0, handle->dsBufferSize[1], &audioPtr, &dataLen, NULL, NULL, 0 );
6508 if ( FAILED( result ) ) {
6509 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking input buffer!";
6510 errorText_ = errorStream_.str();
6514 // Zero the DS buffer
6515 ZeroMemory( audioPtr, dataLen );
6517 // Unlock the DS buffer
6518 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
6519 if ( FAILED( result ) ) {
6520 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking input buffer!";
6521 errorText_ = errorStream_.str();
6525 // If we start recording again, we must begin at beginning of buffer.
6526 handle->bufferPointer[1] = 0;
6530 timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows.
6531 MUTEX_UNLOCK( &stream_.mutex );
6533 if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );
6536 void RtApiDs :: abortStream()
6539 if ( stream_.state == STREAM_STOPPED ) {
6540 errorText_ = "RtApiDs::abortStream(): the stream is already stopped!";
6541 error( RtAudioError::WARNING );
6545 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6546 handle->drainCounter = 2;
6551 void RtApiDs :: callbackEvent()
6553 if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) {
6554 Sleep( 50 ); // sleep 50 milliseconds
6558 if ( stream_.state == STREAM_CLOSED ) {
6559 errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!";
6560 error( RtAudioError::WARNING );
6564 CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
6565 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6567 // Check if we were draining the stream and signal is finished.
6568 if ( handle->drainCounter > stream_.nBuffers + 2 ) {
6570 stream_.state = STREAM_STOPPING;
6571 if ( handle->internalDrain == false )
6572 SetEvent( handle->condition );
6578 // Invoke user callback to get fresh output data UNLESS we are
6580 if ( handle->drainCounter == 0 ) {
6581 RtAudioCallback callback = (RtAudioCallback) info->callback;
6582 double streamTime = getStreamTime();
6583 RtAudioStreamStatus status = 0;
6584 if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
6585 status |= RTAUDIO_OUTPUT_UNDERFLOW;
6586 handle->xrun[0] = false;
6588 if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
6589 status |= RTAUDIO_INPUT_OVERFLOW;
6590 handle->xrun[1] = false;
6592 int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
6593 stream_.bufferSize, streamTime, status, info->userData );
6594 if ( cbReturnValue == 2 ) {
6595 stream_.state = STREAM_STOPPING;
6596 handle->drainCounter = 2;
6600 else if ( cbReturnValue == 1 ) {
6601 handle->drainCounter = 1;
6602 handle->internalDrain = true;
6607 DWORD currentWritePointer, safeWritePointer;
6608 DWORD currentReadPointer, safeReadPointer;
6609 UINT nextWritePointer;
6611 LPVOID buffer1 = NULL;
6612 LPVOID buffer2 = NULL;
6613 DWORD bufferSize1 = 0;
6614 DWORD bufferSize2 = 0;
6619 MUTEX_LOCK( &stream_.mutex );
6620 if ( stream_.state == STREAM_STOPPED ) {
6621 MUTEX_UNLOCK( &stream_.mutex );
6625 if ( buffersRolling == false ) {
6626 if ( stream_.mode == DUPLEX ) {
6627 //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
6629 // It takes a while for the devices to get rolling. As a result,
6630 // there's no guarantee that the capture and write device pointers
6631 // will move in lockstep. Wait here for both devices to start
6632 // rolling, and then set our buffer pointers accordingly.
6633 // e.g. Crystal Drivers: the capture buffer starts up 5700 to 9600
6634 // bytes later than the write buffer.
6636 // Stub: a serious risk of having a pre-emptive scheduling round
6637 // take place between the two GetCurrentPosition calls... but I'm
6638 // really not sure how to solve the problem. Temporarily boost to
6639 // Realtime priority, maybe; but I'm not sure what priority the
6640 // DirectSound service threads run at. We *should* be roughly
6641 // within a ms or so of correct.
6643 LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6644 LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6646 DWORD startSafeWritePointer, startSafeReadPointer;
6648 result = dsWriteBuffer->GetCurrentPosition( NULL, &startSafeWritePointer );
6649 if ( FAILED( result ) ) {
6650 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
6651 errorText_ = errorStream_.str();
6652 MUTEX_UNLOCK( &stream_.mutex );
6653 error( RtAudioError::SYSTEM_ERROR );
6656 result = dsCaptureBuffer->GetCurrentPosition( NULL, &startSafeReadPointer );
6657 if ( FAILED( result ) ) {
6658 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
6659 errorText_ = errorStream_.str();
6660 MUTEX_UNLOCK( &stream_.mutex );
6661 error( RtAudioError::SYSTEM_ERROR );
6665 result = dsWriteBuffer->GetCurrentPosition( NULL, &safeWritePointer );
6666 if ( FAILED( result ) ) {
6667 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
6668 errorText_ = errorStream_.str();
6669 MUTEX_UNLOCK( &stream_.mutex );
6670 error( RtAudioError::SYSTEM_ERROR );
6673 result = dsCaptureBuffer->GetCurrentPosition( NULL, &safeReadPointer );
6674 if ( FAILED( result ) ) {
6675 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
6676 errorText_ = errorStream_.str();
6677 MUTEX_UNLOCK( &stream_.mutex );
6678 error( RtAudioError::SYSTEM_ERROR );
6681 if ( safeWritePointer != startSafeWritePointer && safeReadPointer != startSafeReadPointer ) break;
6685 //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
6687 handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];
6688 if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];
6689 handle->bufferPointer[1] = safeReadPointer;
6691 else if ( stream_.mode == OUTPUT ) {
6693 // Set the proper nextWritePosition after initial startup.
6694 LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6695 result = dsWriteBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer );
6696 if ( FAILED( result ) ) {
6697 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
6698 errorText_ = errorStream_.str();
6699 MUTEX_UNLOCK( &stream_.mutex );
6700 error( RtAudioError::SYSTEM_ERROR );
6703 handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];
6704 if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];
6707 buffersRolling = true;
6710 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
6712 LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6714 if ( handle->drainCounter > 1 ) { // write zeros to the output stream
6715 bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
6716 bufferBytes *= formatBytes( stream_.userFormat );
6717 memset( stream_.userBuffer[0], 0, bufferBytes );
6720 // Setup parameters and do buffer conversion if necessary.
6721 if ( stream_.doConvertBuffer[0] ) {
6722 buffer = stream_.deviceBuffer;
6723 convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
6724 bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[0];
6725 bufferBytes *= formatBytes( stream_.deviceFormat[0] );
6728 buffer = stream_.userBuffer[0];
6729 bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
6730 bufferBytes *= formatBytes( stream_.userFormat );
6733 // No byte swapping necessary in DirectSound implementation.
6735 // Ahhh ... windoze. 16-bit data is signed but 8-bit data is
6736 // unsigned. So, we need to convert our signed 8-bit data here to
6738 if ( stream_.deviceFormat[0] == RTAUDIO_SINT8 )
6739 for ( int i=0; i<bufferBytes; i++ ) buffer[i] = (unsigned char) ( buffer[i] + 128 );
6741 DWORD dsBufferSize = handle->dsBufferSize[0];
6742 nextWritePointer = handle->bufferPointer[0];
6744 DWORD endWrite, leadPointer;
6746 // Find out where the read and "safe write" pointers are.
6747 result = dsBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer );
6748 if ( FAILED( result ) ) {
6749 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
6750 errorText_ = errorStream_.str();
6751 MUTEX_UNLOCK( &stream_.mutex );
6752 error( RtAudioError::SYSTEM_ERROR );
6756 // We will copy our output buffer into the region between
6757 // safeWritePointer and leadPointer. If leadPointer is not
6758 // beyond the next endWrite position, wait until it is.
6759 leadPointer = safeWritePointer + handle->dsPointerLeadTime[0];
6760 //std::cout << "safeWritePointer = " << safeWritePointer << ", leadPointer = " << leadPointer << ", nextWritePointer = " << nextWritePointer << std::endl;
6761 if ( leadPointer > dsBufferSize ) leadPointer -= dsBufferSize;
6762 if ( leadPointer < nextWritePointer ) leadPointer += dsBufferSize; // unwrap offset
6763 endWrite = nextWritePointer + bufferBytes;
6765 // Check whether the entire write region is behind the play pointer.
6766 if ( leadPointer >= endWrite ) break;
6768 // If we are here, then we must wait until the leadPointer advances
6769 // beyond the end of our next write region. We use the
6770 // Sleep() function to suspend operation until that happens.
6771 double millis = ( endWrite - leadPointer ) * 1000.0;
6772 millis /= ( formatBytes( stream_.deviceFormat[0]) * stream_.nDeviceChannels[0] * stream_.sampleRate);
6773 if ( millis < 1.0 ) millis = 1.0;
6774 Sleep( (DWORD) millis );
6777 if ( dsPointerBetween( nextWritePointer, safeWritePointer, currentWritePointer, dsBufferSize )
6778 || dsPointerBetween( endWrite, safeWritePointer, currentWritePointer, dsBufferSize ) ) {
6779 // We've strayed into the forbidden zone ... resync the read pointer.
6780 handle->xrun[0] = true;
6781 nextWritePointer = safeWritePointer + handle->dsPointerLeadTime[0] - bufferBytes;
6782 if ( nextWritePointer >= dsBufferSize ) nextWritePointer -= dsBufferSize;
6783 handle->bufferPointer[0] = nextWritePointer;
6784 endWrite = nextWritePointer + bufferBytes;
6787 // Lock free space in the buffer
6788 result = dsBuffer->Lock( nextWritePointer, bufferBytes, &buffer1,
6789 &bufferSize1, &buffer2, &bufferSize2, 0 );
6790 if ( FAILED( result ) ) {
6791 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking buffer during playback!";
6792 errorText_ = errorStream_.str();
6793 MUTEX_UNLOCK( &stream_.mutex );
6794 error( RtAudioError::SYSTEM_ERROR );
6798 // Copy our buffer into the DS buffer
6799 CopyMemory( buffer1, buffer, bufferSize1 );
6800 if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 );
6802 // Update our buffer offset and unlock sound buffer
6803 dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
6804 if ( FAILED( result ) ) {
6805 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!";
6806 errorText_ = errorStream_.str();
6807 MUTEX_UNLOCK( &stream_.mutex );
6808 error( RtAudioError::SYSTEM_ERROR );
6811 nextWritePointer = ( nextWritePointer + bufferSize1 + bufferSize2 ) % dsBufferSize;
6812 handle->bufferPointer[0] = nextWritePointer;
6815 // Don't bother draining input
6816 if ( handle->drainCounter ) {
6817 handle->drainCounter++;
6821 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
6823 // Setup parameters.
6824 if ( stream_.doConvertBuffer[1] ) {
6825 buffer = stream_.deviceBuffer;
6826 bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[1];
6827 bufferBytes *= formatBytes( stream_.deviceFormat[1] );
6830 buffer = stream_.userBuffer[1];
6831 bufferBytes = stream_.bufferSize * stream_.nUserChannels[1];
6832 bufferBytes *= formatBytes( stream_.userFormat );
6835 LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6836 long nextReadPointer = handle->bufferPointer[1];
6837 DWORD dsBufferSize = handle->dsBufferSize[1];
6839 // Find out where the write and "safe read" pointers are.
6840 result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer );
6841 if ( FAILED( result ) ) {
6842 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
6843 errorText_ = errorStream_.str();
6844 MUTEX_UNLOCK( &stream_.mutex );
6845 error( RtAudioError::SYSTEM_ERROR );
6849 if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset
6850 DWORD endRead = nextReadPointer + bufferBytes;
6852 // Handling depends on whether we are INPUT or DUPLEX.
6853 // If we're in INPUT mode then waiting is a good thing. If we're in DUPLEX mode,
6854 // then a wait here will drag the write pointers into the forbidden zone.
6856 // In DUPLEX mode, rather than wait, we will back off the read pointer until
6857 // it's in a safe position. This causes dropouts, but it seems to be the only
6858 // practical way to sync up the read and write pointers reliably, given the
6859 // the very complex relationship between phase and increment of the read and write
6862 // In order to minimize audible dropouts in DUPLEX mode, we will
6863 // provide a pre-roll period of 0.5 seconds in which we return
6864 // zeros from the read buffer while the pointers sync up.
6866 if ( stream_.mode == DUPLEX ) {
6867 if ( safeReadPointer < endRead ) {
6868 if ( duplexPrerollBytes <= 0 ) {
6869 // Pre-roll time over. Be more agressive.
6870 int adjustment = endRead-safeReadPointer;
6872 handle->xrun[1] = true;
6874 // - large adjustments: we've probably run out of CPU cycles, so just resync exactly,
6875 // and perform fine adjustments later.
6876 // - small adjustments: back off by twice as much.
6877 if ( adjustment >= 2*bufferBytes )
6878 nextReadPointer = safeReadPointer-2*bufferBytes;
6880 nextReadPointer = safeReadPointer-bufferBytes-adjustment;
6882 if ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;
6886 // In pre=roll time. Just do it.
6887 nextReadPointer = safeReadPointer - bufferBytes;
6888 while ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;
6890 endRead = nextReadPointer + bufferBytes;
6893 else { // mode == INPUT
6894 while ( safeReadPointer < endRead && stream_.callbackInfo.isRunning ) {
6895 // See comments for playback.
6896 double millis = (endRead - safeReadPointer) * 1000.0;
6897 millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate);
6898 if ( millis < 1.0 ) millis = 1.0;
6899 Sleep( (DWORD) millis );
6901 // Wake up and find out where we are now.
6902 result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer );
6903 if ( FAILED( result ) ) {
6904 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
6905 errorText_ = errorStream_.str();
6906 MUTEX_UNLOCK( &stream_.mutex );
6907 error( RtAudioError::SYSTEM_ERROR );
6911 if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset
6915 // Lock free space in the buffer
6916 result = dsBuffer->Lock( nextReadPointer, bufferBytes, &buffer1,
6917 &bufferSize1, &buffer2, &bufferSize2, 0 );
6918 if ( FAILED( result ) ) {
6919 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking capture buffer!";
6920 errorText_ = errorStream_.str();
6921 MUTEX_UNLOCK( &stream_.mutex );
6922 error( RtAudioError::SYSTEM_ERROR );
6926 if ( duplexPrerollBytes <= 0 ) {
6927 // Copy our buffer into the DS buffer
6928 CopyMemory( buffer, buffer1, bufferSize1 );
6929 if ( buffer2 != NULL ) CopyMemory( buffer+bufferSize1, buffer2, bufferSize2 );
6932 memset( buffer, 0, bufferSize1 );
6933 if ( buffer2 != NULL ) memset( buffer + bufferSize1, 0, bufferSize2 );
6934 duplexPrerollBytes -= bufferSize1 + bufferSize2;
6937 // Update our buffer offset and unlock sound buffer
6938 nextReadPointer = ( nextReadPointer + bufferSize1 + bufferSize2 ) % dsBufferSize;
6939 dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
6940 if ( FAILED( result ) ) {
6941 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking capture buffer!";
6942 errorText_ = errorStream_.str();
6943 MUTEX_UNLOCK( &stream_.mutex );
6944 error( RtAudioError::SYSTEM_ERROR );
6947 handle->bufferPointer[1] = nextReadPointer;
6949 // No byte swapping necessary in DirectSound implementation.
6951 // If necessary, convert 8-bit data from unsigned to signed.
6952 if ( stream_.deviceFormat[1] == RTAUDIO_SINT8 )
6953 for ( int j=0; j<bufferBytes; j++ ) buffer[j] = (signed char) ( buffer[j] - 128 );
6955 // Do buffer conversion if necessary.
6956 if ( stream_.doConvertBuffer[1] )
6957 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
6961 MUTEX_UNLOCK( &stream_.mutex );
6962 RtApi::tickStreamTime();
6965 // Definitions for utility functions and callbacks
6966 // specific to the DirectSound implementation.
6968 static unsigned __stdcall callbackHandler( void *ptr )
6970 CallbackInfo *info = (CallbackInfo *) ptr;
6971 RtApiDs *object = (RtApiDs *) info->object;
6972 bool* isRunning = &info->isRunning;
6974 while ( *isRunning == true ) {
6975 object->callbackEvent();
6982 static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
6983 LPCTSTR description,
6987 struct DsProbeData& probeInfo = *(struct DsProbeData*) lpContext;
6988 std::vector<struct DsDevice>& dsDevices = *probeInfo.dsDevices;
6991 bool validDevice = false;
6992 if ( probeInfo.isInput == true ) {
6994 LPDIRECTSOUNDCAPTURE object;
6996 hr = DirectSoundCaptureCreate( lpguid, &object, NULL );
6997 if ( hr != DS_OK ) return TRUE;
6999 caps.dwSize = sizeof(caps);
7000 hr = object->GetCaps( &caps );
7001 if ( hr == DS_OK ) {
7002 if ( caps.dwChannels > 0 && caps.dwFormats > 0 )
7009 LPDIRECTSOUND object;
7010 hr = DirectSoundCreate( lpguid, &object, NULL );
7011 if ( hr != DS_OK ) return TRUE;
7013 caps.dwSize = sizeof(caps);
7014 hr = object->GetCaps( &caps );
7015 if ( hr == DS_OK ) {
7016 if ( caps.dwFlags & DSCAPS_PRIMARYMONO || caps.dwFlags & DSCAPS_PRIMARYSTEREO )
7022 // If good device, then save its name and guid.
7023 std::string name = convertCharPointerToStdString( description );
7024 //if ( name == "Primary Sound Driver" || name == "Primary Sound Capture Driver" )
7025 if ( lpguid == NULL )
7026 name = "Default Device";
7027 if ( validDevice ) {
7028 for ( unsigned int i=0; i<dsDevices.size(); i++ ) {
7029 if ( dsDevices[i].name == name ) {
7030 dsDevices[i].found = true;
7031 if ( probeInfo.isInput ) {
7032 dsDevices[i].id[1] = lpguid;
7033 dsDevices[i].validId[1] = true;
7036 dsDevices[i].id[0] = lpguid;
7037 dsDevices[i].validId[0] = true;
7045 device.found = true;
7046 if ( probeInfo.isInput ) {
7047 device.id[1] = lpguid;
7048 device.validId[1] = true;
7051 device.id[0] = lpguid;
7052 device.validId[0] = true;
7054 dsDevices.push_back( device );
7060 static const char* getErrorString( int code )
7064 case DSERR_ALLOCATED:
7065 return "Already allocated";
7067 case DSERR_CONTROLUNAVAIL:
7068 return "Control unavailable";
7070 case DSERR_INVALIDPARAM:
7071 return "Invalid parameter";
7073 case DSERR_INVALIDCALL:
7074 return "Invalid call";
7077 return "Generic error";
7079 case DSERR_PRIOLEVELNEEDED:
7080 return "Priority level needed";
7082 case DSERR_OUTOFMEMORY:
7083 return "Out of memory";
7085 case DSERR_BADFORMAT:
7086 return "The sample rate or the channel format is not supported";
7088 case DSERR_UNSUPPORTED:
7089 return "Not supported";
7091 case DSERR_NODRIVER:
7094 case DSERR_ALREADYINITIALIZED:
7095 return "Already initialized";
7097 case DSERR_NOAGGREGATION:
7098 return "No aggregation";
7100 case DSERR_BUFFERLOST:
7101 return "Buffer lost";
7103 case DSERR_OTHERAPPHASPRIO:
7104 return "Another application already has priority";
7106 case DSERR_UNINITIALIZED:
7107 return "Uninitialized";
7110 return "DirectSound unknown error";
7113 //******************** End of __WINDOWS_DS__ *********************//
7117 #if defined(__LINUX_ALSA__)
7119 #include <alsa/asoundlib.h>
7122 // A structure to hold various information related to the ALSA API
7125 snd_pcm_t *handles[2];
7128 pthread_cond_t runnable_cv;
7132 :synchronized(false), runnable(false) { xrun[0] = false; xrun[1] = false; }
7135 static void *alsaCallbackHandler( void * ptr );
7137 RtApiAlsa :: RtApiAlsa()
7139 // Nothing to do here.
7142 RtApiAlsa :: ~RtApiAlsa()
7144 if ( stream_.state != STREAM_CLOSED ) closeStream();
7147 unsigned int RtApiAlsa :: getDeviceCount( void )
7149 unsigned nDevices = 0;
7150 int result, subdevice, card;
7154 // Count cards and devices
7156 snd_card_next( &card );
7157 while ( card >= 0 ) {
7158 sprintf( name, "hw:%d", card );
7159 result = snd_ctl_open( &handle, name, 0 );
7161 errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << ".";
7162 errorText_ = errorStream_.str();
7163 error( RtAudioError::WARNING );
7168 result = snd_ctl_pcm_next_device( handle, &subdevice );
7170 errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
7171 errorText_ = errorStream_.str();
7172 error( RtAudioError::WARNING );
7175 if ( subdevice < 0 )
7180 snd_ctl_close( handle );
7181 snd_card_next( &card );
7184 result = snd_ctl_open( &handle, "default", 0 );
7187 snd_ctl_close( handle );
7193 RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device )
7195 RtAudio::DeviceInfo info;
7196 info.probed = false;
7198 unsigned nDevices = 0;
7199 int result, subdevice, card;
7203 // Count cards and devices
7206 snd_card_next( &card );
7207 while ( card >= 0 ) {
7208 sprintf( name, "hw:%d", card );
7209 result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
7211 errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << ".";
7212 errorText_ = errorStream_.str();
7213 error( RtAudioError::WARNING );
7218 result = snd_ctl_pcm_next_device( chandle, &subdevice );
7220 errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
7221 errorText_ = errorStream_.str();
7222 error( RtAudioError::WARNING );
7225 if ( subdevice < 0 ) break;
7226 if ( nDevices == device ) {
7227 sprintf( name, "hw:%d,%d", card, subdevice );
7233 snd_ctl_close( chandle );
7234 snd_card_next( &card );
7237 result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
7238 if ( result == 0 ) {
7239 if ( nDevices == device ) {
7240 strcpy( name, "default" );
7246 if ( nDevices == 0 ) {
7247 errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!";
7248 error( RtAudioError::INVALID_USE );
7252 if ( device >= nDevices ) {
7253 errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!";
7254 error( RtAudioError::INVALID_USE );
7260 // If a stream is already open, we cannot probe the stream devices.
7261 // Thus, use the saved results.
7262 if ( stream_.state != STREAM_CLOSED &&
7263 ( stream_.device[0] == device || stream_.device[1] == device ) ) {
7264 snd_ctl_close( chandle );
7265 if ( device >= devices_.size() ) {
7266 errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened.";
7267 error( RtAudioError::WARNING );
7270 return devices_[ device ];
7273 int openMode = SND_PCM_ASYNC;
7274 snd_pcm_stream_t stream;
7275 snd_pcm_info_t *pcminfo;
7276 snd_pcm_info_alloca( &pcminfo );
7278 snd_pcm_hw_params_t *params;
7279 snd_pcm_hw_params_alloca( ¶ms );
7281 // First try for playback unless default device (which has subdev -1)
7282 stream = SND_PCM_STREAM_PLAYBACK;
7283 snd_pcm_info_set_stream( pcminfo, stream );
7284 if ( subdevice != -1 ) {
7285 snd_pcm_info_set_device( pcminfo, subdevice );
7286 snd_pcm_info_set_subdevice( pcminfo, 0 );
7288 result = snd_ctl_pcm_info( chandle, pcminfo );
7290 // Device probably doesn't support playback.
7295 result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK );
7297 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
7298 errorText_ = errorStream_.str();
7299 error( RtAudioError::WARNING );
7303 // The device is open ... fill the parameter structure.
7304 result = snd_pcm_hw_params_any( phandle, params );
7306 snd_pcm_close( phandle );
7307 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
7308 errorText_ = errorStream_.str();
7309 error( RtAudioError::WARNING );
7313 // Get output channel information.
7315 result = snd_pcm_hw_params_get_channels_max( params, &value );
7317 snd_pcm_close( phandle );
7318 errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << ".";
7319 errorText_ = errorStream_.str();
7320 error( RtAudioError::WARNING );
7323 info.outputChannels = value;
7324 snd_pcm_close( phandle );
7327 stream = SND_PCM_STREAM_CAPTURE;
7328 snd_pcm_info_set_stream( pcminfo, stream );
7330 // Now try for capture unless default device (with subdev = -1)
7331 if ( subdevice != -1 ) {
7332 result = snd_ctl_pcm_info( chandle, pcminfo );
7333 snd_ctl_close( chandle );
7335 // Device probably doesn't support capture.
7336 if ( info.outputChannels == 0 ) return info;
7337 goto probeParameters;
7341 snd_ctl_close( chandle );
7343 result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
7345 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
7346 errorText_ = errorStream_.str();
7347 error( RtAudioError::WARNING );
7348 if ( info.outputChannels == 0 ) return info;
7349 goto probeParameters;
7352 // The device is open ... fill the parameter structure.
7353 result = snd_pcm_hw_params_any( phandle, params );
7355 snd_pcm_close( phandle );
7356 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
7357 errorText_ = errorStream_.str();
7358 error( RtAudioError::WARNING );
7359 if ( info.outputChannels == 0 ) return info;
7360 goto probeParameters;
7363 result = snd_pcm_hw_params_get_channels_max( params, &value );
7365 snd_pcm_close( phandle );
7366 errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << ".";
7367 errorText_ = errorStream_.str();
7368 error( RtAudioError::WARNING );
7369 if ( info.outputChannels == 0 ) return info;
7370 goto probeParameters;
7372 info.inputChannels = value;
7373 snd_pcm_close( phandle );
7375 // If device opens for both playback and capture, we determine the channels.
7376 if ( info.outputChannels > 0 && info.inputChannels > 0 )
7377 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
7379 // ALSA doesn't provide default devices so we'll use the first available one.
7380 if ( device == 0 && info.outputChannels > 0 )
7381 info.isDefaultOutput = true;
7382 if ( device == 0 && info.inputChannels > 0 )
7383 info.isDefaultInput = true;
7386 // At this point, we just need to figure out the supported data
7387 // formats and sample rates. We'll proceed by opening the device in
7388 // the direction with the maximum number of channels, or playback if
7389 // they are equal. This might limit our sample rate options, but so
7392 if ( info.outputChannels >= info.inputChannels )
7393 stream = SND_PCM_STREAM_PLAYBACK;
7395 stream = SND_PCM_STREAM_CAPTURE;
7396 snd_pcm_info_set_stream( pcminfo, stream );
7398 result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
7400 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
7401 errorText_ = errorStream_.str();
7402 error( RtAudioError::WARNING );
7406 // The device is open ... fill the parameter structure.
7407 result = snd_pcm_hw_params_any( phandle, params );
7409 snd_pcm_close( phandle );
7410 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
7411 errorText_ = errorStream_.str();
7412 error( RtAudioError::WARNING );
7416 // Test our discrete set of sample rate values.
7417 info.sampleRates.clear();
7418 for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {
7419 if ( snd_pcm_hw_params_test_rate( phandle, params, SAMPLE_RATES[i], 0 ) == 0 ) {
7420 info.sampleRates.push_back( SAMPLE_RATES[i] );
7422 if ( !info.preferredSampleRate || ( SAMPLE_RATES[i] <= 48000 && SAMPLE_RATES[i] > info.preferredSampleRate ) )
7423 info.preferredSampleRate = SAMPLE_RATES[i];
7426 if ( info.sampleRates.size() == 0 ) {
7427 snd_pcm_close( phandle );
7428 errorStream_ << "RtApiAlsa::getDeviceInfo: no supported sample rates found for device (" << name << ").";
7429 errorText_ = errorStream_.str();
7430 error( RtAudioError::WARNING );
7434 // Probe the supported data formats ... we don't care about endian-ness just yet
7435 snd_pcm_format_t format;
7436 info.nativeFormats = 0;
7437 format = SND_PCM_FORMAT_S8;
7438 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7439 info.nativeFormats |= RTAUDIO_SINT8;
7440 format = SND_PCM_FORMAT_S16;
7441 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7442 info.nativeFormats |= RTAUDIO_SINT16;
7443 format = SND_PCM_FORMAT_S24;
7444 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7445 info.nativeFormats |= RTAUDIO_SINT24;
7446 format = SND_PCM_FORMAT_S32;
7447 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7448 info.nativeFormats |= RTAUDIO_SINT32;
7449 format = SND_PCM_FORMAT_FLOAT;
7450 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7451 info.nativeFormats |= RTAUDIO_FLOAT32;
7452 format = SND_PCM_FORMAT_FLOAT64;
7453 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7454 info.nativeFormats |= RTAUDIO_FLOAT64;
7456 // Check that we have at least one supported format
7457 if ( info.nativeFormats == 0 ) {
7458 snd_pcm_close( phandle );
7459 errorStream_ << "RtApiAlsa::getDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio.";
7460 errorText_ = errorStream_.str();
7461 error( RtAudioError::WARNING );
7465 // Get the device name
7467 result = snd_card_get_name( card, &cardname );
7468 if ( result >= 0 ) {
7469 sprintf( name, "hw:%s,%d", cardname, subdevice );
7474 // That's all ... close the device and return
7475 snd_pcm_close( phandle );
7480 void RtApiAlsa :: saveDeviceInfo( void )
7484 unsigned int nDevices = getDeviceCount();
7485 devices_.resize( nDevices );
7486 for ( unsigned int i=0; i<nDevices; i++ )
7487 devices_[i] = getDeviceInfo( i );
7490 bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
7491 unsigned int firstChannel, unsigned int sampleRate,
7492 RtAudioFormat format, unsigned int *bufferSize,
7493 RtAudio::StreamOptions *options )
7496 #if defined(__RTAUDIO_DEBUG__)
7498 snd_output_stdio_attach(&out, stderr, 0);
7501 // I'm not using the "plug" interface ... too much inconsistent behavior.
7503 unsigned nDevices = 0;
7504 int result, subdevice, card;
7508 if ( options && options->flags & RTAUDIO_ALSA_USE_DEFAULT )
7509 snprintf(name, sizeof(name), "%s", "default");
7511 // Count cards and devices
7513 snd_card_next( &card );
7514 while ( card >= 0 ) {
7515 sprintf( name, "hw:%d", card );
7516 result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
7518 errorStream_ << "RtApiAlsa::probeDeviceOpen: control open, card = " << card << ", " << snd_strerror( result ) << ".";
7519 errorText_ = errorStream_.str();
7524 result = snd_ctl_pcm_next_device( chandle, &subdevice );
7525 if ( result < 0 ) break;
7526 if ( subdevice < 0 ) break;
7527 if ( nDevices == device ) {
7528 sprintf( name, "hw:%d,%d", card, subdevice );
7529 snd_ctl_close( chandle );
7534 snd_ctl_close( chandle );
7535 snd_card_next( &card );
7538 result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
7539 if ( result == 0 ) {
7540 if ( nDevices == device ) {
7541 strcpy( name, "default" );
7542 snd_ctl_close( chandle );
7547 snd_ctl_close( chandle );
7549 if ( nDevices == 0 ) {
7550 // This should not happen because a check is made before this function is called.
7551 errorText_ = "RtApiAlsa::probeDeviceOpen: no devices found!";
7555 if ( device >= nDevices ) {
7556 // This should not happen because a check is made before this function is called.
7557 errorText_ = "RtApiAlsa::probeDeviceOpen: device ID is invalid!";
7564 // The getDeviceInfo() function will not work for a device that is
7565 // already open. Thus, we'll probe the system before opening a
7566 // stream and save the results for use by getDeviceInfo().
7567 if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) // only do once
7568 this->saveDeviceInfo();
7570 snd_pcm_stream_t stream;
7571 if ( mode == OUTPUT )
7572 stream = SND_PCM_STREAM_PLAYBACK;
7574 stream = SND_PCM_STREAM_CAPTURE;
7577 int openMode = SND_PCM_ASYNC;
7578 result = snd_pcm_open( &phandle, name, stream, openMode );
7580 if ( mode == OUTPUT )
7581 errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for output.";
7583 errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for input.";
7584 errorText_ = errorStream_.str();
7588 // Fill the parameter structure.
7589 snd_pcm_hw_params_t *hw_params;
7590 snd_pcm_hw_params_alloca( &hw_params );
7591 result = snd_pcm_hw_params_any( phandle, hw_params );
7593 snd_pcm_close( phandle );
7594 errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") parameters, " << snd_strerror( result ) << ".";
7595 errorText_ = errorStream_.str();
7599 #if defined(__RTAUDIO_DEBUG__)
7600 fprintf( stderr, "\nRtApiAlsa: dump hardware params just after device open:\n\n" );
7601 snd_pcm_hw_params_dump( hw_params, out );
7604 // Set access ... check user preference.
7605 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) {
7606 stream_.userInterleaved = false;
7607 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
7609 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
7610 stream_.deviceInterleaved[mode] = true;
7613 stream_.deviceInterleaved[mode] = false;
7616 stream_.userInterleaved = true;
7617 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
7619 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
7620 stream_.deviceInterleaved[mode] = false;
7623 stream_.deviceInterleaved[mode] = true;
7627 snd_pcm_close( phandle );
7628 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") access, " << snd_strerror( result ) << ".";
7629 errorText_ = errorStream_.str();
7633 // Determine how to set the device format.
7634 stream_.userFormat = format;
7635 snd_pcm_format_t deviceFormat = SND_PCM_FORMAT_UNKNOWN;
7637 if ( format == RTAUDIO_SINT8 )
7638 deviceFormat = SND_PCM_FORMAT_S8;
7639 else if ( format == RTAUDIO_SINT16 )
7640 deviceFormat = SND_PCM_FORMAT_S16;
7641 else if ( format == RTAUDIO_SINT24 )
7642 deviceFormat = SND_PCM_FORMAT_S24;
7643 else if ( format == RTAUDIO_SINT32 )
7644 deviceFormat = SND_PCM_FORMAT_S32;
7645 else if ( format == RTAUDIO_FLOAT32 )
7646 deviceFormat = SND_PCM_FORMAT_FLOAT;
7647 else if ( format == RTAUDIO_FLOAT64 )
7648 deviceFormat = SND_PCM_FORMAT_FLOAT64;
7650 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat) == 0) {
7651 stream_.deviceFormat[mode] = format;
7655 // The user requested format is not natively supported by the device.
7656 deviceFormat = SND_PCM_FORMAT_FLOAT64;
7657 if ( snd_pcm_hw_params_test_format( phandle, hw_params, deviceFormat ) == 0 ) {
7658 stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
7662 deviceFormat = SND_PCM_FORMAT_FLOAT;
7663 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7664 stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
7668 deviceFormat = SND_PCM_FORMAT_S32;
7669 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7670 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
7674 deviceFormat = SND_PCM_FORMAT_S24;
7675 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7676 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
7680 deviceFormat = SND_PCM_FORMAT_S16;
7681 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7682 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
7686 deviceFormat = SND_PCM_FORMAT_S8;
7687 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7688 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
7692 // If we get here, no supported format was found.
7693 snd_pcm_close( phandle );
7694 errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device " << device << " data format not supported by RtAudio.";
7695 errorText_ = errorStream_.str();
7699 result = snd_pcm_hw_params_set_format( phandle, hw_params, deviceFormat );
7701 snd_pcm_close( phandle );
7702 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") data format, " << snd_strerror( result ) << ".";
7703 errorText_ = errorStream_.str();
7707 // Determine whether byte-swaping is necessary.
7708 stream_.doByteSwap[mode] = false;
7709 if ( deviceFormat != SND_PCM_FORMAT_S8 ) {
7710 result = snd_pcm_format_cpu_endian( deviceFormat );
7712 stream_.doByteSwap[mode] = true;
7713 else if (result < 0) {
7714 snd_pcm_close( phandle );
7715 errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") endian-ness, " << snd_strerror( result ) << ".";
7716 errorText_ = errorStream_.str();
7721 // Set the sample rate.
7722 result = snd_pcm_hw_params_set_rate_near( phandle, hw_params, (unsigned int*) &sampleRate, 0 );
7724 snd_pcm_close( phandle );
7725 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting sample rate on device (" << name << "), " << snd_strerror( result ) << ".";
7726 errorText_ = errorStream_.str();
7730 // Determine the number of channels for this device. We support a possible
7731 // minimum device channel number > than the value requested by the user.
7732 stream_.nUserChannels[mode] = channels;
7734 result = snd_pcm_hw_params_get_channels_max( hw_params, &value );
7735 unsigned int deviceChannels = value;
7736 if ( result < 0 || deviceChannels < channels + firstChannel ) {
7737 snd_pcm_close( phandle );
7738 errorStream_ << "RtApiAlsa::probeDeviceOpen: requested channel parameters not supported by device (" << name << "), " << snd_strerror( result ) << ".";
7739 errorText_ = errorStream_.str();
7743 result = snd_pcm_hw_params_get_channels_min( hw_params, &value );
7745 snd_pcm_close( phandle );
7746 errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting minimum channels for device (" << name << "), " << snd_strerror( result ) << ".";
7747 errorText_ = errorStream_.str();
7750 deviceChannels = value;
7751 if ( deviceChannels < channels + firstChannel ) deviceChannels = channels + firstChannel;
7752 stream_.nDeviceChannels[mode] = deviceChannels;
7754 // Set the device channels.
7755 result = snd_pcm_hw_params_set_channels( phandle, hw_params, deviceChannels );
7757 snd_pcm_close( phandle );
7758 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting channels for device (" << name << "), " << snd_strerror( result ) << ".";
7759 errorText_ = errorStream_.str();
7763 // Set the buffer (or period) size.
7765 snd_pcm_uframes_t periodSize = *bufferSize;
7766 result = snd_pcm_hw_params_set_period_size_near( phandle, hw_params, &periodSize, &dir );
7768 snd_pcm_close( phandle );
7769 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting period size for device (" << name << "), " << snd_strerror( result ) << ".";
7770 errorText_ = errorStream_.str();
7773 *bufferSize = periodSize;
7775 // Set the buffer number, which in ALSA is referred to as the "period".
7776 unsigned int periods = 0;
7777 if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2;
7778 if ( options && options->numberOfBuffers > 0 ) periods = options->numberOfBuffers;
7779 if ( periods < 2 ) periods = 4; // a fairly safe default value
7780 result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir );
7782 snd_pcm_close( phandle );
7783 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << ".";
7784 errorText_ = errorStream_.str();
7788 // If attempting to setup a duplex stream, the bufferSize parameter
7789 // MUST be the same in both directions!
7790 if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {
7791 snd_pcm_close( phandle );
7792 errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ").";
7793 errorText_ = errorStream_.str();
7797 stream_.bufferSize = *bufferSize;
7799 // Install the hardware configuration
7800 result = snd_pcm_hw_params( phandle, hw_params );
7802 snd_pcm_close( phandle );
7803 errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing hardware configuration on device (" << name << "), " << snd_strerror( result ) << ".";
7804 errorText_ = errorStream_.str();
7808 #if defined(__RTAUDIO_DEBUG__)
7809 fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n");
7810 snd_pcm_hw_params_dump( hw_params, out );
7813 // Set the software configuration to fill buffers with zeros and prevent device stopping on xruns.
7814 snd_pcm_sw_params_t *sw_params = NULL;
7815 snd_pcm_sw_params_alloca( &sw_params );
7816 snd_pcm_sw_params_current( phandle, sw_params );
7817 snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize );
7818 snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, ULONG_MAX );
7819 snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 );
7821 // The following two settings were suggested by Theo Veenker
7822 //snd_pcm_sw_params_set_avail_min( phandle, sw_params, *bufferSize );
7823 //snd_pcm_sw_params_set_xfer_align( phandle, sw_params, 1 );
7825 // here are two options for a fix
7826 //snd_pcm_sw_params_set_silence_size( phandle, sw_params, ULONG_MAX );
7827 snd_pcm_uframes_t val;
7828 snd_pcm_sw_params_get_boundary( sw_params, &val );
7829 snd_pcm_sw_params_set_silence_size( phandle, sw_params, val );
7831 result = snd_pcm_sw_params( phandle, sw_params );
7833 snd_pcm_close( phandle );
7834 errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing software configuration on device (" << name << "), " << snd_strerror( result ) << ".";
7835 errorText_ = errorStream_.str();
7839 #if defined(__RTAUDIO_DEBUG__)
7840 fprintf(stderr, "\nRtApiAlsa: dump software params after installation:\n\n");
7841 snd_pcm_sw_params_dump( sw_params, out );
7844 // Set flags for buffer conversion
7845 stream_.doConvertBuffer[mode] = false;
7846 if ( stream_.userFormat != stream_.deviceFormat[mode] )
7847 stream_.doConvertBuffer[mode] = true;
7848 if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
7849 stream_.doConvertBuffer[mode] = true;
7850 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
7851 stream_.nUserChannels[mode] > 1 )
7852 stream_.doConvertBuffer[mode] = true;
7854 // Allocate the ApiHandle if necessary and then save.
7855 AlsaHandle *apiInfo = 0;
7856 if ( stream_.apiHandle == 0 ) {
7858 apiInfo = (AlsaHandle *) new AlsaHandle;
7860 catch ( std::bad_alloc& ) {
7861 errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating AlsaHandle memory.";
7865 if ( pthread_cond_init( &apiInfo->runnable_cv, NULL ) ) {
7866 errorText_ = "RtApiAlsa::probeDeviceOpen: error initializing pthread condition variable.";
7870 stream_.apiHandle = (void *) apiInfo;
7871 apiInfo->handles[0] = 0;
7872 apiInfo->handles[1] = 0;
7875 apiInfo = (AlsaHandle *) stream_.apiHandle;
7877 apiInfo->handles[mode] = phandle;
7880 // Allocate necessary internal buffers.
7881 unsigned long bufferBytes;
7882 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
7883 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
7884 if ( stream_.userBuffer[mode] == NULL ) {
7885 errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating user buffer memory.";
7889 if ( stream_.doConvertBuffer[mode] ) {
7891 bool makeBuffer = true;
7892 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
7893 if ( mode == INPUT ) {
7894 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
7895 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
7896 if ( bufferBytes <= bytesOut ) makeBuffer = false;
7901 bufferBytes *= *bufferSize;
7902 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
7903 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
7904 if ( stream_.deviceBuffer == NULL ) {
7905 errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating device buffer memory.";
7911 stream_.sampleRate = sampleRate;
7912 stream_.nBuffers = periods;
7913 stream_.device[mode] = device;
7914 stream_.state = STREAM_STOPPED;
7916 // Setup the buffer conversion information structure.
7917 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
7919 // Setup thread if necessary.
7920 if ( stream_.mode == OUTPUT && mode == INPUT ) {
7921 // We had already set up an output stream.
7922 stream_.mode = DUPLEX;
7923 // Link the streams if possible.
7924 apiInfo->synchronized = false;
7925 if ( snd_pcm_link( apiInfo->handles[0], apiInfo->handles[1] ) == 0 )
7926 apiInfo->synchronized = true;
7928 errorText_ = "RtApiAlsa::probeDeviceOpen: unable to synchronize input and output devices.";
7929 error( RtAudioError::WARNING );
7933 stream_.mode = mode;
7935 // Setup callback thread.
7936 stream_.callbackInfo.object = (void *) this;
7938 // Set the thread attributes for joinable and realtime scheduling
7939 // priority (optional). The higher priority will only take affect
7940 // if the program is run as root or suid. Note, under Linux
7941 // processes with CAP_SYS_NICE privilege, a user can change
7942 // scheduling policy and priority (thus need not be root). See
7943 // POSIX "capabilities".
7944 pthread_attr_t attr;
7945 pthread_attr_init( &attr );
7946 pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
7947 #ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
7948 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
7949 stream_.callbackInfo.doRealtime = true;
7950 struct sched_param param;
7951 int priority = options->priority;
7952 int min = sched_get_priority_min( SCHED_RR );
7953 int max = sched_get_priority_max( SCHED_RR );
7954 if ( priority < min ) priority = min;
7955 else if ( priority > max ) priority = max;
7956 param.sched_priority = priority;
7958 // Set the policy BEFORE the priority. Otherwise it fails.
7959 pthread_attr_setschedpolicy(&attr, SCHED_RR);
7960 pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM);
7961 // This is definitely required. Otherwise it fails.
7962 pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED);
7963 pthread_attr_setschedparam(&attr, ¶m);
7966 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
7968 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
7971 stream_.callbackInfo.isRunning = true;
7972 result = pthread_create( &stream_.callbackInfo.thread, &attr, alsaCallbackHandler, &stream_.callbackInfo );
7973 pthread_attr_destroy( &attr );
7975 // Failed. Try instead with default attributes.
7976 result = pthread_create( &stream_.callbackInfo.thread, NULL, alsaCallbackHandler, &stream_.callbackInfo );
7978 stream_.callbackInfo.isRunning = false;
7979 errorText_ = "RtApiAlsa::error creating callback thread!";
7989 pthread_cond_destroy( &apiInfo->runnable_cv );
7990 if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
7991 if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
7993 stream_.apiHandle = 0;
7996 if ( phandle) snd_pcm_close( phandle );
7998 for ( int i=0; i<2; i++ ) {
7999 if ( stream_.userBuffer[i] ) {
8000 free( stream_.userBuffer[i] );
8001 stream_.userBuffer[i] = 0;
8005 if ( stream_.deviceBuffer ) {
8006 free( stream_.deviceBuffer );
8007 stream_.deviceBuffer = 0;
8010 stream_.state = STREAM_CLOSED;
8014 void RtApiAlsa :: closeStream()
8016 if ( stream_.state == STREAM_CLOSED ) {
8017 errorText_ = "RtApiAlsa::closeStream(): no open stream to close!";
8018 error( RtAudioError::WARNING );
8022 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
8023 stream_.callbackInfo.isRunning = false;
8024 MUTEX_LOCK( &stream_.mutex );
8025 if ( stream_.state == STREAM_STOPPED ) {
8026 apiInfo->runnable = true;
8027 pthread_cond_signal( &apiInfo->runnable_cv );
8029 MUTEX_UNLOCK( &stream_.mutex );
8030 pthread_join( stream_.callbackInfo.thread, NULL );
8032 if ( stream_.state == STREAM_RUNNING ) {
8033 stream_.state = STREAM_STOPPED;
8034 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
8035 snd_pcm_drop( apiInfo->handles[0] );
8036 if ( stream_.mode == INPUT || stream_.mode == DUPLEX )
8037 snd_pcm_drop( apiInfo->handles[1] );
8041 pthread_cond_destroy( &apiInfo->runnable_cv );
8042 if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
8043 if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
8045 stream_.apiHandle = 0;
8048 for ( int i=0; i<2; i++ ) {
8049 if ( stream_.userBuffer[i] ) {
8050 free( stream_.userBuffer[i] );
8051 stream_.userBuffer[i] = 0;
8055 if ( stream_.deviceBuffer ) {
8056 free( stream_.deviceBuffer );
8057 stream_.deviceBuffer = 0;
8060 stream_.mode = UNINITIALIZED;
8061 stream_.state = STREAM_CLOSED;
8064 void RtApiAlsa :: startStream()
8066 // This method calls snd_pcm_prepare if the device isn't already in that state.
8069 if ( stream_.state == STREAM_RUNNING ) {
8070 errorText_ = "RtApiAlsa::startStream(): the stream is already running!";
8071 error( RtAudioError::WARNING );
8075 MUTEX_LOCK( &stream_.mutex );
8078 snd_pcm_state_t state;
8079 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
8080 snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
8081 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
8082 state = snd_pcm_state( handle[0] );
8083 if ( state != SND_PCM_STATE_PREPARED ) {
8084 result = snd_pcm_prepare( handle[0] );
8086 errorStream_ << "RtApiAlsa::startStream: error preparing output pcm device, " << snd_strerror( result ) << ".";
8087 errorText_ = errorStream_.str();
8093 if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
8094 result = snd_pcm_drop(handle[1]); // fix to remove stale data received since device has been open
8095 state = snd_pcm_state( handle[1] );
8096 if ( state != SND_PCM_STATE_PREPARED ) {
8097 result = snd_pcm_prepare( handle[1] );
8099 errorStream_ << "RtApiAlsa::startStream: error preparing input pcm device, " << snd_strerror( result ) << ".";
8100 errorText_ = errorStream_.str();
8106 stream_.state = STREAM_RUNNING;
8109 apiInfo->runnable = true;
8110 pthread_cond_signal( &apiInfo->runnable_cv );
8111 MUTEX_UNLOCK( &stream_.mutex );
8113 if ( result >= 0 ) return;
8114 error( RtAudioError::SYSTEM_ERROR );
8117 void RtApiAlsa :: stopStream()
8120 if ( stream_.state == STREAM_STOPPED ) {
8121 errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!";
8122 error( RtAudioError::WARNING );
8126 stream_.state = STREAM_STOPPED;
8127 MUTEX_LOCK( &stream_.mutex );
8130 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
8131 snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
8132 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
8133 if ( apiInfo->synchronized )
8134 result = snd_pcm_drop( handle[0] );
8136 result = snd_pcm_drain( handle[0] );
8138 errorStream_ << "RtApiAlsa::stopStream: error draining output pcm device, " << snd_strerror( result ) << ".";
8139 errorText_ = errorStream_.str();
8144 if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
8145 result = snd_pcm_drop( handle[1] );
8147 errorStream_ << "RtApiAlsa::stopStream: error stopping input pcm device, " << snd_strerror( result ) << ".";
8148 errorText_ = errorStream_.str();
8154 apiInfo->runnable = false; // fixes high CPU usage when stopped
8155 MUTEX_UNLOCK( &stream_.mutex );
8157 if ( result >= 0 ) return;
8158 error( RtAudioError::SYSTEM_ERROR );
8161 void RtApiAlsa :: abortStream()
8164 if ( stream_.state == STREAM_STOPPED ) {
8165 errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!";
8166 error( RtAudioError::WARNING );
8170 stream_.state = STREAM_STOPPED;
8171 MUTEX_LOCK( &stream_.mutex );
8174 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
8175 snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
8176 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
8177 result = snd_pcm_drop( handle[0] );
8179 errorStream_ << "RtApiAlsa::abortStream: error aborting output pcm device, " << snd_strerror( result ) << ".";
8180 errorText_ = errorStream_.str();
8185 if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
8186 result = snd_pcm_drop( handle[1] );
8188 errorStream_ << "RtApiAlsa::abortStream: error aborting input pcm device, " << snd_strerror( result ) << ".";
8189 errorText_ = errorStream_.str();
8195 apiInfo->runnable = false; // fixes high CPU usage when stopped
8196 MUTEX_UNLOCK( &stream_.mutex );
8198 if ( result >= 0 ) return;
8199 error( RtAudioError::SYSTEM_ERROR );
8202 void RtApiAlsa :: callbackEvent()
8204 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
8205 if ( stream_.state == STREAM_STOPPED ) {
8206 MUTEX_LOCK( &stream_.mutex );
8207 while ( !apiInfo->runnable )
8208 pthread_cond_wait( &apiInfo->runnable_cv, &stream_.mutex );
8210 if ( stream_.state != STREAM_RUNNING ) {
8211 MUTEX_UNLOCK( &stream_.mutex );
8214 MUTEX_UNLOCK( &stream_.mutex );
8217 if ( stream_.state == STREAM_CLOSED ) {
8218 errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!";
8219 error( RtAudioError::WARNING );
8223 int doStopStream = 0;
8224 RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
8225 double streamTime = getStreamTime();
8226 RtAudioStreamStatus status = 0;
8227 if ( stream_.mode != INPUT && apiInfo->xrun[0] == true ) {
8228 status |= RTAUDIO_OUTPUT_UNDERFLOW;
8229 apiInfo->xrun[0] = false;
8231 if ( stream_.mode != OUTPUT && apiInfo->xrun[1] == true ) {
8232 status |= RTAUDIO_INPUT_OVERFLOW;
8233 apiInfo->xrun[1] = false;
8235 doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
8236 stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
8238 if ( doStopStream == 2 ) {
8243 MUTEX_LOCK( &stream_.mutex );
8245 // The state might change while waiting on a mutex.
8246 if ( stream_.state == STREAM_STOPPED ) goto unlock;
8252 snd_pcm_sframes_t frames;
8253 RtAudioFormat format;
8254 handle = (snd_pcm_t **) apiInfo->handles;
8256 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
8258 // Setup parameters.
8259 if ( stream_.doConvertBuffer[1] ) {
8260 buffer = stream_.deviceBuffer;
8261 channels = stream_.nDeviceChannels[1];
8262 format = stream_.deviceFormat[1];
8265 buffer = stream_.userBuffer[1];
8266 channels = stream_.nUserChannels[1];
8267 format = stream_.userFormat;
8270 // Read samples from device in interleaved/non-interleaved format.
8271 if ( stream_.deviceInterleaved[1] )
8272 result = snd_pcm_readi( handle[1], buffer, stream_.bufferSize );
8274 void *bufs[channels];
8275 size_t offset = stream_.bufferSize * formatBytes( format );
8276 for ( int i=0; i<channels; i++ )
8277 bufs[i] = (void *) (buffer + (i * offset));
8278 result = snd_pcm_readn( handle[1], bufs, stream_.bufferSize );
8281 if ( result < (int) stream_.bufferSize ) {
8282 // Either an error or overrun occured.
8283 if ( result == -EPIPE ) {
8284 snd_pcm_state_t state = snd_pcm_state( handle[1] );
8285 if ( state == SND_PCM_STATE_XRUN ) {
8286 apiInfo->xrun[1] = true;
8287 result = snd_pcm_prepare( handle[1] );
8289 errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after overrun, " << snd_strerror( result ) << ".";
8290 errorText_ = errorStream_.str();
8294 errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
8295 errorText_ = errorStream_.str();
8299 errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << ".";
8300 errorText_ = errorStream_.str();
8302 error( RtAudioError::WARNING );
8306 // Do byte swapping if necessary.
8307 if ( stream_.doByteSwap[1] )
8308 byteSwapBuffer( buffer, stream_.bufferSize * channels, format );
8310 // Do buffer conversion if necessary.
8311 if ( stream_.doConvertBuffer[1] )
8312 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
8314 // Check stream latency
8315 result = snd_pcm_delay( handle[1], &frames );
8316 if ( result == 0 && frames > 0 ) stream_.latency[1] = frames;
8321 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
8323 // Setup parameters and do buffer conversion if necessary.
8324 if ( stream_.doConvertBuffer[0] ) {
8325 buffer = stream_.deviceBuffer;
8326 convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
8327 channels = stream_.nDeviceChannels[0];
8328 format = stream_.deviceFormat[0];
8331 buffer = stream_.userBuffer[0];
8332 channels = stream_.nUserChannels[0];
8333 format = stream_.userFormat;
8336 // Do byte swapping if necessary.
8337 if ( stream_.doByteSwap[0] )
8338 byteSwapBuffer(buffer, stream_.bufferSize * channels, format);
8340 // Write samples to device in interleaved/non-interleaved format.
8341 if ( stream_.deviceInterleaved[0] )
8342 result = snd_pcm_writei( handle[0], buffer, stream_.bufferSize );
8344 void *bufs[channels];
8345 size_t offset = stream_.bufferSize * formatBytes( format );
8346 for ( int i=0; i<channels; i++ )
8347 bufs[i] = (void *) (buffer + (i * offset));
8348 result = snd_pcm_writen( handle[0], bufs, stream_.bufferSize );
8351 if ( result < (int) stream_.bufferSize ) {
8352 // Either an error or underrun occured.
8353 if ( result == -EPIPE ) {
8354 snd_pcm_state_t state = snd_pcm_state( handle[0] );
8355 if ( state == SND_PCM_STATE_XRUN ) {
8356 apiInfo->xrun[0] = true;
8357 result = snd_pcm_prepare( handle[0] );
8359 errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after underrun, " << snd_strerror( result ) << ".";
8360 errorText_ = errorStream_.str();
8363 errorText_ = "RtApiAlsa::callbackEvent: audio write error, underrun.";
8366 errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
8367 errorText_ = errorStream_.str();
8371 errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << ".";
8372 errorText_ = errorStream_.str();
8374 error( RtAudioError::WARNING );
8378 // Check stream latency
8379 result = snd_pcm_delay( handle[0], &frames );
8380 if ( result == 0 && frames > 0 ) stream_.latency[0] = frames;
8384 MUTEX_UNLOCK( &stream_.mutex );
8386 RtApi::tickStreamTime();
8387 if ( doStopStream == 1 ) this->stopStream();
8390 static void *alsaCallbackHandler( void *ptr )
8392 CallbackInfo *info = (CallbackInfo *) ptr;
8393 RtApiAlsa *object = (RtApiAlsa *) info->object;
8394 bool *isRunning = &info->isRunning;
8396 #ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
8397 if ( info->doRealtime ) {
8398 std::cerr << "RtAudio alsa: " <<
8399 (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") <<
8400 "running realtime scheduling" << std::endl;
8404 while ( *isRunning == true ) {
8405 pthread_testcancel();
8406 object->callbackEvent();
8409 pthread_exit( NULL );
8412 //******************** End of __LINUX_ALSA__ *********************//
8415 #if defined(__LINUX_PULSE__)
8417 // Code written by Peter Meerwald, pmeerw@pmeerw.net
8418 // and Tristan Matthews.
8420 #include <pulse/error.h>
8421 #include <pulse/simple.h>
8424 static const unsigned int SUPPORTED_SAMPLERATES[] = { 8000, 16000, 22050, 32000,
8425 44100, 48000, 96000, 0};
8427 struct rtaudio_pa_format_mapping_t {
8428 RtAudioFormat rtaudio_format;
8429 pa_sample_format_t pa_format;
8432 static const rtaudio_pa_format_mapping_t supported_sampleformats[] = {
8433 {RTAUDIO_SINT16, PA_SAMPLE_S16LE},
8434 {RTAUDIO_SINT32, PA_SAMPLE_S32LE},
8435 {RTAUDIO_FLOAT32, PA_SAMPLE_FLOAT32LE},
8436 {0, PA_SAMPLE_INVALID}};
8438 struct PulseAudioHandle {
8442 pthread_cond_t runnable_cv;
8444 PulseAudioHandle() : s_play(0), s_rec(0), runnable(false) { }
8447 RtApiPulse::~RtApiPulse()
8449 if ( stream_.state != STREAM_CLOSED )
8453 unsigned int RtApiPulse::getDeviceCount( void )
8458 RtAudio::DeviceInfo RtApiPulse::getDeviceInfo( unsigned int /*device*/ )
8460 RtAudio::DeviceInfo info;
8462 info.name = "PulseAudio";
8463 info.outputChannels = 2;
8464 info.inputChannels = 2;
8465 info.duplexChannels = 2;
8466 info.isDefaultOutput = true;
8467 info.isDefaultInput = true;
8469 for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr )
8470 info.sampleRates.push_back( *sr );
8472 info.preferredSampleRate = 48000;
8473 info.nativeFormats = RTAUDIO_SINT16 | RTAUDIO_SINT32 | RTAUDIO_FLOAT32;
8478 static void *pulseaudio_callback( void * user )
8480 CallbackInfo *cbi = static_cast<CallbackInfo *>( user );
8481 RtApiPulse *context = static_cast<RtApiPulse *>( cbi->object );
8482 volatile bool *isRunning = &cbi->isRunning;
8484 #ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
8485 if (cbi->doRealtime) {
8486 std::cerr << "RtAudio pulse: " <<
8487 (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") <<
8488 "running realtime scheduling" << std::endl;
8492 while ( *isRunning ) {
8493 pthread_testcancel();
8494 context->callbackEvent();
8497 pthread_exit( NULL );
8500 void RtApiPulse::closeStream( void )
8502 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8504 stream_.callbackInfo.isRunning = false;
8506 MUTEX_LOCK( &stream_.mutex );
8507 if ( stream_.state == STREAM_STOPPED ) {
8508 pah->runnable = true;
8509 pthread_cond_signal( &pah->runnable_cv );
8511 MUTEX_UNLOCK( &stream_.mutex );
8513 pthread_join( pah->thread, 0 );
8514 if ( pah->s_play ) {
8515 pa_simple_flush( pah->s_play, NULL );
8516 pa_simple_free( pah->s_play );
8519 pa_simple_free( pah->s_rec );
8521 pthread_cond_destroy( &pah->runnable_cv );
8523 stream_.apiHandle = 0;
8526 if ( stream_.userBuffer[0] ) {
8527 free( stream_.userBuffer[0] );
8528 stream_.userBuffer[0] = 0;
8530 if ( stream_.userBuffer[1] ) {
8531 free( stream_.userBuffer[1] );
8532 stream_.userBuffer[1] = 0;
8535 stream_.state = STREAM_CLOSED;
8536 stream_.mode = UNINITIALIZED;
8539 void RtApiPulse::callbackEvent( void )
8541 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8543 if ( stream_.state == STREAM_STOPPED ) {
8544 MUTEX_LOCK( &stream_.mutex );
8545 while ( !pah->runnable )
8546 pthread_cond_wait( &pah->runnable_cv, &stream_.mutex );
8548 if ( stream_.state != STREAM_RUNNING ) {
8549 MUTEX_UNLOCK( &stream_.mutex );
8552 MUTEX_UNLOCK( &stream_.mutex );
8555 if ( stream_.state == STREAM_CLOSED ) {
8556 errorText_ = "RtApiPulse::callbackEvent(): the stream is closed ... "
8557 "this shouldn't happen!";
8558 error( RtAudioError::WARNING );
8562 RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
8563 double streamTime = getStreamTime();
8564 RtAudioStreamStatus status = 0;
8565 int doStopStream = callback( stream_.userBuffer[OUTPUT], stream_.userBuffer[INPUT],
8566 stream_.bufferSize, streamTime, status,
8567 stream_.callbackInfo.userData );
8569 if ( doStopStream == 2 ) {
8574 MUTEX_LOCK( &stream_.mutex );
8575 void *pulse_in = stream_.doConvertBuffer[INPUT] ? stream_.deviceBuffer : stream_.userBuffer[INPUT];
8576 void *pulse_out = stream_.doConvertBuffer[OUTPUT] ? stream_.deviceBuffer : stream_.userBuffer[OUTPUT];
8578 if ( stream_.state != STREAM_RUNNING )
8583 if (stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
8584 if ( stream_.doConvertBuffer[OUTPUT] ) {
8585 convertBuffer( stream_.deviceBuffer,
8586 stream_.userBuffer[OUTPUT],
8587 stream_.convertInfo[OUTPUT] );
8588 bytes = stream_.nDeviceChannels[OUTPUT] * stream_.bufferSize *
8589 formatBytes( stream_.deviceFormat[OUTPUT] );
8591 bytes = stream_.nUserChannels[OUTPUT] * stream_.bufferSize *
8592 formatBytes( stream_.userFormat );
8594 if ( pa_simple_write( pah->s_play, pulse_out, bytes, &pa_error ) < 0 ) {
8595 errorStream_ << "RtApiPulse::callbackEvent: audio write error, " <<
8596 pa_strerror( pa_error ) << ".";
8597 errorText_ = errorStream_.str();
8598 error( RtAudioError::WARNING );
8602 if ( stream_.mode == INPUT || stream_.mode == DUPLEX) {
8603 if ( stream_.doConvertBuffer[INPUT] )
8604 bytes = stream_.nDeviceChannels[INPUT] * stream_.bufferSize *
8605 formatBytes( stream_.deviceFormat[INPUT] );
8607 bytes = stream_.nUserChannels[INPUT] * stream_.bufferSize *
8608 formatBytes( stream_.userFormat );
8610 if ( pa_simple_read( pah->s_rec, pulse_in, bytes, &pa_error ) < 0 ) {
8611 errorStream_ << "RtApiPulse::callbackEvent: audio read error, " <<
8612 pa_strerror( pa_error ) << ".";
8613 errorText_ = errorStream_.str();
8614 error( RtAudioError::WARNING );
8616 if ( stream_.doConvertBuffer[INPUT] ) {
8617 convertBuffer( stream_.userBuffer[INPUT],
8618 stream_.deviceBuffer,
8619 stream_.convertInfo[INPUT] );
8624 MUTEX_UNLOCK( &stream_.mutex );
8625 RtApi::tickStreamTime();
8627 if ( doStopStream == 1 )
8631 void RtApiPulse::startStream( void )
8633 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8635 if ( stream_.state == STREAM_CLOSED ) {
8636 errorText_ = "RtApiPulse::startStream(): the stream is not open!";
8637 error( RtAudioError::INVALID_USE );
8640 if ( stream_.state == STREAM_RUNNING ) {
8641 errorText_ = "RtApiPulse::startStream(): the stream is already running!";
8642 error( RtAudioError::WARNING );
8646 MUTEX_LOCK( &stream_.mutex );
8648 stream_.state = STREAM_RUNNING;
8650 pah->runnable = true;
8651 pthread_cond_signal( &pah->runnable_cv );
8652 MUTEX_UNLOCK( &stream_.mutex );
8655 void RtApiPulse::stopStream( void )
8657 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8659 if ( stream_.state == STREAM_CLOSED ) {
8660 errorText_ = "RtApiPulse::stopStream(): the stream is not open!";
8661 error( RtAudioError::INVALID_USE );
8664 if ( stream_.state == STREAM_STOPPED ) {
8665 errorText_ = "RtApiPulse::stopStream(): the stream is already stopped!";
8666 error( RtAudioError::WARNING );
8670 stream_.state = STREAM_STOPPED;
8671 MUTEX_LOCK( &stream_.mutex );
8673 if ( pah && pah->s_play ) {
8675 if ( pa_simple_drain( pah->s_play, &pa_error ) < 0 ) {
8676 errorStream_ << "RtApiPulse::stopStream: error draining output device, " <<
8677 pa_strerror( pa_error ) << ".";
8678 errorText_ = errorStream_.str();
8679 MUTEX_UNLOCK( &stream_.mutex );
8680 error( RtAudioError::SYSTEM_ERROR );
8685 stream_.state = STREAM_STOPPED;
8686 MUTEX_UNLOCK( &stream_.mutex );
8689 void RtApiPulse::abortStream( void )
8691 PulseAudioHandle *pah = static_cast<PulseAudioHandle*>( stream_.apiHandle );
8693 if ( stream_.state == STREAM_CLOSED ) {
8694 errorText_ = "RtApiPulse::abortStream(): the stream is not open!";
8695 error( RtAudioError::INVALID_USE );
8698 if ( stream_.state == STREAM_STOPPED ) {
8699 errorText_ = "RtApiPulse::abortStream(): the stream is already stopped!";
8700 error( RtAudioError::WARNING );
8704 stream_.state = STREAM_STOPPED;
8705 MUTEX_LOCK( &stream_.mutex );
8707 if ( pah && pah->s_play ) {
8709 if ( pa_simple_flush( pah->s_play, &pa_error ) < 0 ) {
8710 errorStream_ << "RtApiPulse::abortStream: error flushing output device, " <<
8711 pa_strerror( pa_error ) << ".";
8712 errorText_ = errorStream_.str();
8713 MUTEX_UNLOCK( &stream_.mutex );
8714 error( RtAudioError::SYSTEM_ERROR );
8719 stream_.state = STREAM_STOPPED;
8720 MUTEX_UNLOCK( &stream_.mutex );
8723 bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode,
8724 unsigned int channels, unsigned int firstChannel,
8725 unsigned int sampleRate, RtAudioFormat format,
8726 unsigned int *bufferSize, RtAudio::StreamOptions *options )
8728 PulseAudioHandle *pah = 0;
8729 unsigned long bufferBytes = 0;
8732 if ( device != 0 ) return false;
8733 if ( mode != INPUT && mode != OUTPUT ) return false;
8734 if ( channels != 1 && channels != 2 ) {
8735 errorText_ = "RtApiPulse::probeDeviceOpen: unsupported number of channels.";
8738 ss.channels = channels;
8740 if ( firstChannel != 0 ) return false;
8742 bool sr_found = false;
8743 for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr ) {
8744 if ( sampleRate == *sr ) {
8746 stream_.sampleRate = sampleRate;
8747 ss.rate = sampleRate;
8752 errorText_ = "RtApiPulse::probeDeviceOpen: unsupported sample rate.";
8757 for ( const rtaudio_pa_format_mapping_t *sf = supported_sampleformats;
8758 sf->rtaudio_format && sf->pa_format != PA_SAMPLE_INVALID; ++sf ) {
8759 if ( format == sf->rtaudio_format ) {
8761 stream_.userFormat = sf->rtaudio_format;
8762 stream_.deviceFormat[mode] = stream_.userFormat;
8763 ss.format = sf->pa_format;
8767 if ( !sf_found ) { // Use internal data format conversion.
8768 stream_.userFormat = format;
8769 stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
8770 ss.format = PA_SAMPLE_FLOAT32LE;
8773 // Set other stream parameters.
8774 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
8775 else stream_.userInterleaved = true;
8776 stream_.deviceInterleaved[mode] = true;
8777 stream_.nBuffers = 1;
8778 stream_.doByteSwap[mode] = false;
8779 stream_.nUserChannels[mode] = channels;
8780 stream_.nDeviceChannels[mode] = channels + firstChannel;
8781 stream_.channelOffset[mode] = 0;
8782 std::string streamName = "RtAudio";
8784 // Set flags for buffer conversion.
8785 stream_.doConvertBuffer[mode] = false;
8786 if ( stream_.userFormat != stream_.deviceFormat[mode] )
8787 stream_.doConvertBuffer[mode] = true;
8788 if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
8789 stream_.doConvertBuffer[mode] = true;
8791 // Allocate necessary internal buffers.
8792 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
8793 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
8794 if ( stream_.userBuffer[mode] == NULL ) {
8795 errorText_ = "RtApiPulse::probeDeviceOpen: error allocating user buffer memory.";
8798 stream_.bufferSize = *bufferSize;
8800 if ( stream_.doConvertBuffer[mode] ) {
8802 bool makeBuffer = true;
8803 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
8804 if ( mode == INPUT ) {
8805 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
8806 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
8807 if ( bufferBytes <= bytesOut ) makeBuffer = false;
8812 bufferBytes *= *bufferSize;
8813 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
8814 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
8815 if ( stream_.deviceBuffer == NULL ) {
8816 errorText_ = "RtApiPulse::probeDeviceOpen: error allocating device buffer memory.";
8822 stream_.device[mode] = device;
8824 // Setup the buffer conversion information structure.
8825 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
8827 if ( !stream_.apiHandle ) {
8828 PulseAudioHandle *pah = new PulseAudioHandle;
8830 errorText_ = "RtApiPulse::probeDeviceOpen: error allocating memory for handle.";
8834 stream_.apiHandle = pah;
8835 if ( pthread_cond_init( &pah->runnable_cv, NULL ) != 0 ) {
8836 errorText_ = "RtApiPulse::probeDeviceOpen: error creating condition variable.";
8840 pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8843 if ( options && !options->streamName.empty() ) streamName = options->streamName;
8846 pa_buffer_attr buffer_attr;
8847 buffer_attr.fragsize = bufferBytes;
8848 buffer_attr.maxlength = -1;
8850 pah->s_rec = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_RECORD, NULL, "Record", &ss, NULL, &buffer_attr, &error );
8851 if ( !pah->s_rec ) {
8852 errorText_ = "RtApiPulse::probeDeviceOpen: error connecting input to PulseAudio server.";
8857 pah->s_play = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_PLAYBACK, NULL, "Playback", &ss, NULL, NULL, &error );
8858 if ( !pah->s_play ) {
8859 errorText_ = "RtApiPulse::probeDeviceOpen: error connecting output to PulseAudio server.";
8867 if ( stream_.mode == UNINITIALIZED )
8868 stream_.mode = mode;
8869 else if ( stream_.mode == mode )
8872 stream_.mode = DUPLEX;
8874 if ( !stream_.callbackInfo.isRunning ) {
8875 stream_.callbackInfo.object = this;
8877 stream_.state = STREAM_STOPPED;
8878 // Set the thread attributes for joinable and realtime scheduling
8879 // priority (optional). The higher priority will only take affect
8880 // if the program is run as root or suid. Note, under Linux
8881 // processes with CAP_SYS_NICE privilege, a user can change
8882 // scheduling policy and priority (thus need not be root). See
8883 // POSIX "capabilities".
8884 pthread_attr_t attr;
8885 pthread_attr_init( &attr );
8886 pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
8887 #ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
8888 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
8889 stream_.callbackInfo.doRealtime = true;
8890 struct sched_param param;
8891 int priority = options->priority;
8892 int min = sched_get_priority_min( SCHED_RR );
8893 int max = sched_get_priority_max( SCHED_RR );
8894 if ( priority < min ) priority = min;
8895 else if ( priority > max ) priority = max;
8896 param.sched_priority = priority;
8898 // Set the policy BEFORE the priority. Otherwise it fails.
8899 pthread_attr_setschedpolicy(&attr, SCHED_RR);
8900 pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM);
8901 // This is definitely required. Otherwise it fails.
8902 pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED);
8903 pthread_attr_setschedparam(&attr, ¶m);
8906 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
8908 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
8911 stream_.callbackInfo.isRunning = true;
8912 int result = pthread_create( &pah->thread, &attr, pulseaudio_callback, (void *)&stream_.callbackInfo);
8913 pthread_attr_destroy(&attr);
8915 // Failed. Try instead with default attributes.
8916 result = pthread_create( &pah->thread, NULL, pulseaudio_callback, (void *)&stream_.callbackInfo);
8918 stream_.callbackInfo.isRunning = false;
8919 errorText_ = "RtApiPulse::probeDeviceOpen: error creating thread.";
8928 if ( pah && stream_.callbackInfo.isRunning ) {
8929 pthread_cond_destroy( &pah->runnable_cv );
8931 stream_.apiHandle = 0;
8934 for ( int i=0; i<2; i++ ) {
8935 if ( stream_.userBuffer[i] ) {
8936 free( stream_.userBuffer[i] );
8937 stream_.userBuffer[i] = 0;
8941 if ( stream_.deviceBuffer ) {
8942 free( stream_.deviceBuffer );
8943 stream_.deviceBuffer = 0;
8946 stream_.state = STREAM_CLOSED;
8950 //******************** End of __LINUX_PULSE__ *********************//
8953 #if defined(__LINUX_OSS__)
8956 #include <sys/ioctl.h>
8959 #include <sys/soundcard.h>
8963 static void *ossCallbackHandler(void * ptr);
8965 // A structure to hold various information related to the OSS API
8968 int id[2]; // device ids
8971 pthread_cond_t runnable;
8974 :triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
8977 RtApiOss :: RtApiOss()
8979 // Nothing to do here.
8982 RtApiOss :: ~RtApiOss()
8984 if ( stream_.state != STREAM_CLOSED ) closeStream();
8987 unsigned int RtApiOss :: getDeviceCount( void )
8989 int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
8990 if ( mixerfd == -1 ) {
8991 errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'.";
8992 error( RtAudioError::WARNING );
8996 oss_sysinfo sysinfo;
8997 if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) {
8999 errorText_ = "RtApiOss::getDeviceCount: error getting sysinfo, OSS version >= 4.0 is required.";
9000 error( RtAudioError::WARNING );
9005 return sysinfo.numaudios;
9008 RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device )
9010 RtAudio::DeviceInfo info;
9011 info.probed = false;
9013 int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
9014 if ( mixerfd == -1 ) {
9015 errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'.";
9016 error( RtAudioError::WARNING );
9020 oss_sysinfo sysinfo;
9021 int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
9022 if ( result == -1 ) {
9024 errorText_ = "RtApiOss::getDeviceInfo: error getting sysinfo, OSS version >= 4.0 is required.";
9025 error( RtAudioError::WARNING );
9029 unsigned nDevices = sysinfo.numaudios;
9030 if ( nDevices == 0 ) {
9032 errorText_ = "RtApiOss::getDeviceInfo: no devices found!";
9033 error( RtAudioError::INVALID_USE );
9037 if ( device >= nDevices ) {
9039 errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!";
9040 error( RtAudioError::INVALID_USE );
9044 oss_audioinfo ainfo;
9046 result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
9048 if ( result == -1 ) {
9049 errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
9050 errorText_ = errorStream_.str();
9051 error( RtAudioError::WARNING );
9056 if ( ainfo.caps & PCM_CAP_OUTPUT ) info.outputChannels = ainfo.max_channels;
9057 if ( ainfo.caps & PCM_CAP_INPUT ) info.inputChannels = ainfo.max_channels;
9058 if ( ainfo.caps & PCM_CAP_DUPLEX ) {
9059 if ( info.outputChannels > 0 && info.inputChannels > 0 && ainfo.caps & PCM_CAP_DUPLEX )
9060 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
9063 // Probe data formats ... do for input
9064 unsigned long mask = ainfo.iformats;
9065 if ( mask & AFMT_S16_LE || mask & AFMT_S16_BE )
9066 info.nativeFormats |= RTAUDIO_SINT16;
9067 if ( mask & AFMT_S8 )
9068 info.nativeFormats |= RTAUDIO_SINT8;
9069 if ( mask & AFMT_S32_LE || mask & AFMT_S32_BE )
9070 info.nativeFormats |= RTAUDIO_SINT32;
9072 if ( mask & AFMT_FLOAT )
9073 info.nativeFormats |= RTAUDIO_FLOAT32;
9075 if ( mask & AFMT_S24_LE || mask & AFMT_S24_BE )
9076 info.nativeFormats |= RTAUDIO_SINT24;
9078 // Check that we have at least one supported format
9079 if ( info.nativeFormats == 0 ) {
9080 errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio.";
9081 errorText_ = errorStream_.str();
9082 error( RtAudioError::WARNING );
9086 // Probe the supported sample rates.
9087 info.sampleRates.clear();
9088 if ( ainfo.nrates ) {
9089 for ( unsigned int i=0; i<ainfo.nrates; i++ ) {
9090 for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
9091 if ( ainfo.rates[i] == SAMPLE_RATES[k] ) {
9092 info.sampleRates.push_back( SAMPLE_RATES[k] );
9094 if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
9095 info.preferredSampleRate = SAMPLE_RATES[k];
9103 // Check min and max rate values;
9104 for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
9105 if ( ainfo.min_rate <= (int) SAMPLE_RATES[k] && ainfo.max_rate >= (int) SAMPLE_RATES[k] ) {
9106 info.sampleRates.push_back( SAMPLE_RATES[k] );
9108 if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
9109 info.preferredSampleRate = SAMPLE_RATES[k];
9114 if ( info.sampleRates.size() == 0 ) {
9115 errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ").";
9116 errorText_ = errorStream_.str();
9117 error( RtAudioError::WARNING );
9121 info.name = ainfo.name;
9128 bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
9129 unsigned int firstChannel, unsigned int sampleRate,
9130 RtAudioFormat format, unsigned int *bufferSize,
9131 RtAudio::StreamOptions *options )
9133 int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
9134 if ( mixerfd == -1 ) {
9135 errorText_ = "RtApiOss::probeDeviceOpen: error opening '/dev/mixer'.";
9139 oss_sysinfo sysinfo;
9140 int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
9141 if ( result == -1 ) {
9143 errorText_ = "RtApiOss::probeDeviceOpen: error getting sysinfo, OSS version >= 4.0 is required.";
9147 unsigned nDevices = sysinfo.numaudios;
9148 if ( nDevices == 0 ) {
9149 // This should not happen because a check is made before this function is called.
9151 errorText_ = "RtApiOss::probeDeviceOpen: no devices found!";
9155 if ( device >= nDevices ) {
9156 // This should not happen because a check is made before this function is called.
9158 errorText_ = "RtApiOss::probeDeviceOpen: device ID is invalid!";
9162 oss_audioinfo ainfo;
9164 result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
9166 if ( result == -1 ) {
9167 errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
9168 errorText_ = errorStream_.str();
9172 // Check if device supports input or output
9173 if ( ( mode == OUTPUT && !( ainfo.caps & PCM_CAP_OUTPUT ) ) ||
9174 ( mode == INPUT && !( ainfo.caps & PCM_CAP_INPUT ) ) ) {
9175 if ( mode == OUTPUT )
9176 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support output.";
9178 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support input.";
9179 errorText_ = errorStream_.str();
9184 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9185 if ( mode == OUTPUT )
9187 else { // mode == INPUT
9188 if (stream_.mode == OUTPUT && stream_.device[0] == device) {
9189 // We just set the same device for playback ... close and reopen for duplex (OSS only).
9190 close( handle->id[0] );
9192 if ( !( ainfo.caps & PCM_CAP_DUPLEX ) ) {
9193 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support duplex mode.";
9194 errorText_ = errorStream_.str();
9197 // Check that the number previously set channels is the same.
9198 if ( stream_.nUserChannels[0] != channels ) {
9199 errorStream_ << "RtApiOss::probeDeviceOpen: input/output channels must be equal for OSS duplex device (" << ainfo.name << ").";
9200 errorText_ = errorStream_.str();
9209 // Set exclusive access if specified.
9210 if ( options && options->flags & RTAUDIO_HOG_DEVICE ) flags |= O_EXCL;
9212 // Try to open the device.
9214 fd = open( ainfo.devnode, flags, 0 );
9216 if ( errno == EBUSY )
9217 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") is busy.";
9219 errorStream_ << "RtApiOss::probeDeviceOpen: error opening device (" << ainfo.name << ").";
9220 errorText_ = errorStream_.str();
9224 // For duplex operation, specifically set this mode (this doesn't seem to work).
9226 if ( flags | O_RDWR ) {
9227 result = ioctl( fd, SNDCTL_DSP_SETDUPLEX, NULL );
9228 if ( result == -1) {
9229 errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ").";
9230 errorText_ = errorStream_.str();
9236 // Check the device channel support.
9237 stream_.nUserChannels[mode] = channels;
9238 if ( ainfo.max_channels < (int)(channels + firstChannel) ) {
9240 errorStream_ << "RtApiOss::probeDeviceOpen: the device (" << ainfo.name << ") does not support requested channel parameters.";
9241 errorText_ = errorStream_.str();
9245 // Set the number of channels.
9246 int deviceChannels = channels + firstChannel;
9247 result = ioctl( fd, SNDCTL_DSP_CHANNELS, &deviceChannels );
9248 if ( result == -1 || deviceChannels < (int)(channels + firstChannel) ) {
9250 errorStream_ << "RtApiOss::probeDeviceOpen: error setting channel parameters on device (" << ainfo.name << ").";
9251 errorText_ = errorStream_.str();
9254 stream_.nDeviceChannels[mode] = deviceChannels;
9256 // Get the data format mask
9258 result = ioctl( fd, SNDCTL_DSP_GETFMTS, &mask );
9259 if ( result == -1 ) {
9261 errorStream_ << "RtApiOss::probeDeviceOpen: error getting device (" << ainfo.name << ") data formats.";
9262 errorText_ = errorStream_.str();
9266 // Determine how to set the device format.
9267 stream_.userFormat = format;
9268 int deviceFormat = -1;
9269 stream_.doByteSwap[mode] = false;
9270 if ( format == RTAUDIO_SINT8 ) {
9271 if ( mask & AFMT_S8 ) {
9272 deviceFormat = AFMT_S8;
9273 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
9276 else if ( format == RTAUDIO_SINT16 ) {
9277 if ( mask & AFMT_S16_NE ) {
9278 deviceFormat = AFMT_S16_NE;
9279 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
9281 else if ( mask & AFMT_S16_OE ) {
9282 deviceFormat = AFMT_S16_OE;
9283 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
9284 stream_.doByteSwap[mode] = true;
9287 else if ( format == RTAUDIO_SINT24 ) {
9288 if ( mask & AFMT_S24_NE ) {
9289 deviceFormat = AFMT_S24_NE;
9290 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
9292 else if ( mask & AFMT_S24_OE ) {
9293 deviceFormat = AFMT_S24_OE;
9294 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
9295 stream_.doByteSwap[mode] = true;
9298 else if ( format == RTAUDIO_SINT32 ) {
9299 if ( mask & AFMT_S32_NE ) {
9300 deviceFormat = AFMT_S32_NE;
9301 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
9303 else if ( mask & AFMT_S32_OE ) {
9304 deviceFormat = AFMT_S32_OE;
9305 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
9306 stream_.doByteSwap[mode] = true;
9310 if ( deviceFormat == -1 ) {
9311 // The user requested format is not natively supported by the device.
9312 if ( mask & AFMT_S16_NE ) {
9313 deviceFormat = AFMT_S16_NE;
9314 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
9316 else if ( mask & AFMT_S32_NE ) {
9317 deviceFormat = AFMT_S32_NE;
9318 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
9320 else if ( mask & AFMT_S24_NE ) {
9321 deviceFormat = AFMT_S24_NE;
9322 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
9324 else if ( mask & AFMT_S16_OE ) {
9325 deviceFormat = AFMT_S16_OE;
9326 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
9327 stream_.doByteSwap[mode] = true;
9329 else if ( mask & AFMT_S32_OE ) {
9330 deviceFormat = AFMT_S32_OE;
9331 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
9332 stream_.doByteSwap[mode] = true;
9334 else if ( mask & AFMT_S24_OE ) {
9335 deviceFormat = AFMT_S24_OE;
9336 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
9337 stream_.doByteSwap[mode] = true;
9339 else if ( mask & AFMT_S8) {
9340 deviceFormat = AFMT_S8;
9341 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
9345 if ( stream_.deviceFormat[mode] == 0 ) {
9346 // This really shouldn't happen ...
9348 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") data format not supported by RtAudio.";
9349 errorText_ = errorStream_.str();
9353 // Set the data format.
9354 int temp = deviceFormat;
9355 result = ioctl( fd, SNDCTL_DSP_SETFMT, &deviceFormat );
9356 if ( result == -1 || deviceFormat != temp ) {
9358 errorStream_ << "RtApiOss::probeDeviceOpen: error setting data format on device (" << ainfo.name << ").";
9359 errorText_ = errorStream_.str();
9363 // Attempt to set the buffer size. According to OSS, the minimum
9364 // number of buffers is two. The supposed minimum buffer size is 16
9365 // bytes, so that will be our lower bound. The argument to this
9366 // call is in the form 0xMMMMSSSS (hex), where the buffer size (in
9367 // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM.
9368 // We'll check the actual value used near the end of the setup
9370 int ossBufferBytes = *bufferSize * formatBytes( stream_.deviceFormat[mode] ) * deviceChannels;
9371 if ( ossBufferBytes < 16 ) ossBufferBytes = 16;
9373 if ( options ) buffers = options->numberOfBuffers;
9374 if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) buffers = 2;
9375 if ( buffers < 2 ) buffers = 3;
9376 temp = ((int) buffers << 16) + (int)( log10( (double)ossBufferBytes ) / log10( 2.0 ) );
9377 result = ioctl( fd, SNDCTL_DSP_SETFRAGMENT, &temp );
9378 if ( result == -1 ) {
9380 errorStream_ << "RtApiOss::probeDeviceOpen: error setting buffer size on device (" << ainfo.name << ").";
9381 errorText_ = errorStream_.str();
9384 stream_.nBuffers = buffers;
9386 // Save buffer size (in sample frames).
9387 *bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels );
9388 stream_.bufferSize = *bufferSize;
9390 // Set the sample rate.
9391 int srate = sampleRate;
9392 result = ioctl( fd, SNDCTL_DSP_SPEED, &srate );
9393 if ( result == -1 ) {
9395 errorStream_ << "RtApiOss::probeDeviceOpen: error setting sample rate (" << sampleRate << ") on device (" << ainfo.name << ").";
9396 errorText_ = errorStream_.str();
9400 // Verify the sample rate setup worked.
9401 if ( abs( srate - (int)sampleRate ) > 100 ) {
9403 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ").";
9404 errorText_ = errorStream_.str();
9407 stream_.sampleRate = sampleRate;
9409 if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device) {
9410 // We're doing duplex setup here.
9411 stream_.deviceFormat[0] = stream_.deviceFormat[1];
9412 stream_.nDeviceChannels[0] = deviceChannels;
9415 // Set interleaving parameters.
9416 stream_.userInterleaved = true;
9417 stream_.deviceInterleaved[mode] = true;
9418 if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
9419 stream_.userInterleaved = false;
9421 // Set flags for buffer conversion
9422 stream_.doConvertBuffer[mode] = false;
9423 if ( stream_.userFormat != stream_.deviceFormat[mode] )
9424 stream_.doConvertBuffer[mode] = true;
9425 if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
9426 stream_.doConvertBuffer[mode] = true;
9427 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
9428 stream_.nUserChannels[mode] > 1 )
9429 stream_.doConvertBuffer[mode] = true;
9431 // Allocate the stream handles if necessary and then save.
9432 if ( stream_.apiHandle == 0 ) {
9434 handle = new OssHandle;
9436 catch ( std::bad_alloc& ) {
9437 errorText_ = "RtApiOss::probeDeviceOpen: error allocating OssHandle memory.";
9441 if ( pthread_cond_init( &handle->runnable, NULL ) ) {
9442 errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable.";
9446 stream_.apiHandle = (void *) handle;
9449 handle = (OssHandle *) stream_.apiHandle;
9451 handle->id[mode] = fd;
9453 // Allocate necessary internal buffers.
9454 unsigned long bufferBytes;
9455 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
9456 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
9457 if ( stream_.userBuffer[mode] == NULL ) {
9458 errorText_ = "RtApiOss::probeDeviceOpen: error allocating user buffer memory.";
9462 if ( stream_.doConvertBuffer[mode] ) {
9464 bool makeBuffer = true;
9465 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
9466 if ( mode == INPUT ) {
9467 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
9468 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
9469 if ( bufferBytes <= bytesOut ) makeBuffer = false;
9474 bufferBytes *= *bufferSize;
9475 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
9476 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
9477 if ( stream_.deviceBuffer == NULL ) {
9478 errorText_ = "RtApiOss::probeDeviceOpen: error allocating device buffer memory.";
9484 stream_.device[mode] = device;
9485 stream_.state = STREAM_STOPPED;
9487 // Setup the buffer conversion information structure.
9488 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
9490 // Setup thread if necessary.
9491 if ( stream_.mode == OUTPUT && mode == INPUT ) {
9492 // We had already set up an output stream.
9493 stream_.mode = DUPLEX;
9494 if ( stream_.device[0] == device ) handle->id[0] = fd;
9497 stream_.mode = mode;
9499 // Setup callback thread.
9500 stream_.callbackInfo.object = (void *) this;
9502 // Set the thread attributes for joinable and realtime scheduling
9503 // priority. The higher priority will only take affect if the
9504 // program is run as root or suid.
9505 pthread_attr_t attr;
9506 pthread_attr_init( &attr );
9507 pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
9508 #ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
9509 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
9510 stream_.callbackInfo.doRealtime = true;
9511 struct sched_param param;
9512 int priority = options->priority;
9513 int min = sched_get_priority_min( SCHED_RR );
9514 int max = sched_get_priority_max( SCHED_RR );
9515 if ( priority < min ) priority = min;
9516 else if ( priority > max ) priority = max;
9517 param.sched_priority = priority;
9519 // Set the policy BEFORE the priority. Otherwise it fails.
9520 pthread_attr_setschedpolicy(&attr, SCHED_RR);
9521 pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM);
9522 // This is definitely required. Otherwise it fails.
9523 pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED);
9524 pthread_attr_setschedparam(&attr, ¶m);
9527 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
9529 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
9532 stream_.callbackInfo.isRunning = true;
9533 result = pthread_create( &stream_.callbackInfo.thread, &attr, ossCallbackHandler, &stream_.callbackInfo );
9534 pthread_attr_destroy( &attr );
9536 // Failed. Try instead with default attributes.
9537 result = pthread_create( &stream_.callbackInfo.thread, NULL, ossCallbackHandler, &stream_.callbackInfo );
9539 stream_.callbackInfo.isRunning = false;
9540 errorText_ = "RtApiOss::error creating callback thread!";
9550 pthread_cond_destroy( &handle->runnable );
9551 if ( handle->id[0] ) close( handle->id[0] );
9552 if ( handle->id[1] ) close( handle->id[1] );
9554 stream_.apiHandle = 0;
9557 for ( int i=0; i<2; i++ ) {
9558 if ( stream_.userBuffer[i] ) {
9559 free( stream_.userBuffer[i] );
9560 stream_.userBuffer[i] = 0;
9564 if ( stream_.deviceBuffer ) {
9565 free( stream_.deviceBuffer );
9566 stream_.deviceBuffer = 0;
9569 stream_.state = STREAM_CLOSED;
9573 void RtApiOss :: closeStream()
9575 if ( stream_.state == STREAM_CLOSED ) {
9576 errorText_ = "RtApiOss::closeStream(): no open stream to close!";
9577 error( RtAudioError::WARNING );
9581 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9582 stream_.callbackInfo.isRunning = false;
9583 MUTEX_LOCK( &stream_.mutex );
9584 if ( stream_.state == STREAM_STOPPED )
9585 pthread_cond_signal( &handle->runnable );
9586 MUTEX_UNLOCK( &stream_.mutex );
9587 pthread_join( stream_.callbackInfo.thread, NULL );
9589 if ( stream_.state == STREAM_RUNNING ) {
9590 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
9591 ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
9593 ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
9594 stream_.state = STREAM_STOPPED;
9598 pthread_cond_destroy( &handle->runnable );
9599 if ( handle->id[0] ) close( handle->id[0] );
9600 if ( handle->id[1] ) close( handle->id[1] );
9602 stream_.apiHandle = 0;
9605 for ( int i=0; i<2; i++ ) {
9606 if ( stream_.userBuffer[i] ) {
9607 free( stream_.userBuffer[i] );
9608 stream_.userBuffer[i] = 0;
9612 if ( stream_.deviceBuffer ) {
9613 free( stream_.deviceBuffer );
9614 stream_.deviceBuffer = 0;
9617 stream_.mode = UNINITIALIZED;
9618 stream_.state = STREAM_CLOSED;
9621 void RtApiOss :: startStream()
9624 if ( stream_.state == STREAM_RUNNING ) {
9625 errorText_ = "RtApiOss::startStream(): the stream is already running!";
9626 error( RtAudioError::WARNING );
9630 MUTEX_LOCK( &stream_.mutex );
9632 stream_.state = STREAM_RUNNING;
9634 // No need to do anything else here ... OSS automatically starts
9635 // when fed samples.
9637 MUTEX_UNLOCK( &stream_.mutex );
9639 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9640 pthread_cond_signal( &handle->runnable );
9643 void RtApiOss :: stopStream()
9646 if ( stream_.state == STREAM_STOPPED ) {
9647 errorText_ = "RtApiOss::stopStream(): the stream is already stopped!";
9648 error( RtAudioError::WARNING );
9652 MUTEX_LOCK( &stream_.mutex );
9654 // The state might change while waiting on a mutex.
9655 if ( stream_.state == STREAM_STOPPED ) {
9656 MUTEX_UNLOCK( &stream_.mutex );
9661 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9662 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
9664 // Flush the output with zeros a few times.
9667 RtAudioFormat format;
9669 if ( stream_.doConvertBuffer[0] ) {
9670 buffer = stream_.deviceBuffer;
9671 samples = stream_.bufferSize * stream_.nDeviceChannels[0];
9672 format = stream_.deviceFormat[0];
9675 buffer = stream_.userBuffer[0];
9676 samples = stream_.bufferSize * stream_.nUserChannels[0];
9677 format = stream_.userFormat;
9680 memset( buffer, 0, samples * formatBytes(format) );
9681 for ( unsigned int i=0; i<stream_.nBuffers+1; i++ ) {
9682 result = write( handle->id[0], buffer, samples * formatBytes(format) );
9683 if ( result == -1 ) {
9684 errorText_ = "RtApiOss::stopStream: audio write error.";
9685 error( RtAudioError::WARNING );
9689 result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
9690 if ( result == -1 ) {
9691 errorStream_ << "RtApiOss::stopStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
9692 errorText_ = errorStream_.str();
9695 handle->triggered = false;
9698 if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
9699 result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
9700 if ( result == -1 ) {
9701 errorStream_ << "RtApiOss::stopStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
9702 errorText_ = errorStream_.str();
9708 stream_.state = STREAM_STOPPED;
9709 MUTEX_UNLOCK( &stream_.mutex );
9711 if ( result != -1 ) return;
9712 error( RtAudioError::SYSTEM_ERROR );
9715 void RtApiOss :: abortStream()
9718 if ( stream_.state == STREAM_STOPPED ) {
9719 errorText_ = "RtApiOss::abortStream(): the stream is already stopped!";
9720 error( RtAudioError::WARNING );
9724 MUTEX_LOCK( &stream_.mutex );
9726 // The state might change while waiting on a mutex.
9727 if ( stream_.state == STREAM_STOPPED ) {
9728 MUTEX_UNLOCK( &stream_.mutex );
9733 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9734 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
9735 result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
9736 if ( result == -1 ) {
9737 errorStream_ << "RtApiOss::abortStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
9738 errorText_ = errorStream_.str();
9741 handle->triggered = false;
9744 if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
9745 result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
9746 if ( result == -1 ) {
9747 errorStream_ << "RtApiOss::abortStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
9748 errorText_ = errorStream_.str();
9754 stream_.state = STREAM_STOPPED;
9755 MUTEX_UNLOCK( &stream_.mutex );
9757 if ( result != -1 ) return;
9758 error( RtAudioError::SYSTEM_ERROR );
9761 void RtApiOss :: callbackEvent()
9763 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9764 if ( stream_.state == STREAM_STOPPED ) {
9765 MUTEX_LOCK( &stream_.mutex );
9766 pthread_cond_wait( &handle->runnable, &stream_.mutex );
9767 if ( stream_.state != STREAM_RUNNING ) {
9768 MUTEX_UNLOCK( &stream_.mutex );
9771 MUTEX_UNLOCK( &stream_.mutex );
9774 if ( stream_.state == STREAM_CLOSED ) {
9775 errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!";
9776 error( RtAudioError::WARNING );
9780 // Invoke user callback to get fresh output data.
9781 int doStopStream = 0;
9782 RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
9783 double streamTime = getStreamTime();
9784 RtAudioStreamStatus status = 0;
9785 if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
9786 status |= RTAUDIO_OUTPUT_UNDERFLOW;
9787 handle->xrun[0] = false;
9789 if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
9790 status |= RTAUDIO_INPUT_OVERFLOW;
9791 handle->xrun[1] = false;
9793 doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
9794 stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
9795 if ( doStopStream == 2 ) {
9796 this->abortStream();
9800 MUTEX_LOCK( &stream_.mutex );
9802 // The state might change while waiting on a mutex.
9803 if ( stream_.state == STREAM_STOPPED ) goto unlock;
9808 RtAudioFormat format;
9810 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
9812 // Setup parameters and do buffer conversion if necessary.
9813 if ( stream_.doConvertBuffer[0] ) {
9814 buffer = stream_.deviceBuffer;
9815 convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
9816 samples = stream_.bufferSize * stream_.nDeviceChannels[0];
9817 format = stream_.deviceFormat[0];
9820 buffer = stream_.userBuffer[0];
9821 samples = stream_.bufferSize * stream_.nUserChannels[0];
9822 format = stream_.userFormat;
9825 // Do byte swapping if necessary.
9826 if ( stream_.doByteSwap[0] )
9827 byteSwapBuffer( buffer, samples, format );
9829 if ( stream_.mode == DUPLEX && handle->triggered == false ) {
9831 ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
9832 result = write( handle->id[0], buffer, samples * formatBytes(format) );
9833 trig = PCM_ENABLE_INPUT|PCM_ENABLE_OUTPUT;
9834 ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
9835 handle->triggered = true;
9838 // Write samples to device.
9839 result = write( handle->id[0], buffer, samples * formatBytes(format) );
9841 if ( result == -1 ) {
9842 // We'll assume this is an underrun, though there isn't a
9843 // specific means for determining that.
9844 handle->xrun[0] = true;
9845 errorText_ = "RtApiOss::callbackEvent: audio write error.";
9846 error( RtAudioError::WARNING );
9847 // Continue on to input section.
9851 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
9853 // Setup parameters.
9854 if ( stream_.doConvertBuffer[1] ) {
9855 buffer = stream_.deviceBuffer;
9856 samples = stream_.bufferSize * stream_.nDeviceChannels[1];
9857 format = stream_.deviceFormat[1];
9860 buffer = stream_.userBuffer[1];
9861 samples = stream_.bufferSize * stream_.nUserChannels[1];
9862 format = stream_.userFormat;
9865 // Read samples from device.
9866 result = read( handle->id[1], buffer, samples * formatBytes(format) );
9868 if ( result == -1 ) {
9869 // We'll assume this is an overrun, though there isn't a
9870 // specific means for determining that.
9871 handle->xrun[1] = true;
9872 errorText_ = "RtApiOss::callbackEvent: audio read error.";
9873 error( RtAudioError::WARNING );
9877 // Do byte swapping if necessary.
9878 if ( stream_.doByteSwap[1] )
9879 byteSwapBuffer( buffer, samples, format );
9881 // Do buffer conversion if necessary.
9882 if ( stream_.doConvertBuffer[1] )
9883 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
9887 MUTEX_UNLOCK( &stream_.mutex );
9889 RtApi::tickStreamTime();
9890 if ( doStopStream == 1 ) this->stopStream();
9893 static void *ossCallbackHandler( void *ptr )
9895 CallbackInfo *info = (CallbackInfo *) ptr;
9896 RtApiOss *object = (RtApiOss *) info->object;
9897 bool *isRunning = &info->isRunning;
9899 #ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
9900 if (info->doRealtime) {
9901 std::cerr << "RtAudio oss: " <<
9902 (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") <<
9903 "running realtime scheduling" << std::endl;
9907 while ( *isRunning == true ) {
9908 pthread_testcancel();
9909 object->callbackEvent();
9912 pthread_exit( NULL );
9915 //******************** End of __LINUX_OSS__ *********************//
9919 // *************************************************** //
9921 // Protected common (OS-independent) RtAudio methods.
9923 // *************************************************** //
9925 // This method can be modified to control the behavior of error
9926 // message printing.
9927 void RtApi :: error( RtAudioError::Type type )
9929 errorStream_.str(""); // clear the ostringstream
9931 RtAudioErrorCallback errorCallback = (RtAudioErrorCallback) stream_.callbackInfo.errorCallback;
9932 if ( errorCallback ) {
9933 // abortStream() can generate new error messages. Ignore them. Just keep original one.
9935 if ( firstErrorOccurred_ )
9938 firstErrorOccurred_ = true;
9939 const std::string errorMessage = errorText_;
9941 if ( type != RtAudioError::WARNING && stream_.state != STREAM_STOPPED) {
9942 stream_.callbackInfo.isRunning = false; // exit from the thread
9946 errorCallback( type, errorMessage );
9947 firstErrorOccurred_ = false;
9951 if ( type == RtAudioError::WARNING && showWarnings_ == true )
9952 std::cerr << '\n' << errorText_ << "\n\n";
9953 else if ( type != RtAudioError::WARNING )
9954 throw( RtAudioError( errorText_, type ) );
9957 void RtApi :: verifyStream()
9959 if ( stream_.state == STREAM_CLOSED ) {
9960 errorText_ = "RtApi:: a stream is not open!";
9961 error( RtAudioError::INVALID_USE );
9965 void RtApi :: clearStreamInfo()
9967 stream_.mode = UNINITIALIZED;
9968 stream_.state = STREAM_CLOSED;
9969 stream_.sampleRate = 0;
9970 stream_.bufferSize = 0;
9971 stream_.nBuffers = 0;
9972 stream_.userFormat = 0;
9973 stream_.userInterleaved = true;
9974 stream_.streamTime = 0.0;
9975 stream_.apiHandle = 0;
9976 stream_.deviceBuffer = 0;
9977 stream_.callbackInfo.callback = 0;
9978 stream_.callbackInfo.userData = 0;
9979 stream_.callbackInfo.isRunning = false;
9980 stream_.callbackInfo.errorCallback = 0;
9981 for ( int i=0; i<2; i++ ) {
9982 stream_.device[i] = 11111;
9983 stream_.doConvertBuffer[i] = false;
9984 stream_.deviceInterleaved[i] = true;
9985 stream_.doByteSwap[i] = false;
9986 stream_.nUserChannels[i] = 0;
9987 stream_.nDeviceChannels[i] = 0;
9988 stream_.channelOffset[i] = 0;
9989 stream_.deviceFormat[i] = 0;
9990 stream_.latency[i] = 0;
9991 stream_.userBuffer[i] = 0;
9992 stream_.convertInfo[i].channels = 0;
9993 stream_.convertInfo[i].inJump = 0;
9994 stream_.convertInfo[i].outJump = 0;
9995 stream_.convertInfo[i].inFormat = 0;
9996 stream_.convertInfo[i].outFormat = 0;
9997 stream_.convertInfo[i].inOffset.clear();
9998 stream_.convertInfo[i].outOffset.clear();
10002 unsigned int RtApi :: formatBytes( RtAudioFormat format )
10004 if ( format == RTAUDIO_SINT16 )
10006 else if ( format == RTAUDIO_SINT32 || format == RTAUDIO_FLOAT32 )
10008 else if ( format == RTAUDIO_FLOAT64 )
10010 else if ( format == RTAUDIO_SINT24 )
10012 else if ( format == RTAUDIO_SINT8 )
10015 errorText_ = "RtApi::formatBytes: undefined format.";
10016 error( RtAudioError::WARNING );
10021 void RtApi :: setConvertInfo( StreamMode mode, unsigned int firstChannel )
10023 if ( mode == INPUT ) { // convert device to user buffer
10024 stream_.convertInfo[mode].inJump = stream_.nDeviceChannels[1];
10025 stream_.convertInfo[mode].outJump = stream_.nUserChannels[1];
10026 stream_.convertInfo[mode].inFormat = stream_.deviceFormat[1];
10027 stream_.convertInfo[mode].outFormat = stream_.userFormat;
10029 else { // convert user to device buffer
10030 stream_.convertInfo[mode].inJump = stream_.nUserChannels[0];
10031 stream_.convertInfo[mode].outJump = stream_.nDeviceChannels[0];
10032 stream_.convertInfo[mode].inFormat = stream_.userFormat;
10033 stream_.convertInfo[mode].outFormat = stream_.deviceFormat[0];
10036 if ( stream_.convertInfo[mode].inJump < stream_.convertInfo[mode].outJump )
10037 stream_.convertInfo[mode].channels = stream_.convertInfo[mode].inJump;
10039 stream_.convertInfo[mode].channels = stream_.convertInfo[mode].outJump;
10041 // Set up the interleave/deinterleave offsets.
10042 if ( stream_.deviceInterleaved[mode] != stream_.userInterleaved ) {
10043 if ( ( mode == OUTPUT && stream_.deviceInterleaved[mode] ) ||
10044 ( mode == INPUT && stream_.userInterleaved ) ) {
10045 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
10046 stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
10047 stream_.convertInfo[mode].outOffset.push_back( k );
10048 stream_.convertInfo[mode].inJump = 1;
10052 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
10053 stream_.convertInfo[mode].inOffset.push_back( k );
10054 stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
10055 stream_.convertInfo[mode].outJump = 1;
10059 else { // no (de)interleaving
10060 if ( stream_.userInterleaved ) {
10061 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
10062 stream_.convertInfo[mode].inOffset.push_back( k );
10063 stream_.convertInfo[mode].outOffset.push_back( k );
10067 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
10068 stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
10069 stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
10070 stream_.convertInfo[mode].inJump = 1;
10071 stream_.convertInfo[mode].outJump = 1;
10076 // Add channel offset.
10077 if ( firstChannel > 0 ) {
10078 if ( stream_.deviceInterleaved[mode] ) {
10079 if ( mode == OUTPUT ) {
10080 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
10081 stream_.convertInfo[mode].outOffset[k] += firstChannel;
10084 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
10085 stream_.convertInfo[mode].inOffset[k] += firstChannel;
10089 if ( mode == OUTPUT ) {
10090 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
10091 stream_.convertInfo[mode].outOffset[k] += ( firstChannel * stream_.bufferSize );
10094 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
10095 stream_.convertInfo[mode].inOffset[k] += ( firstChannel * stream_.bufferSize );
10101 void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info )
10103 // This function does format conversion, input/output channel compensation, and
10104 // data interleaving/deinterleaving. 24-bit integers are assumed to occupy
10105 // the lower three bytes of a 32-bit integer.
10107 // Clear our device buffer when in/out duplex device channels are different
10108 if ( outBuffer == stream_.deviceBuffer && stream_.mode == DUPLEX &&
10109 ( stream_.nDeviceChannels[0] < stream_.nDeviceChannels[1] ) )
10110 memset( outBuffer, 0, stream_.bufferSize * info.outJump * formatBytes( info.outFormat ) );
10113 if (info.outFormat == RTAUDIO_FLOAT64) {
10115 Float64 *out = (Float64 *)outBuffer;
10117 if (info.inFormat == RTAUDIO_SINT8) {
10118 signed char *in = (signed char *)inBuffer;
10119 scale = 1.0 / 127.5;
10120 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10121 for (j=0; j<info.channels; j++) {
10122 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
10123 out[info.outOffset[j]] += 0.5;
10124 out[info.outOffset[j]] *= scale;
10127 out += info.outJump;
10130 else if (info.inFormat == RTAUDIO_SINT16) {
10131 Int16 *in = (Int16 *)inBuffer;
10132 scale = 1.0 / 32767.5;
10133 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10134 for (j=0; j<info.channels; j++) {
10135 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
10136 out[info.outOffset[j]] += 0.5;
10137 out[info.outOffset[j]] *= scale;
10140 out += info.outJump;
10143 else if (info.inFormat == RTAUDIO_SINT24) {
10144 Int24 *in = (Int24 *)inBuffer;
10145 scale = 1.0 / 8388607.5;
10146 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10147 for (j=0; j<info.channels; j++) {
10148 out[info.outOffset[j]] = (Float64) (in[info.inOffset[j]].asInt());
10149 out[info.outOffset[j]] += 0.5;
10150 out[info.outOffset[j]] *= scale;
10153 out += info.outJump;
10156 else if (info.inFormat == RTAUDIO_SINT32) {
10157 Int32 *in = (Int32 *)inBuffer;
10158 scale = 1.0 / 2147483647.5;
10159 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10160 for (j=0; j<info.channels; j++) {
10161 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
10162 out[info.outOffset[j]] += 0.5;
10163 out[info.outOffset[j]] *= scale;
10166 out += info.outJump;
10169 else if (info.inFormat == RTAUDIO_FLOAT32) {
10170 Float32 *in = (Float32 *)inBuffer;
10171 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10172 for (j=0; j<info.channels; j++) {
10173 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
10176 out += info.outJump;
10179 else if (info.inFormat == RTAUDIO_FLOAT64) {
10180 // Channel compensation and/or (de)interleaving only.
10181 Float64 *in = (Float64 *)inBuffer;
10182 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10183 for (j=0; j<info.channels; j++) {
10184 out[info.outOffset[j]] = in[info.inOffset[j]];
10187 out += info.outJump;
10191 else if (info.outFormat == RTAUDIO_FLOAT32) {
10193 Float32 *out = (Float32 *)outBuffer;
10195 if (info.inFormat == RTAUDIO_SINT8) {
10196 signed char *in = (signed char *)inBuffer;
10197 scale = (Float32) ( 1.0 / 127.5 );
10198 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10199 for (j=0; j<info.channels; j++) {
10200 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
10201 out[info.outOffset[j]] += 0.5;
10202 out[info.outOffset[j]] *= scale;
10205 out += info.outJump;
10208 else if (info.inFormat == RTAUDIO_SINT16) {
10209 Int16 *in = (Int16 *)inBuffer;
10210 scale = (Float32) ( 1.0 / 32767.5 );
10211 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10212 for (j=0; j<info.channels; j++) {
10213 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
10214 out[info.outOffset[j]] += 0.5;
10215 out[info.outOffset[j]] *= scale;
10218 out += info.outJump;
10221 else if (info.inFormat == RTAUDIO_SINT24) {
10222 Int24 *in = (Int24 *)inBuffer;
10223 scale = (Float32) ( 1.0 / 8388607.5 );
10224 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10225 for (j=0; j<info.channels; j++) {
10226 out[info.outOffset[j]] = (Float32) (in[info.inOffset[j]].asInt());
10227 out[info.outOffset[j]] += 0.5;
10228 out[info.outOffset[j]] *= scale;
10231 out += info.outJump;
10234 else if (info.inFormat == RTAUDIO_SINT32) {
10235 Int32 *in = (Int32 *)inBuffer;
10236 scale = (Float32) ( 1.0 / 2147483647.5 );
10237 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10238 for (j=0; j<info.channels; j++) {
10239 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
10240 out[info.outOffset[j]] += 0.5;
10241 out[info.outOffset[j]] *= scale;
10244 out += info.outJump;
10247 else if (info.inFormat == RTAUDIO_FLOAT32) {
10248 // Channel compensation and/or (de)interleaving only.
10249 Float32 *in = (Float32 *)inBuffer;
10250 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10251 for (j=0; j<info.channels; j++) {
10252 out[info.outOffset[j]] = in[info.inOffset[j]];
10255 out += info.outJump;
10258 else if (info.inFormat == RTAUDIO_FLOAT64) {
10259 Float64 *in = (Float64 *)inBuffer;
10260 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10261 for (j=0; j<info.channels; j++) {
10262 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
10265 out += info.outJump;
10269 else if (info.outFormat == RTAUDIO_SINT32) {
10270 Int32 *out = (Int32 *)outBuffer;
10271 if (info.inFormat == RTAUDIO_SINT8) {
10272 signed char *in = (signed char *)inBuffer;
10273 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10274 for (j=0; j<info.channels; j++) {
10275 out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
10276 out[info.outOffset[j]] <<= 24;
10279 out += info.outJump;
10282 else if (info.inFormat == RTAUDIO_SINT16) {
10283 Int16 *in = (Int16 *)inBuffer;
10284 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10285 for (j=0; j<info.channels; j++) {
10286 out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
10287 out[info.outOffset[j]] <<= 16;
10290 out += info.outJump;
10293 else if (info.inFormat == RTAUDIO_SINT24) {
10294 Int24 *in = (Int24 *)inBuffer;
10295 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10296 for (j=0; j<info.channels; j++) {
10297 out[info.outOffset[j]] = (Int32) in[info.inOffset[j]].asInt();
10298 out[info.outOffset[j]] <<= 8;
10301 out += info.outJump;
10304 else if (info.inFormat == RTAUDIO_SINT32) {
10305 // Channel compensation and/or (de)interleaving only.
10306 Int32 *in = (Int32 *)inBuffer;
10307 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10308 for (j=0; j<info.channels; j++) {
10309 out[info.outOffset[j]] = in[info.inOffset[j]];
10312 out += info.outJump;
10315 else if (info.inFormat == RTAUDIO_FLOAT32) {
10316 Float32 *in = (Float32 *)inBuffer;
10317 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10318 for (j=0; j<info.channels; j++) {
10319 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
10322 out += info.outJump;
10325 else if (info.inFormat == RTAUDIO_FLOAT64) {
10326 Float64 *in = (Float64 *)inBuffer;
10327 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10328 for (j=0; j<info.channels; j++) {
10329 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
10332 out += info.outJump;
10336 else if (info.outFormat == RTAUDIO_SINT24) {
10337 Int24 *out = (Int24 *)outBuffer;
10338 if (info.inFormat == RTAUDIO_SINT8) {
10339 signed char *in = (signed char *)inBuffer;
10340 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10341 for (j=0; j<info.channels; j++) {
10342 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] << 16);
10343 //out[info.outOffset[j]] <<= 16;
10346 out += info.outJump;
10349 else if (info.inFormat == RTAUDIO_SINT16) {
10350 Int16 *in = (Int16 *)inBuffer;
10351 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10352 for (j=0; j<info.channels; j++) {
10353 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] << 8);
10354 //out[info.outOffset[j]] <<= 8;
10357 out += info.outJump;
10360 else if (info.inFormat == RTAUDIO_SINT24) {
10361 // Channel compensation and/or (de)interleaving only.
10362 Int24 *in = (Int24 *)inBuffer;
10363 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10364 for (j=0; j<info.channels; j++) {
10365 out[info.outOffset[j]] = in[info.inOffset[j]];
10368 out += info.outJump;
10371 else if (info.inFormat == RTAUDIO_SINT32) {
10372 Int32 *in = (Int32 *)inBuffer;
10373 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10374 for (j=0; j<info.channels; j++) {
10375 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] >> 8);
10376 //out[info.outOffset[j]] >>= 8;
10379 out += info.outJump;
10382 else if (info.inFormat == RTAUDIO_FLOAT32) {
10383 Float32 *in = (Float32 *)inBuffer;
10384 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10385 for (j=0; j<info.channels; j++) {
10386 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
10389 out += info.outJump;
10392 else if (info.inFormat == RTAUDIO_FLOAT64) {
10393 Float64 *in = (Float64 *)inBuffer;
10394 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10395 for (j=0; j<info.channels; j++) {
10396 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
10399 out += info.outJump;
10403 else if (info.outFormat == RTAUDIO_SINT16) {
10404 Int16 *out = (Int16 *)outBuffer;
10405 if (info.inFormat == RTAUDIO_SINT8) {
10406 signed char *in = (signed char *)inBuffer;
10407 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10408 for (j=0; j<info.channels; j++) {
10409 out[info.outOffset[j]] = (Int16) in[info.inOffset[j]];
10410 out[info.outOffset[j]] <<= 8;
10413 out += info.outJump;
10416 else if (info.inFormat == RTAUDIO_SINT16) {
10417 // Channel compensation and/or (de)interleaving only.
10418 Int16 *in = (Int16 *)inBuffer;
10419 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10420 for (j=0; j<info.channels; j++) {
10421 out[info.outOffset[j]] = in[info.inOffset[j]];
10424 out += info.outJump;
10427 else if (info.inFormat == RTAUDIO_SINT24) {
10428 Int24 *in = (Int24 *)inBuffer;
10429 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10430 for (j=0; j<info.channels; j++) {
10431 out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]].asInt() >> 8);
10434 out += info.outJump;
10437 else if (info.inFormat == RTAUDIO_SINT32) {
10438 Int32 *in = (Int32 *)inBuffer;
10439 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10440 for (j=0; j<info.channels; j++) {
10441 out[info.outOffset[j]] = (Int16) ((in[info.inOffset[j]] >> 16) & 0x0000ffff);
10444 out += info.outJump;
10447 else if (info.inFormat == RTAUDIO_FLOAT32) {
10448 Float32 *in = (Float32 *)inBuffer;
10449 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10450 for (j=0; j<info.channels; j++) {
10451 out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
10454 out += info.outJump;
10457 else if (info.inFormat == RTAUDIO_FLOAT64) {
10458 Float64 *in = (Float64 *)inBuffer;
10459 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10460 for (j=0; j<info.channels; j++) {
10461 out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
10464 out += info.outJump;
10468 else if (info.outFormat == RTAUDIO_SINT8) {
10469 signed char *out = (signed char *)outBuffer;
10470 if (info.inFormat == RTAUDIO_SINT8) {
10471 // Channel compensation and/or (de)interleaving only.
10472 signed char *in = (signed char *)inBuffer;
10473 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10474 for (j=0; j<info.channels; j++) {
10475 out[info.outOffset[j]] = in[info.inOffset[j]];
10478 out += info.outJump;
10481 if (info.inFormat == RTAUDIO_SINT16) {
10482 Int16 *in = (Int16 *)inBuffer;
10483 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10484 for (j=0; j<info.channels; j++) {
10485 out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 8) & 0x00ff);
10488 out += info.outJump;
10491 else if (info.inFormat == RTAUDIO_SINT24) {
10492 Int24 *in = (Int24 *)inBuffer;
10493 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10494 for (j=0; j<info.channels; j++) {
10495 out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]].asInt() >> 16);
10498 out += info.outJump;
10501 else if (info.inFormat == RTAUDIO_SINT32) {
10502 Int32 *in = (Int32 *)inBuffer;
10503 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10504 for (j=0; j<info.channels; j++) {
10505 out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 24) & 0x000000ff);
10508 out += info.outJump;
10511 else if (info.inFormat == RTAUDIO_FLOAT32) {
10512 Float32 *in = (Float32 *)inBuffer;
10513 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10514 for (j=0; j<info.channels; j++) {
10515 out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
10518 out += info.outJump;
10521 else if (info.inFormat == RTAUDIO_FLOAT64) {
10522 Float64 *in = (Float64 *)inBuffer;
10523 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10524 for (j=0; j<info.channels; j++) {
10525 out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
10528 out += info.outJump;
10534 //static inline uint16_t bswap_16(uint16_t x) { return (x>>8) | (x<<8); }
10535 //static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); }
10536 //static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); }
10538 void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format )
10544 if ( format == RTAUDIO_SINT16 ) {
10545 for ( unsigned int i=0; i<samples; i++ ) {
10546 // Swap 1st and 2nd bytes.
10551 // Increment 2 bytes.
10555 else if ( format == RTAUDIO_SINT32 ||
10556 format == RTAUDIO_FLOAT32 ) {
10557 for ( unsigned int i=0; i<samples; i++ ) {
10558 // Swap 1st and 4th bytes.
10563 // Swap 2nd and 3rd bytes.
10569 // Increment 3 more bytes.
10573 else if ( format == RTAUDIO_SINT24 ) {
10574 for ( unsigned int i=0; i<samples; i++ ) {
10575 // Swap 1st and 3rd bytes.
10580 // Increment 2 more bytes.
10584 else if ( format == RTAUDIO_FLOAT64 ) {
10585 for ( unsigned int i=0; i<samples; i++ ) {
10586 // Swap 1st and 8th bytes
10591 // Swap 2nd and 7th bytes
10597 // Swap 3rd and 6th bytes
10603 // Swap 4th and 5th bytes
10609 // Increment 5 more bytes.
10615 // Indentation settings for Vim and Emacs
10617 // Local Variables:
10618 // c-basic-offset: 2
10619 // indent-tabs-mode: nil
10622 // vim: et sts=2 sw=2