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 // determine whether index falls within capture or render 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 device audio client.";
4754 hr = captureAudioClient->GetMixFormat( &deviceFormat );
4755 if ( FAILED( hr ) ) {
4756 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format.";
4760 stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
4761 captureAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
4764 if ( mode != OUTPUT ) {
4765 errorType = RtAudioError::INVALID_USE;
4766 errorText_ = "RtApiWasapi::probeDeviceOpen: Render device selected as input device.";
4770 // retrieve renderAudioClient from devicePtr
4771 IAudioClient*& renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
4773 hr = renderDevices->Item( device, &devicePtr );
4774 if ( FAILED( hr ) ) {
4775 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device handle.";
4779 hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
4780 NULL, ( void** ) &renderAudioClient );
4781 if ( FAILED( hr ) ) {
4782 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client.";
4786 hr = renderAudioClient->GetMixFormat( &deviceFormat );
4787 if ( FAILED( hr ) ) {
4788 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format.";
4792 stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
4793 renderAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
4797 if ( ( stream_.mode == OUTPUT && mode == INPUT ) ||
4798 ( stream_.mode == INPUT && mode == OUTPUT ) ) {
4799 stream_.mode = DUPLEX;
4802 stream_.mode = mode;
4805 stream_.device[mode] = device;
4806 stream_.doByteSwap[mode] = false;
4807 stream_.sampleRate = sampleRate;
4808 stream_.bufferSize = *bufferSize;
4809 stream_.nBuffers = 1;
4810 stream_.nUserChannels[mode] = channels;
4811 stream_.channelOffset[mode] = firstChannel;
4812 stream_.userFormat = format;
4813 stream_.deviceFormat[mode] = getDeviceInfo( device ).nativeFormats;
4815 if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
4816 stream_.userInterleaved = false;
4818 stream_.userInterleaved = true;
4819 stream_.deviceInterleaved[mode] = true;
4821 // Set flags for buffer conversion.
4822 stream_.doConvertBuffer[mode] = false;
4823 if ( stream_.userFormat != stream_.deviceFormat[mode] ||
4824 stream_.nUserChannels[0] != stream_.nDeviceChannels[0] ||
4825 stream_.nUserChannels[1] != stream_.nDeviceChannels[1] )
4826 stream_.doConvertBuffer[mode] = true;
4827 else if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
4828 stream_.nUserChannels[mode] > 1 )
4829 stream_.doConvertBuffer[mode] = true;
4831 if ( stream_.doConvertBuffer[mode] )
4832 setConvertInfo( mode, 0 );
4834 // Allocate necessary internal buffers
4835 bufferBytes = stream_.nUserChannels[mode] * stream_.bufferSize * formatBytes( stream_.userFormat );
4837 stream_.userBuffer[mode] = ( char* ) calloc( bufferBytes, 1 );
4838 if ( !stream_.userBuffer[mode] ) {
4839 errorType = RtAudioError::MEMORY_ERROR;
4840 errorText_ = "RtApiWasapi::probeDeviceOpen: Error allocating user buffer memory.";
4844 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME )
4845 stream_.callbackInfo.priority = 15;
4847 stream_.callbackInfo.priority = 0;
4849 ///! TODO: RTAUDIO_MINIMIZE_LATENCY // Provide stream buffers directly to callback
4850 ///! TODO: RTAUDIO_HOG_DEVICE // Exclusive mode
4852 methodResult = SUCCESS;
4856 SAFE_RELEASE( captureDevices );
4857 SAFE_RELEASE( renderDevices );
4858 SAFE_RELEASE( devicePtr );
4859 CoTaskMemFree( deviceFormat );
4861 // if method failed, close the stream
4862 if ( methodResult == FAILURE )
4865 if ( !errorText_.empty() )
4867 return methodResult;
4870 //=============================================================================
4872 DWORD WINAPI RtApiWasapi::runWasapiThread( void* wasapiPtr )
4875 ( ( RtApiWasapi* ) wasapiPtr )->wasapiThread();
4880 DWORD WINAPI RtApiWasapi::stopWasapiThread( void* wasapiPtr )
4883 ( ( RtApiWasapi* ) wasapiPtr )->stopStream();
4888 DWORD WINAPI RtApiWasapi::abortWasapiThread( void* wasapiPtr )
4891 ( ( RtApiWasapi* ) wasapiPtr )->abortStream();
4896 //-----------------------------------------------------------------------------
4898 void RtApiWasapi::wasapiThread()
4900 // as this is a new thread, we must CoInitialize it
4901 CoInitialize( NULL );
4905 IAudioClient* captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
4906 IAudioClient* renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
4907 IAudioCaptureClient* captureClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureClient;
4908 IAudioRenderClient* renderClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderClient;
4909 HANDLE captureEvent = ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent;
4910 HANDLE renderEvent = ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent;
4912 WAVEFORMATEX* captureFormat = NULL;
4913 WAVEFORMATEX* renderFormat = NULL;
4914 float captureSrRatio = 0.0f;
4915 float renderSrRatio = 0.0f;
4916 WasapiBuffer captureBuffer;
4917 WasapiBuffer renderBuffer;
4918 WasapiResampler* captureResampler = NULL;
4919 WasapiResampler* renderResampler = NULL;
4921 // declare local stream variables
4922 RtAudioCallback callback = ( RtAudioCallback ) stream_.callbackInfo.callback;
4923 BYTE* streamBuffer = NULL;
4924 unsigned long captureFlags = 0;
4925 unsigned int bufferFrameCount = 0;
4926 unsigned int numFramesPadding = 0;
4927 unsigned int convBufferSize = 0;
4928 bool callbackPushed = true;
4929 bool callbackPulled = false;
4930 bool callbackStopped = false;
4931 int callbackResult = 0;
4933 // convBuffer is used to store converted buffers between WASAPI and the user
4934 char* convBuffer = NULL;
4935 unsigned int convBuffSize = 0;
4936 unsigned int deviceBuffSize = 0;
4939 RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
4941 // Attempt to assign "Pro Audio" characteristic to thread
4942 HMODULE AvrtDll = LoadLibrary( (LPCTSTR) "AVRT.dll" );
4944 DWORD taskIndex = 0;
4945 TAvSetMmThreadCharacteristicsPtr AvSetMmThreadCharacteristicsPtr = ( TAvSetMmThreadCharacteristicsPtr ) GetProcAddress( AvrtDll, "AvSetMmThreadCharacteristicsW" );
4946 AvSetMmThreadCharacteristicsPtr( L"Pro Audio", &taskIndex );
4947 FreeLibrary( AvrtDll );
4950 // start capture stream if applicable
4951 if ( captureAudioClient ) {
4952 hr = captureAudioClient->GetMixFormat( &captureFormat );
4953 if ( FAILED( hr ) ) {
4954 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
4958 // init captureResampler
4959 captureResampler = new WasapiResampler( stream_.deviceFormat[INPUT] == RTAUDIO_FLOAT32 || stream_.deviceFormat[INPUT] == RTAUDIO_FLOAT64,
4960 formatBytes( stream_.deviceFormat[INPUT] ) * 8, stream_.nDeviceChannels[INPUT],
4961 captureFormat->nSamplesPerSec, stream_.sampleRate );
4963 captureSrRatio = ( ( float ) captureFormat->nSamplesPerSec / stream_.sampleRate );
4965 // initialize capture stream according to desire buffer size
4966 float desiredBufferSize = stream_.bufferSize * captureSrRatio;
4967 REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / captureFormat->nSamplesPerSec );
4969 if ( !captureClient ) {
4970 hr = captureAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
4971 AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
4972 desiredBufferPeriod,
4973 desiredBufferPeriod,
4976 if ( FAILED( hr ) ) {
4977 errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize capture audio client.";
4981 hr = captureAudioClient->GetService( __uuidof( IAudioCaptureClient ),
4982 ( void** ) &captureClient );
4983 if ( FAILED( hr ) ) {
4984 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture client handle.";
4988 // configure captureEvent to trigger on every available capture buffer
4989 captureEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
4990 if ( !captureEvent ) {
4991 errorType = RtAudioError::SYSTEM_ERROR;
4992 errorText_ = "RtApiWasapi::wasapiThread: Unable to create capture event.";
4996 hr = captureAudioClient->SetEventHandle( captureEvent );
4997 if ( FAILED( hr ) ) {
4998 errorText_ = "RtApiWasapi::wasapiThread: Unable to set capture event handle.";
5002 ( ( WasapiHandle* ) stream_.apiHandle )->captureClient = captureClient;
5003 ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent = captureEvent;
5006 unsigned int inBufferSize = 0;
5007 hr = captureAudioClient->GetBufferSize( &inBufferSize );
5008 if ( FAILED( hr ) ) {
5009 errorText_ = "RtApiWasapi::wasapiThread: Unable to get capture buffer size.";
5013 // scale outBufferSize according to stream->user sample rate ratio
5014 unsigned int outBufferSize = ( unsigned int ) ceilf( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT];
5015 inBufferSize *= stream_.nDeviceChannels[INPUT];
5017 // set captureBuffer size
5018 captureBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[INPUT] ) );
5020 // reset the capture stream
5021 hr = captureAudioClient->Reset();
5022 if ( FAILED( hr ) ) {
5023 errorText_ = "RtApiWasapi::wasapiThread: Unable to reset capture stream.";
5027 // start the capture stream
5028 hr = captureAudioClient->Start();
5029 if ( FAILED( hr ) ) {
5030 errorText_ = "RtApiWasapi::wasapiThread: Unable to start capture stream.";
5035 // start render stream if applicable
5036 if ( renderAudioClient ) {
5037 hr = renderAudioClient->GetMixFormat( &renderFormat );
5038 if ( FAILED( hr ) ) {
5039 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
5043 // init renderResampler
5044 renderResampler = new WasapiResampler( stream_.deviceFormat[OUTPUT] == RTAUDIO_FLOAT32 || stream_.deviceFormat[OUTPUT] == RTAUDIO_FLOAT64,
5045 formatBytes( stream_.deviceFormat[OUTPUT] ) * 8, stream_.nDeviceChannels[OUTPUT],
5046 stream_.sampleRate, renderFormat->nSamplesPerSec );
5048 renderSrRatio = ( ( float ) renderFormat->nSamplesPerSec / stream_.sampleRate );
5050 // initialize render stream according to desire buffer size
5051 float desiredBufferSize = stream_.bufferSize * renderSrRatio;
5052 REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / renderFormat->nSamplesPerSec );
5054 if ( !renderClient ) {
5055 hr = renderAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
5056 AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
5057 desiredBufferPeriod,
5058 desiredBufferPeriod,
5061 if ( FAILED( hr ) ) {
5062 errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize render audio client.";
5066 hr = renderAudioClient->GetService( __uuidof( IAudioRenderClient ),
5067 ( void** ) &renderClient );
5068 if ( FAILED( hr ) ) {
5069 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render client handle.";
5073 // configure renderEvent to trigger on every available render buffer
5074 renderEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
5075 if ( !renderEvent ) {
5076 errorType = RtAudioError::SYSTEM_ERROR;
5077 errorText_ = "RtApiWasapi::wasapiThread: Unable to create render event.";
5081 hr = renderAudioClient->SetEventHandle( renderEvent );
5082 if ( FAILED( hr ) ) {
5083 errorText_ = "RtApiWasapi::wasapiThread: Unable to set render event handle.";
5087 ( ( WasapiHandle* ) stream_.apiHandle )->renderClient = renderClient;
5088 ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent = renderEvent;
5091 unsigned int outBufferSize = 0;
5092 hr = renderAudioClient->GetBufferSize( &outBufferSize );
5093 if ( FAILED( hr ) ) {
5094 errorText_ = "RtApiWasapi::wasapiThread: Unable to get render buffer size.";
5098 // scale inBufferSize according to user->stream sample rate ratio
5099 unsigned int inBufferSize = ( unsigned int ) ceilf( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT];
5100 outBufferSize *= stream_.nDeviceChannels[OUTPUT];
5102 // set renderBuffer size
5103 renderBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[OUTPUT] ) );
5105 // reset the render stream
5106 hr = renderAudioClient->Reset();
5107 if ( FAILED( hr ) ) {
5108 errorText_ = "RtApiWasapi::wasapiThread: Unable to reset render stream.";
5112 // start the render stream
5113 hr = renderAudioClient->Start();
5114 if ( FAILED( hr ) ) {
5115 errorText_ = "RtApiWasapi::wasapiThread: Unable to start render stream.";
5120 // malloc buffer memory
5121 if ( stream_.mode == INPUT )
5123 using namespace std; // for ceilf
5124 convBuffSize = ( size_t ) ( ceilf( stream_.bufferSize * captureSrRatio ) ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
5125 deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
5127 else if ( stream_.mode == OUTPUT )
5129 convBuffSize = ( size_t ) ( ceilf( stream_.bufferSize * renderSrRatio ) ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
5130 deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
5132 else if ( stream_.mode == DUPLEX )
5134 convBuffSize = std::max( ( size_t ) ( ceilf( stream_.bufferSize * captureSrRatio ) ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
5135 ( size_t ) ( ceilf( stream_.bufferSize * renderSrRatio ) ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
5136 deviceBuffSize = std::max( stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
5137 stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
5140 convBuffSize *= 2; // allow overflow for *SrRatio remainders
5141 convBuffer = ( char* ) malloc( convBuffSize );
5142 stream_.deviceBuffer = ( char* ) malloc( deviceBuffSize );
5143 if ( !convBuffer || !stream_.deviceBuffer ) {
5144 errorType = RtAudioError::MEMORY_ERROR;
5145 errorText_ = "RtApiWasapi::wasapiThread: Error allocating device buffer memory.";
5149 // stream process loop
5150 while ( stream_.state != STREAM_STOPPING ) {
5151 if ( !callbackPulled ) {
5154 // 1. Pull callback buffer from inputBuffer
5155 // 2. If 1. was successful: Convert callback buffer to user sample rate and channel count
5156 // Convert callback buffer to user format
5158 if ( captureAudioClient )
5160 int samplesToPull = ( unsigned int ) floorf( stream_.bufferSize * captureSrRatio );
5161 if ( captureSrRatio != 1 )
5163 // account for remainders
5168 while ( convBufferSize < stream_.bufferSize )
5170 // Pull callback buffer from inputBuffer
5171 callbackPulled = captureBuffer.pullBuffer( convBuffer,
5172 samplesToPull * stream_.nDeviceChannels[INPUT],
5173 stream_.deviceFormat[INPUT] );
5175 if ( !callbackPulled )
5180 // Convert callback buffer to user sample rate
5181 unsigned int deviceBufferOffset = convBufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
5182 unsigned int convSamples = 0;
5184 captureResampler->Convert( stream_.deviceBuffer + deviceBufferOffset,
5189 convBufferSize += convSamples;
5190 samplesToPull = 1; // now pull one sample at a time until we have stream_.bufferSize samples
5193 if ( callbackPulled )
5195 if ( stream_.doConvertBuffer[INPUT] ) {
5196 // Convert callback buffer to user format
5197 convertBuffer( stream_.userBuffer[INPUT],
5198 stream_.deviceBuffer,
5199 stream_.convertInfo[INPUT] );
5202 // no further conversion, simple copy deviceBuffer to userBuffer
5203 memcpy( stream_.userBuffer[INPUT],
5204 stream_.deviceBuffer,
5205 stream_.bufferSize * stream_.nUserChannels[INPUT] * formatBytes( stream_.userFormat ) );
5210 // if there is no capture stream, set callbackPulled flag
5211 callbackPulled = true;
5216 // 1. Execute user callback method
5217 // 2. Handle return value from callback
5219 // if callback has not requested the stream to stop
5220 if ( callbackPulled && !callbackStopped ) {
5221 // Execute user callback method
5222 callbackResult = callback( stream_.userBuffer[OUTPUT],
5223 stream_.userBuffer[INPUT],
5226 captureFlags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY ? RTAUDIO_INPUT_OVERFLOW : 0,
5227 stream_.callbackInfo.userData );
5229 // Handle return value from callback
5230 if ( callbackResult == 1 ) {
5231 // instantiate a thread to stop this thread
5232 HANDLE threadHandle = CreateThread( NULL, 0, stopWasapiThread, this, 0, NULL );
5233 if ( !threadHandle ) {
5234 errorType = RtAudioError::THREAD_ERROR;
5235 errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream stop thread.";
5238 else if ( !CloseHandle( threadHandle ) ) {
5239 errorType = RtAudioError::THREAD_ERROR;
5240 errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream stop thread handle.";
5244 callbackStopped = true;
5246 else if ( callbackResult == 2 ) {
5247 // instantiate a thread to stop this thread
5248 HANDLE threadHandle = CreateThread( NULL, 0, abortWasapiThread, this, 0, NULL );
5249 if ( !threadHandle ) {
5250 errorType = RtAudioError::THREAD_ERROR;
5251 errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream abort thread.";
5254 else if ( !CloseHandle( threadHandle ) ) {
5255 errorType = RtAudioError::THREAD_ERROR;
5256 errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream abort thread handle.";
5260 callbackStopped = true;
5267 // 1. Convert callback buffer to stream format
5268 // 2. Convert callback buffer to stream sample rate and channel count
5269 // 3. Push callback buffer into outputBuffer
5271 if ( renderAudioClient && callbackPulled )
5273 // if the last call to renderBuffer.PushBuffer() was successful
5274 if ( callbackPushed || convBufferSize == 0 )
5276 if ( stream_.doConvertBuffer[OUTPUT] )
5278 // Convert callback buffer to stream format
5279 convertBuffer( stream_.deviceBuffer,
5280 stream_.userBuffer[OUTPUT],
5281 stream_.convertInfo[OUTPUT] );
5285 // Convert callback buffer to stream sample rate
5286 renderResampler->Convert( convBuffer,
5287 stream_.deviceBuffer,
5292 // Push callback buffer into outputBuffer
5293 callbackPushed = renderBuffer.pushBuffer( convBuffer,
5294 convBufferSize * stream_.nDeviceChannels[OUTPUT],
5295 stream_.deviceFormat[OUTPUT] );
5298 // if there is no render stream, set callbackPushed flag
5299 callbackPushed = true;
5304 // 1. Get capture buffer from stream
5305 // 2. Push capture buffer into inputBuffer
5306 // 3. If 2. was successful: Release capture buffer
5308 if ( captureAudioClient ) {
5309 // if the callback input buffer was not pulled from captureBuffer, wait for next capture event
5310 if ( !callbackPulled ) {
5311 WaitForSingleObject( captureEvent, INFINITE );
5314 // Get capture buffer from stream
5315 hr = captureClient->GetBuffer( &streamBuffer,
5317 &captureFlags, NULL, NULL );
5318 if ( FAILED( hr ) ) {
5319 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture buffer.";
5323 if ( bufferFrameCount != 0 ) {
5324 // Push capture buffer into inputBuffer
5325 if ( captureBuffer.pushBuffer( ( char* ) streamBuffer,
5326 bufferFrameCount * stream_.nDeviceChannels[INPUT],
5327 stream_.deviceFormat[INPUT] ) )
5329 // Release capture buffer
5330 hr = captureClient->ReleaseBuffer( bufferFrameCount );
5331 if ( FAILED( hr ) ) {
5332 errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
5338 // Inform WASAPI that capture was unsuccessful
5339 hr = captureClient->ReleaseBuffer( 0 );
5340 if ( FAILED( hr ) ) {
5341 errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
5348 // Inform WASAPI that capture was unsuccessful
5349 hr = captureClient->ReleaseBuffer( 0 );
5350 if ( FAILED( hr ) ) {
5351 errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
5359 // 1. Get render buffer from stream
5360 // 2. Pull next buffer from outputBuffer
5361 // 3. If 2. was successful: Fill render buffer with next buffer
5362 // Release render buffer
5364 if ( renderAudioClient ) {
5365 // if the callback output buffer was not pushed to renderBuffer, wait for next render event
5366 if ( callbackPulled && !callbackPushed ) {
5367 WaitForSingleObject( renderEvent, INFINITE );
5370 // Get render buffer from stream
5371 hr = renderAudioClient->GetBufferSize( &bufferFrameCount );
5372 if ( FAILED( hr ) ) {
5373 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer size.";
5377 hr = renderAudioClient->GetCurrentPadding( &numFramesPadding );
5378 if ( FAILED( hr ) ) {
5379 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer padding.";
5383 bufferFrameCount -= numFramesPadding;
5385 if ( bufferFrameCount != 0 ) {
5386 hr = renderClient->GetBuffer( bufferFrameCount, &streamBuffer );
5387 if ( FAILED( hr ) ) {
5388 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer.";
5392 // Pull next buffer from outputBuffer
5393 // Fill render buffer with next buffer
5394 if ( renderBuffer.pullBuffer( ( char* ) streamBuffer,
5395 bufferFrameCount * stream_.nDeviceChannels[OUTPUT],
5396 stream_.deviceFormat[OUTPUT] ) )
5398 // Release render buffer
5399 hr = renderClient->ReleaseBuffer( bufferFrameCount, 0 );
5400 if ( FAILED( hr ) ) {
5401 errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
5407 // Inform WASAPI that render was unsuccessful
5408 hr = renderClient->ReleaseBuffer( 0, 0 );
5409 if ( FAILED( hr ) ) {
5410 errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
5417 // Inform WASAPI that render was unsuccessful
5418 hr = renderClient->ReleaseBuffer( 0, 0 );
5419 if ( FAILED( hr ) ) {
5420 errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
5426 // if the callback buffer was pushed renderBuffer reset callbackPulled flag
5427 if ( callbackPushed ) {
5428 // unsetting the callbackPulled flag lets the stream know that
5429 // the audio device is ready for another callback output buffer.
5430 callbackPulled = false;
5433 RtApi::tickStreamTime();
5440 CoTaskMemFree( captureFormat );
5441 CoTaskMemFree( renderFormat );
5443 free ( convBuffer );
5444 delete renderResampler;
5445 delete captureResampler;
5449 if ( !errorText_.empty() )
5452 // update stream state
5453 stream_.state = STREAM_STOPPED;
5456 //******************** End of __WINDOWS_WASAPI__ *********************//
5460 #if defined(__WINDOWS_DS__) // Windows DirectSound API
5462 // Modified by Robin Davies, October 2005
5463 // - Improvements to DirectX pointer chasing.
5464 // - Bug fix for non-power-of-two Asio granularity used by Edirol PCR-A30.
5465 // - Auto-call CoInitialize for DSOUND and ASIO platforms.
5466 // Various revisions for RtAudio 4.0 by Gary Scavone, April 2007
5467 // Changed device query structure for RtAudio 4.0.7, January 2010
5469 #include <windows.h>
5470 #include <process.h>
5471 #include <mmsystem.h>
5475 #include <algorithm>
5477 #if defined(__MINGW32__)
5478 // missing from latest mingw winapi
5479 #define WAVE_FORMAT_96M08 0x00010000 /* 96 kHz, Mono, 8-bit */
5480 #define WAVE_FORMAT_96S08 0x00020000 /* 96 kHz, Stereo, 8-bit */
5481 #define WAVE_FORMAT_96M16 0x00040000 /* 96 kHz, Mono, 16-bit */
5482 #define WAVE_FORMAT_96S16 0x00080000 /* 96 kHz, Stereo, 16-bit */
5485 #define MINIMUM_DEVICE_BUFFER_SIZE 32768
5487 #ifdef _MSC_VER // if Microsoft Visual C++
5488 #pragma comment( lib, "winmm.lib" ) // then, auto-link winmm.lib. Otherwise, it has to be added manually.
5491 static inline DWORD dsPointerBetween( DWORD pointer, DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize )
5493 if ( pointer > bufferSize ) pointer -= bufferSize;
5494 if ( laterPointer < earlierPointer ) laterPointer += bufferSize;
5495 if ( pointer < earlierPointer ) pointer += bufferSize;
5496 return pointer >= earlierPointer && pointer < laterPointer;
5499 // A structure to hold various information related to the DirectSound
5500 // API implementation.
5502 unsigned int drainCounter; // Tracks callback counts when draining
5503 bool internalDrain; // Indicates if stop is initiated from callback or not.
5507 UINT bufferPointer[2];
5508 DWORD dsBufferSize[2];
5509 DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by.
5513 :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; }
5516 // Declarations for utility functions, callbacks, and structures
5517 // specific to the DirectSound implementation.
5518 static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
5519 LPCTSTR description,
5523 static const char* getErrorString( int code );
5525 static unsigned __stdcall callbackHandler( void *ptr );
5534 : found(false) { validId[0] = false; validId[1] = false; }
5537 struct DsProbeData {
5539 std::vector<struct DsDevice>* dsDevices;
5542 RtApiDs :: RtApiDs()
5544 // Dsound will run both-threaded. If CoInitialize fails, then just
5545 // accept whatever the mainline chose for a threading model.
5546 coInitialized_ = false;
5547 HRESULT hr = CoInitialize( NULL );
5548 if ( !FAILED( hr ) ) coInitialized_ = true;
5551 RtApiDs :: ~RtApiDs()
5553 if ( stream_.state != STREAM_CLOSED ) closeStream();
5554 if ( coInitialized_ ) CoUninitialize(); // balanced call.
5557 // The DirectSound default output is always the first device.
5558 unsigned int RtApiDs :: getDefaultOutputDevice( void )
5563 // The DirectSound default input is always the first input device,
5564 // which is the first capture device enumerated.
5565 unsigned int RtApiDs :: getDefaultInputDevice( void )
5570 unsigned int RtApiDs :: getDeviceCount( void )
5572 // Set query flag for previously found devices to false, so that we
5573 // can check for any devices that have disappeared.
5574 for ( unsigned int i=0; i<dsDevices.size(); i++ )
5575 dsDevices[i].found = false;
5577 // Query DirectSound devices.
5578 struct DsProbeData probeInfo;
5579 probeInfo.isInput = false;
5580 probeInfo.dsDevices = &dsDevices;
5581 HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );
5582 if ( FAILED( result ) ) {
5583 errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!";
5584 errorText_ = errorStream_.str();
5585 error( RtAudioError::WARNING );
5588 // Query DirectSoundCapture devices.
5589 probeInfo.isInput = true;
5590 result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );
5591 if ( FAILED( result ) ) {
5592 errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!";
5593 errorText_ = errorStream_.str();
5594 error( RtAudioError::WARNING );
5597 // Clean out any devices that may have disappeared (code update submitted by Eli Zehngut).
5598 for ( unsigned int i=0; i<dsDevices.size(); ) {
5599 if ( dsDevices[i].found == false ) dsDevices.erase( dsDevices.begin() + i );
5603 return static_cast<unsigned int>(dsDevices.size());
5606 RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device )
5608 RtAudio::DeviceInfo info;
5609 info.probed = false;
5611 if ( dsDevices.size() == 0 ) {
5612 // Force a query of all devices
5614 if ( dsDevices.size() == 0 ) {
5615 errorText_ = "RtApiDs::getDeviceInfo: no devices found!";
5616 error( RtAudioError::INVALID_USE );
5621 if ( device >= dsDevices.size() ) {
5622 errorText_ = "RtApiDs::getDeviceInfo: device ID is invalid!";
5623 error( RtAudioError::INVALID_USE );
5628 if ( dsDevices[ device ].validId[0] == false ) goto probeInput;
5630 LPDIRECTSOUND output;
5632 result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );
5633 if ( FAILED( result ) ) {
5634 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!";
5635 errorText_ = errorStream_.str();
5636 error( RtAudioError::WARNING );
5640 outCaps.dwSize = sizeof( outCaps );
5641 result = output->GetCaps( &outCaps );
5642 if ( FAILED( result ) ) {
5644 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting capabilities!";
5645 errorText_ = errorStream_.str();
5646 error( RtAudioError::WARNING );
5650 // Get output channel information.
5651 info.outputChannels = ( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1;
5653 // Get sample rate information.
5654 info.sampleRates.clear();
5655 for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
5656 if ( SAMPLE_RATES[k] >= (unsigned int) outCaps.dwMinSecondarySampleRate &&
5657 SAMPLE_RATES[k] <= (unsigned int) outCaps.dwMaxSecondarySampleRate ) {
5658 info.sampleRates.push_back( SAMPLE_RATES[k] );
5660 if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
5661 info.preferredSampleRate = SAMPLE_RATES[k];
5665 // Get format information.
5666 if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT ) info.nativeFormats |= RTAUDIO_SINT16;
5667 if ( outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) info.nativeFormats |= RTAUDIO_SINT8;
5671 if ( getDefaultOutputDevice() == device )
5672 info.isDefaultOutput = true;
5674 if ( dsDevices[ device ].validId[1] == false ) {
5675 info.name = dsDevices[ device ].name;
5682 LPDIRECTSOUNDCAPTURE input;
5683 result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );
5684 if ( FAILED( result ) ) {
5685 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";
5686 errorText_ = errorStream_.str();
5687 error( RtAudioError::WARNING );
5692 inCaps.dwSize = sizeof( inCaps );
5693 result = input->GetCaps( &inCaps );
5694 if ( FAILED( result ) ) {
5696 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsDevices[ device ].name << ")!";
5697 errorText_ = errorStream_.str();
5698 error( RtAudioError::WARNING );
5702 // Get input channel information.
5703 info.inputChannels = inCaps.dwChannels;
5705 // Get sample rate and format information.
5706 std::vector<unsigned int> rates;
5707 if ( inCaps.dwChannels >= 2 ) {
5708 if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.nativeFormats |= RTAUDIO_SINT16;
5709 if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.nativeFormats |= RTAUDIO_SINT16;
5710 if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.nativeFormats |= RTAUDIO_SINT16;
5711 if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.nativeFormats |= RTAUDIO_SINT16;
5712 if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.nativeFormats |= RTAUDIO_SINT8;
5713 if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.nativeFormats |= RTAUDIO_SINT8;
5714 if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.nativeFormats |= RTAUDIO_SINT8;
5715 if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.nativeFormats |= RTAUDIO_SINT8;
5717 if ( info.nativeFormats & RTAUDIO_SINT16 ) {
5718 if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) rates.push_back( 11025 );
5719 if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) rates.push_back( 22050 );
5720 if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) rates.push_back( 44100 );
5721 if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) rates.push_back( 96000 );
5723 else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
5724 if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) rates.push_back( 11025 );
5725 if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) rates.push_back( 22050 );
5726 if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) rates.push_back( 44100 );
5727 if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) rates.push_back( 96000 );
5730 else if ( inCaps.dwChannels == 1 ) {
5731 if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.nativeFormats |= RTAUDIO_SINT16;
5732 if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.nativeFormats |= RTAUDIO_SINT16;
5733 if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.nativeFormats |= RTAUDIO_SINT16;
5734 if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.nativeFormats |= RTAUDIO_SINT16;
5735 if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.nativeFormats |= RTAUDIO_SINT8;
5736 if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.nativeFormats |= RTAUDIO_SINT8;
5737 if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.nativeFormats |= RTAUDIO_SINT8;
5738 if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.nativeFormats |= RTAUDIO_SINT8;
5740 if ( info.nativeFormats & RTAUDIO_SINT16 ) {
5741 if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) rates.push_back( 11025 );
5742 if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) rates.push_back( 22050 );
5743 if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) rates.push_back( 44100 );
5744 if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) rates.push_back( 96000 );
5746 else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
5747 if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) rates.push_back( 11025 );
5748 if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) rates.push_back( 22050 );
5749 if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) rates.push_back( 44100 );
5750 if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) rates.push_back( 96000 );
5753 else info.inputChannels = 0; // technically, this would be an error
5757 if ( info.inputChannels == 0 ) return info;
5759 // Copy the supported rates to the info structure but avoid duplication.
5761 for ( unsigned int i=0; i<rates.size(); i++ ) {
5763 for ( unsigned int j=0; j<info.sampleRates.size(); j++ ) {
5764 if ( rates[i] == info.sampleRates[j] ) {
5769 if ( found == false ) info.sampleRates.push_back( rates[i] );
5771 std::sort( info.sampleRates.begin(), info.sampleRates.end() );
5773 // If device opens for both playback and capture, we determine the channels.
5774 if ( info.outputChannels > 0 && info.inputChannels > 0 )
5775 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
5777 if ( device == 0 ) info.isDefaultInput = true;
5779 // Copy name and return.
5780 info.name = dsDevices[ device ].name;
5785 bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
5786 unsigned int firstChannel, unsigned int sampleRate,
5787 RtAudioFormat format, unsigned int *bufferSize,
5788 RtAudio::StreamOptions *options )
5790 if ( channels + firstChannel > 2 ) {
5791 errorText_ = "RtApiDs::probeDeviceOpen: DirectSound does not support more than 2 channels per device.";
5795 size_t nDevices = dsDevices.size();
5796 if ( nDevices == 0 ) {
5797 // This should not happen because a check is made before this function is called.
5798 errorText_ = "RtApiDs::probeDeviceOpen: no devices found!";
5802 if ( device >= nDevices ) {
5803 // This should not happen because a check is made before this function is called.
5804 errorText_ = "RtApiDs::probeDeviceOpen: device ID is invalid!";
5808 if ( mode == OUTPUT ) {
5809 if ( dsDevices[ device ].validId[0] == false ) {
5810 errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support output!";
5811 errorText_ = errorStream_.str();
5815 else { // mode == INPUT
5816 if ( dsDevices[ device ].validId[1] == false ) {
5817 errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support input!";
5818 errorText_ = errorStream_.str();
5823 // According to a note in PortAudio, using GetDesktopWindow()
5824 // instead of GetForegroundWindow() is supposed to avoid problems
5825 // that occur when the application's window is not the foreground
5826 // window. Also, if the application window closes before the
5827 // DirectSound buffer, DirectSound can crash. In the past, I had
5828 // problems when using GetDesktopWindow() but it seems fine now
5829 // (January 2010). I'll leave it commented here.
5830 // HWND hWnd = GetForegroundWindow();
5831 HWND hWnd = GetDesktopWindow();
5833 // Check the numberOfBuffers parameter and limit the lowest value to
5834 // two. This is a judgement call and a value of two is probably too
5835 // low for capture, but it should work for playback.
5837 if ( options ) nBuffers = options->numberOfBuffers;
5838 if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) nBuffers = 2;
5839 if ( nBuffers < 2 ) nBuffers = 3;
5841 // Check the lower range of the user-specified buffer size and set
5842 // (arbitrarily) to a lower bound of 32.
5843 if ( *bufferSize < 32 ) *bufferSize = 32;
5845 // Create the wave format structure. The data format setting will
5846 // be determined later.
5847 WAVEFORMATEX waveFormat;
5848 ZeroMemory( &waveFormat, sizeof(WAVEFORMATEX) );
5849 waveFormat.wFormatTag = WAVE_FORMAT_PCM;
5850 waveFormat.nChannels = channels + firstChannel;
5851 waveFormat.nSamplesPerSec = (unsigned long) sampleRate;
5853 // Determine the device buffer size. By default, we'll use the value
5854 // defined above (32K), but we will grow it to make allowances for
5855 // very large software buffer sizes.
5856 DWORD dsBufferSize = MINIMUM_DEVICE_BUFFER_SIZE;
5857 DWORD dsPointerLeadTime = 0;
5859 void *ohandle = 0, *bhandle = 0;
5861 if ( mode == OUTPUT ) {
5863 LPDIRECTSOUND output;
5864 result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );
5865 if ( FAILED( result ) ) {
5866 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!";
5867 errorText_ = errorStream_.str();
5872 outCaps.dwSize = sizeof( outCaps );
5873 result = output->GetCaps( &outCaps );
5874 if ( FAILED( result ) ) {
5876 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting capabilities (" << dsDevices[ device ].name << ")!";
5877 errorText_ = errorStream_.str();
5881 // Check channel information.
5882 if ( channels + firstChannel == 2 && !( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ) {
5883 errorStream_ << "RtApiDs::getDeviceInfo: the output device (" << dsDevices[ device ].name << ") does not support stereo playback.";
5884 errorText_ = errorStream_.str();
5888 // Check format information. Use 16-bit format unless not
5889 // supported or user requests 8-bit.
5890 if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT &&
5891 !( format == RTAUDIO_SINT8 && outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) ) {
5892 waveFormat.wBitsPerSample = 16;
5893 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
5896 waveFormat.wBitsPerSample = 8;
5897 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
5899 stream_.userFormat = format;
5901 // Update wave format structure and buffer information.
5902 waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
5903 waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
5904 dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
5906 // If the user wants an even bigger buffer, increase the device buffer size accordingly.
5907 while ( dsPointerLeadTime * 2U > dsBufferSize )
5910 // Set cooperative level to DSSCL_EXCLUSIVE ... sound stops when window focus changes.
5911 // result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE );
5912 // Set cooperative level to DSSCL_PRIORITY ... sound remains when window focus changes.
5913 result = output->SetCooperativeLevel( hWnd, DSSCL_PRIORITY );
5914 if ( FAILED( result ) ) {
5916 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting cooperative level (" << dsDevices[ device ].name << ")!";
5917 errorText_ = errorStream_.str();
5921 // Even though we will write to the secondary buffer, we need to
5922 // access the primary buffer to set the correct output format
5923 // (since the default is 8-bit, 22 kHz!). Setup the DS primary
5924 // buffer description.
5925 DSBUFFERDESC bufferDescription;
5926 ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
5927 bufferDescription.dwSize = sizeof( DSBUFFERDESC );
5928 bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER;
5930 // Obtain the primary buffer
5931 LPDIRECTSOUNDBUFFER buffer;
5932 result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
5933 if ( FAILED( result ) ) {
5935 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") accessing primary buffer (" << dsDevices[ device ].name << ")!";
5936 errorText_ = errorStream_.str();
5940 // Set the primary DS buffer sound format.
5941 result = buffer->SetFormat( &waveFormat );
5942 if ( FAILED( result ) ) {
5944 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting primary buffer format (" << dsDevices[ device ].name << ")!";
5945 errorText_ = errorStream_.str();
5949 // Setup the secondary DS buffer description.
5950 ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
5951 bufferDescription.dwSize = sizeof( DSBUFFERDESC );
5952 bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
5953 DSBCAPS_GLOBALFOCUS |
5954 DSBCAPS_GETCURRENTPOSITION2 |
5955 DSBCAPS_LOCHARDWARE ); // Force hardware mixing
5956 bufferDescription.dwBufferBytes = dsBufferSize;
5957 bufferDescription.lpwfxFormat = &waveFormat;
5959 // Try to create the secondary DS buffer. If that doesn't work,
5960 // try to use software mixing. Otherwise, there's a problem.
5961 result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
5962 if ( FAILED( result ) ) {
5963 bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
5964 DSBCAPS_GLOBALFOCUS |
5965 DSBCAPS_GETCURRENTPOSITION2 |
5966 DSBCAPS_LOCSOFTWARE ); // Force software mixing
5967 result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
5968 if ( FAILED( result ) ) {
5970 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating secondary buffer (" << dsDevices[ device ].name << ")!";
5971 errorText_ = errorStream_.str();
5976 // Get the buffer size ... might be different from what we specified.
5978 dsbcaps.dwSize = sizeof( DSBCAPS );
5979 result = buffer->GetCaps( &dsbcaps );
5980 if ( FAILED( result ) ) {
5983 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";
5984 errorText_ = errorStream_.str();
5988 dsBufferSize = dsbcaps.dwBufferBytes;
5990 // Lock the DS buffer
5993 result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );
5994 if ( FAILED( result ) ) {
5997 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking buffer (" << dsDevices[ device ].name << ")!";
5998 errorText_ = errorStream_.str();
6002 // Zero the DS buffer
6003 ZeroMemory( audioPtr, dataLen );
6005 // Unlock the DS buffer
6006 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
6007 if ( FAILED( result ) ) {
6010 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking buffer (" << dsDevices[ device ].name << ")!";
6011 errorText_ = errorStream_.str();
6015 ohandle = (void *) output;
6016 bhandle = (void *) buffer;
6019 if ( mode == INPUT ) {
6021 LPDIRECTSOUNDCAPTURE input;
6022 result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );
6023 if ( FAILED( result ) ) {
6024 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";
6025 errorText_ = errorStream_.str();
6030 inCaps.dwSize = sizeof( inCaps );
6031 result = input->GetCaps( &inCaps );
6032 if ( FAILED( result ) ) {
6034 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsDevices[ device ].name << ")!";
6035 errorText_ = errorStream_.str();
6039 // Check channel information.
6040 if ( inCaps.dwChannels < channels + firstChannel ) {
6041 errorText_ = "RtApiDs::getDeviceInfo: the input device does not support requested input channels.";
6045 // Check format information. Use 16-bit format unless user
6047 DWORD deviceFormats;
6048 if ( channels + firstChannel == 2 ) {
6049 deviceFormats = WAVE_FORMAT_1S08 | WAVE_FORMAT_2S08 | WAVE_FORMAT_4S08 | WAVE_FORMAT_96S08;
6050 if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
6051 waveFormat.wBitsPerSample = 8;
6052 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
6054 else { // assume 16-bit is supported
6055 waveFormat.wBitsPerSample = 16;
6056 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
6059 else { // channel == 1
6060 deviceFormats = WAVE_FORMAT_1M08 | WAVE_FORMAT_2M08 | WAVE_FORMAT_4M08 | WAVE_FORMAT_96M08;
6061 if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
6062 waveFormat.wBitsPerSample = 8;
6063 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
6065 else { // assume 16-bit is supported
6066 waveFormat.wBitsPerSample = 16;
6067 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
6070 stream_.userFormat = format;
6072 // Update wave format structure and buffer information.
6073 waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
6074 waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
6075 dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
6077 // If the user wants an even bigger buffer, increase the device buffer size accordingly.
6078 while ( dsPointerLeadTime * 2U > dsBufferSize )
6081 // Setup the secondary DS buffer description.
6082 DSCBUFFERDESC bufferDescription;
6083 ZeroMemory( &bufferDescription, sizeof( DSCBUFFERDESC ) );
6084 bufferDescription.dwSize = sizeof( DSCBUFFERDESC );
6085 bufferDescription.dwFlags = 0;
6086 bufferDescription.dwReserved = 0;
6087 bufferDescription.dwBufferBytes = dsBufferSize;
6088 bufferDescription.lpwfxFormat = &waveFormat;
6090 // Create the capture buffer.
6091 LPDIRECTSOUNDCAPTUREBUFFER buffer;
6092 result = input->CreateCaptureBuffer( &bufferDescription, &buffer, NULL );
6093 if ( FAILED( result ) ) {
6095 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating input buffer (" << dsDevices[ device ].name << ")!";
6096 errorText_ = errorStream_.str();
6100 // Get the buffer size ... might be different from what we specified.
6102 dscbcaps.dwSize = sizeof( DSCBCAPS );
6103 result = buffer->GetCaps( &dscbcaps );
6104 if ( FAILED( result ) ) {
6107 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";
6108 errorText_ = errorStream_.str();
6112 dsBufferSize = dscbcaps.dwBufferBytes;
6114 // NOTE: We could have a problem here if this is a duplex stream
6115 // and the play and capture hardware buffer sizes are different
6116 // (I'm actually not sure if that is a problem or not).
6117 // Currently, we are not verifying that.
6119 // Lock the capture buffer
6122 result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );
6123 if ( FAILED( result ) ) {
6126 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking input buffer (" << dsDevices[ device ].name << ")!";
6127 errorText_ = errorStream_.str();
6132 ZeroMemory( audioPtr, dataLen );
6134 // Unlock the buffer
6135 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
6136 if ( FAILED( result ) ) {
6139 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking input buffer (" << dsDevices[ device ].name << ")!";
6140 errorText_ = errorStream_.str();
6144 ohandle = (void *) input;
6145 bhandle = (void *) buffer;
6148 // Set various stream parameters
6149 DsHandle *handle = 0;
6150 stream_.nDeviceChannels[mode] = channels + firstChannel;
6151 stream_.nUserChannels[mode] = channels;
6152 stream_.bufferSize = *bufferSize;
6153 stream_.channelOffset[mode] = firstChannel;
6154 stream_.deviceInterleaved[mode] = true;
6155 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
6156 else stream_.userInterleaved = true;
6158 // Set flag for buffer conversion
6159 stream_.doConvertBuffer[mode] = false;
6160 if (stream_.nUserChannels[mode] != stream_.nDeviceChannels[mode])
6161 stream_.doConvertBuffer[mode] = true;
6162 if (stream_.userFormat != stream_.deviceFormat[mode])
6163 stream_.doConvertBuffer[mode] = true;
6164 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
6165 stream_.nUserChannels[mode] > 1 )
6166 stream_.doConvertBuffer[mode] = true;
6168 // Allocate necessary internal buffers
6169 long bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
6170 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
6171 if ( stream_.userBuffer[mode] == NULL ) {
6172 errorText_ = "RtApiDs::probeDeviceOpen: error allocating user buffer memory.";
6176 if ( stream_.doConvertBuffer[mode] ) {
6178 bool makeBuffer = true;
6179 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
6180 if ( mode == INPUT ) {
6181 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
6182 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
6183 if ( bufferBytes <= (long) bytesOut ) makeBuffer = false;
6188 bufferBytes *= *bufferSize;
6189 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
6190 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
6191 if ( stream_.deviceBuffer == NULL ) {
6192 errorText_ = "RtApiDs::probeDeviceOpen: error allocating device buffer memory.";
6198 // Allocate our DsHandle structures for the stream.
6199 if ( stream_.apiHandle == 0 ) {
6201 handle = new DsHandle;
6203 catch ( std::bad_alloc& ) {
6204 errorText_ = "RtApiDs::probeDeviceOpen: error allocating AsioHandle memory.";
6208 // Create a manual-reset event.
6209 handle->condition = CreateEvent( NULL, // no security
6210 TRUE, // manual-reset
6211 FALSE, // non-signaled initially
6213 stream_.apiHandle = (void *) handle;
6216 handle = (DsHandle *) stream_.apiHandle;
6217 handle->id[mode] = ohandle;
6218 handle->buffer[mode] = bhandle;
6219 handle->dsBufferSize[mode] = dsBufferSize;
6220 handle->dsPointerLeadTime[mode] = dsPointerLeadTime;
6222 stream_.device[mode] = device;
6223 stream_.state = STREAM_STOPPED;
6224 if ( stream_.mode == OUTPUT && mode == INPUT )
6225 // We had already set up an output stream.
6226 stream_.mode = DUPLEX;
6228 stream_.mode = mode;
6229 stream_.nBuffers = nBuffers;
6230 stream_.sampleRate = sampleRate;
6232 // Setup the buffer conversion information structure.
6233 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
6235 // Setup the callback thread.
6236 if ( stream_.callbackInfo.isRunning == false ) {
6238 stream_.callbackInfo.isRunning = true;
6239 stream_.callbackInfo.object = (void *) this;
6240 stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &callbackHandler,
6241 &stream_.callbackInfo, 0, &threadId );
6242 if ( stream_.callbackInfo.thread == 0 ) {
6243 errorText_ = "RtApiDs::probeDeviceOpen: error creating callback thread!";
6247 // Boost DS thread priority
6248 SetThreadPriority( (HANDLE) stream_.callbackInfo.thread, THREAD_PRIORITY_HIGHEST );
6254 if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
6255 LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
6256 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6257 if ( buffer ) buffer->Release();
6260 if ( handle->buffer[1] ) {
6261 LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
6262 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6263 if ( buffer ) buffer->Release();
6266 CloseHandle( handle->condition );
6268 stream_.apiHandle = 0;
6271 for ( int i=0; i<2; i++ ) {
6272 if ( stream_.userBuffer[i] ) {
6273 free( stream_.userBuffer[i] );
6274 stream_.userBuffer[i] = 0;
6278 if ( stream_.deviceBuffer ) {
6279 free( stream_.deviceBuffer );
6280 stream_.deviceBuffer = 0;
6283 stream_.state = STREAM_CLOSED;
6287 void RtApiDs :: closeStream()
6289 if ( stream_.state == STREAM_CLOSED ) {
6290 errorText_ = "RtApiDs::closeStream(): no open stream to close!";
6291 error( RtAudioError::WARNING );
6295 // Stop the callback thread.
6296 stream_.callbackInfo.isRunning = false;
6297 WaitForSingleObject( (HANDLE) stream_.callbackInfo.thread, INFINITE );
6298 CloseHandle( (HANDLE) stream_.callbackInfo.thread );
6300 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6302 if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
6303 LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
6304 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6311 if ( handle->buffer[1] ) {
6312 LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
6313 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6320 CloseHandle( handle->condition );
6322 stream_.apiHandle = 0;
6325 for ( int i=0; i<2; i++ ) {
6326 if ( stream_.userBuffer[i] ) {
6327 free( stream_.userBuffer[i] );
6328 stream_.userBuffer[i] = 0;
6332 if ( stream_.deviceBuffer ) {
6333 free( stream_.deviceBuffer );
6334 stream_.deviceBuffer = 0;
6337 stream_.mode = UNINITIALIZED;
6338 stream_.state = STREAM_CLOSED;
6341 void RtApiDs :: startStream()
6344 if ( stream_.state == STREAM_RUNNING ) {
6345 errorText_ = "RtApiDs::startStream(): the stream is already running!";
6346 error( RtAudioError::WARNING );
6350 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6352 // Increase scheduler frequency on lesser windows (a side-effect of
6353 // increasing timer accuracy). On greater windows (Win2K or later),
6354 // this is already in effect.
6355 timeBeginPeriod( 1 );
6357 buffersRolling = false;
6358 duplexPrerollBytes = 0;
6360 if ( stream_.mode == DUPLEX ) {
6361 // 0.5 seconds of silence in DUPLEX mode while the devices spin up and synchronize.
6362 duplexPrerollBytes = (int) ( 0.5 * stream_.sampleRate * formatBytes( stream_.deviceFormat[1] ) * stream_.nDeviceChannels[1] );
6366 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
6368 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6369 result = buffer->Play( 0, 0, DSBPLAY_LOOPING );
6370 if ( FAILED( result ) ) {
6371 errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting output buffer!";
6372 errorText_ = errorStream_.str();
6377 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
6379 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6380 result = buffer->Start( DSCBSTART_LOOPING );
6381 if ( FAILED( result ) ) {
6382 errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting input buffer!";
6383 errorText_ = errorStream_.str();
6388 handle->drainCounter = 0;
6389 handle->internalDrain = false;
6390 ResetEvent( handle->condition );
6391 stream_.state = STREAM_RUNNING;
6394 if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );
6397 void RtApiDs :: stopStream()
6400 if ( stream_.state == STREAM_STOPPED ) {
6401 errorText_ = "RtApiDs::stopStream(): the stream is already stopped!";
6402 error( RtAudioError::WARNING );
6409 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6410 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
6411 if ( handle->drainCounter == 0 ) {
6412 handle->drainCounter = 2;
6413 WaitForSingleObject( handle->condition, INFINITE ); // block until signaled
6416 stream_.state = STREAM_STOPPED;
6418 MUTEX_LOCK( &stream_.mutex );
6420 // Stop the buffer and clear memory
6421 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6422 result = buffer->Stop();
6423 if ( FAILED( result ) ) {
6424 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping output buffer!";
6425 errorText_ = errorStream_.str();
6429 // Lock the buffer and clear it so that if we start to play again,
6430 // we won't have old data playing.
6431 result = buffer->Lock( 0, handle->dsBufferSize[0], &audioPtr, &dataLen, NULL, NULL, 0 );
6432 if ( FAILED( result ) ) {
6433 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking output buffer!";
6434 errorText_ = errorStream_.str();
6438 // Zero the DS buffer
6439 ZeroMemory( audioPtr, dataLen );
6441 // Unlock the DS buffer
6442 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
6443 if ( FAILED( result ) ) {
6444 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking output buffer!";
6445 errorText_ = errorStream_.str();
6449 // If we start playing again, we must begin at beginning of buffer.
6450 handle->bufferPointer[0] = 0;
6453 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
6454 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6458 stream_.state = STREAM_STOPPED;
6460 if ( stream_.mode != DUPLEX )
6461 MUTEX_LOCK( &stream_.mutex );
6463 result = buffer->Stop();
6464 if ( FAILED( result ) ) {
6465 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping input buffer!";
6466 errorText_ = errorStream_.str();
6470 // Lock the buffer and clear it so that if we start to play again,
6471 // we won't have old data playing.
6472 result = buffer->Lock( 0, handle->dsBufferSize[1], &audioPtr, &dataLen, NULL, NULL, 0 );
6473 if ( FAILED( result ) ) {
6474 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking input buffer!";
6475 errorText_ = errorStream_.str();
6479 // Zero the DS buffer
6480 ZeroMemory( audioPtr, dataLen );
6482 // Unlock the DS buffer
6483 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
6484 if ( FAILED( result ) ) {
6485 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking input buffer!";
6486 errorText_ = errorStream_.str();
6490 // If we start recording again, we must begin at beginning of buffer.
6491 handle->bufferPointer[1] = 0;
6495 timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows.
6496 MUTEX_UNLOCK( &stream_.mutex );
6498 if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );
6501 void RtApiDs :: abortStream()
6504 if ( stream_.state == STREAM_STOPPED ) {
6505 errorText_ = "RtApiDs::abortStream(): the stream is already stopped!";
6506 error( RtAudioError::WARNING );
6510 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6511 handle->drainCounter = 2;
6516 void RtApiDs :: callbackEvent()
6518 if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) {
6519 Sleep( 50 ); // sleep 50 milliseconds
6523 if ( stream_.state == STREAM_CLOSED ) {
6524 errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!";
6525 error( RtAudioError::WARNING );
6529 CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
6530 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6532 // Check if we were draining the stream and signal is finished.
6533 if ( handle->drainCounter > stream_.nBuffers + 2 ) {
6535 stream_.state = STREAM_STOPPING;
6536 if ( handle->internalDrain == false )
6537 SetEvent( handle->condition );
6543 // Invoke user callback to get fresh output data UNLESS we are
6545 if ( handle->drainCounter == 0 ) {
6546 RtAudioCallback callback = (RtAudioCallback) info->callback;
6547 double streamTime = getStreamTime();
6548 RtAudioStreamStatus status = 0;
6549 if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
6550 status |= RTAUDIO_OUTPUT_UNDERFLOW;
6551 handle->xrun[0] = false;
6553 if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
6554 status |= RTAUDIO_INPUT_OVERFLOW;
6555 handle->xrun[1] = false;
6557 int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
6558 stream_.bufferSize, streamTime, status, info->userData );
6559 if ( cbReturnValue == 2 ) {
6560 stream_.state = STREAM_STOPPING;
6561 handle->drainCounter = 2;
6565 else if ( cbReturnValue == 1 ) {
6566 handle->drainCounter = 1;
6567 handle->internalDrain = true;
6572 DWORD currentWritePointer, safeWritePointer;
6573 DWORD currentReadPointer, safeReadPointer;
6574 UINT nextWritePointer;
6576 LPVOID buffer1 = NULL;
6577 LPVOID buffer2 = NULL;
6578 DWORD bufferSize1 = 0;
6579 DWORD bufferSize2 = 0;
6584 MUTEX_LOCK( &stream_.mutex );
6585 if ( stream_.state == STREAM_STOPPED ) {
6586 MUTEX_UNLOCK( &stream_.mutex );
6590 if ( buffersRolling == false ) {
6591 if ( stream_.mode == DUPLEX ) {
6592 //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
6594 // It takes a while for the devices to get rolling. As a result,
6595 // there's no guarantee that the capture and write device pointers
6596 // will move in lockstep. Wait here for both devices to start
6597 // rolling, and then set our buffer pointers accordingly.
6598 // e.g. Crystal Drivers: the capture buffer starts up 5700 to 9600
6599 // bytes later than the write buffer.
6601 // Stub: a serious risk of having a pre-emptive scheduling round
6602 // take place between the two GetCurrentPosition calls... but I'm
6603 // really not sure how to solve the problem. Temporarily boost to
6604 // Realtime priority, maybe; but I'm not sure what priority the
6605 // DirectSound service threads run at. We *should* be roughly
6606 // within a ms or so of correct.
6608 LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6609 LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6611 DWORD startSafeWritePointer, startSafeReadPointer;
6613 result = dsWriteBuffer->GetCurrentPosition( NULL, &startSafeWritePointer );
6614 if ( FAILED( result ) ) {
6615 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
6616 errorText_ = errorStream_.str();
6617 MUTEX_UNLOCK( &stream_.mutex );
6618 error( RtAudioError::SYSTEM_ERROR );
6621 result = dsCaptureBuffer->GetCurrentPosition( NULL, &startSafeReadPointer );
6622 if ( FAILED( result ) ) {
6623 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
6624 errorText_ = errorStream_.str();
6625 MUTEX_UNLOCK( &stream_.mutex );
6626 error( RtAudioError::SYSTEM_ERROR );
6630 result = dsWriteBuffer->GetCurrentPosition( NULL, &safeWritePointer );
6631 if ( FAILED( result ) ) {
6632 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
6633 errorText_ = errorStream_.str();
6634 MUTEX_UNLOCK( &stream_.mutex );
6635 error( RtAudioError::SYSTEM_ERROR );
6638 result = dsCaptureBuffer->GetCurrentPosition( NULL, &safeReadPointer );
6639 if ( FAILED( result ) ) {
6640 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
6641 errorText_ = errorStream_.str();
6642 MUTEX_UNLOCK( &stream_.mutex );
6643 error( RtAudioError::SYSTEM_ERROR );
6646 if ( safeWritePointer != startSafeWritePointer && safeReadPointer != startSafeReadPointer ) break;
6650 //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
6652 handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];
6653 if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];
6654 handle->bufferPointer[1] = safeReadPointer;
6656 else if ( stream_.mode == OUTPUT ) {
6658 // Set the proper nextWritePosition after initial startup.
6659 LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6660 result = dsWriteBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer );
6661 if ( FAILED( result ) ) {
6662 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
6663 errorText_ = errorStream_.str();
6664 MUTEX_UNLOCK( &stream_.mutex );
6665 error( RtAudioError::SYSTEM_ERROR );
6668 handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];
6669 if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];
6672 buffersRolling = true;
6675 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
6677 LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6679 if ( handle->drainCounter > 1 ) { // write zeros to the output stream
6680 bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
6681 bufferBytes *= formatBytes( stream_.userFormat );
6682 memset( stream_.userBuffer[0], 0, bufferBytes );
6685 // Setup parameters and do buffer conversion if necessary.
6686 if ( stream_.doConvertBuffer[0] ) {
6687 buffer = stream_.deviceBuffer;
6688 convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
6689 bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[0];
6690 bufferBytes *= formatBytes( stream_.deviceFormat[0] );
6693 buffer = stream_.userBuffer[0];
6694 bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
6695 bufferBytes *= formatBytes( stream_.userFormat );
6698 // No byte swapping necessary in DirectSound implementation.
6700 // Ahhh ... windoze. 16-bit data is signed but 8-bit data is
6701 // unsigned. So, we need to convert our signed 8-bit data here to
6703 if ( stream_.deviceFormat[0] == RTAUDIO_SINT8 )
6704 for ( int i=0; i<bufferBytes; i++ ) buffer[i] = (unsigned char) ( buffer[i] + 128 );
6706 DWORD dsBufferSize = handle->dsBufferSize[0];
6707 nextWritePointer = handle->bufferPointer[0];
6709 DWORD endWrite, leadPointer;
6711 // Find out where the read and "safe write" pointers are.
6712 result = dsBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer );
6713 if ( FAILED( result ) ) {
6714 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
6715 errorText_ = errorStream_.str();
6716 MUTEX_UNLOCK( &stream_.mutex );
6717 error( RtAudioError::SYSTEM_ERROR );
6721 // We will copy our output buffer into the region between
6722 // safeWritePointer and leadPointer. If leadPointer is not
6723 // beyond the next endWrite position, wait until it is.
6724 leadPointer = safeWritePointer + handle->dsPointerLeadTime[0];
6725 //std::cout << "safeWritePointer = " << safeWritePointer << ", leadPointer = " << leadPointer << ", nextWritePointer = " << nextWritePointer << std::endl;
6726 if ( leadPointer > dsBufferSize ) leadPointer -= dsBufferSize;
6727 if ( leadPointer < nextWritePointer ) leadPointer += dsBufferSize; // unwrap offset
6728 endWrite = nextWritePointer + bufferBytes;
6730 // Check whether the entire write region is behind the play pointer.
6731 if ( leadPointer >= endWrite ) break;
6733 // If we are here, then we must wait until the leadPointer advances
6734 // beyond the end of our next write region. We use the
6735 // Sleep() function to suspend operation until that happens.
6736 double millis = ( endWrite - leadPointer ) * 1000.0;
6737 millis /= ( formatBytes( stream_.deviceFormat[0]) * stream_.nDeviceChannels[0] * stream_.sampleRate);
6738 if ( millis < 1.0 ) millis = 1.0;
6739 Sleep( (DWORD) millis );
6742 if ( dsPointerBetween( nextWritePointer, safeWritePointer, currentWritePointer, dsBufferSize )
6743 || dsPointerBetween( endWrite, safeWritePointer, currentWritePointer, dsBufferSize ) ) {
6744 // We've strayed into the forbidden zone ... resync the read pointer.
6745 handle->xrun[0] = true;
6746 nextWritePointer = safeWritePointer + handle->dsPointerLeadTime[0] - bufferBytes;
6747 if ( nextWritePointer >= dsBufferSize ) nextWritePointer -= dsBufferSize;
6748 handle->bufferPointer[0] = nextWritePointer;
6749 endWrite = nextWritePointer + bufferBytes;
6752 // Lock free space in the buffer
6753 result = dsBuffer->Lock( nextWritePointer, bufferBytes, &buffer1,
6754 &bufferSize1, &buffer2, &bufferSize2, 0 );
6755 if ( FAILED( result ) ) {
6756 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking buffer during playback!";
6757 errorText_ = errorStream_.str();
6758 MUTEX_UNLOCK( &stream_.mutex );
6759 error( RtAudioError::SYSTEM_ERROR );
6763 // Copy our buffer into the DS buffer
6764 CopyMemory( buffer1, buffer, bufferSize1 );
6765 if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 );
6767 // Update our buffer offset and unlock sound buffer
6768 dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
6769 if ( FAILED( result ) ) {
6770 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!";
6771 errorText_ = errorStream_.str();
6772 MUTEX_UNLOCK( &stream_.mutex );
6773 error( RtAudioError::SYSTEM_ERROR );
6776 nextWritePointer = ( nextWritePointer + bufferSize1 + bufferSize2 ) % dsBufferSize;
6777 handle->bufferPointer[0] = nextWritePointer;
6780 // Don't bother draining input
6781 if ( handle->drainCounter ) {
6782 handle->drainCounter++;
6786 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
6788 // Setup parameters.
6789 if ( stream_.doConvertBuffer[1] ) {
6790 buffer = stream_.deviceBuffer;
6791 bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[1];
6792 bufferBytes *= formatBytes( stream_.deviceFormat[1] );
6795 buffer = stream_.userBuffer[1];
6796 bufferBytes = stream_.bufferSize * stream_.nUserChannels[1];
6797 bufferBytes *= formatBytes( stream_.userFormat );
6800 LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6801 long nextReadPointer = handle->bufferPointer[1];
6802 DWORD dsBufferSize = handle->dsBufferSize[1];
6804 // Find out where the write and "safe read" pointers are.
6805 result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer );
6806 if ( FAILED( result ) ) {
6807 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
6808 errorText_ = errorStream_.str();
6809 MUTEX_UNLOCK( &stream_.mutex );
6810 error( RtAudioError::SYSTEM_ERROR );
6814 if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset
6815 DWORD endRead = nextReadPointer + bufferBytes;
6817 // Handling depends on whether we are INPUT or DUPLEX.
6818 // If we're in INPUT mode then waiting is a good thing. If we're in DUPLEX mode,
6819 // then a wait here will drag the write pointers into the forbidden zone.
6821 // In DUPLEX mode, rather than wait, we will back off the read pointer until
6822 // it's in a safe position. This causes dropouts, but it seems to be the only
6823 // practical way to sync up the read and write pointers reliably, given the
6824 // the very complex relationship between phase and increment of the read and write
6827 // In order to minimize audible dropouts in DUPLEX mode, we will
6828 // provide a pre-roll period of 0.5 seconds in which we return
6829 // zeros from the read buffer while the pointers sync up.
6831 if ( stream_.mode == DUPLEX ) {
6832 if ( safeReadPointer < endRead ) {
6833 if ( duplexPrerollBytes <= 0 ) {
6834 // Pre-roll time over. Be more agressive.
6835 int adjustment = endRead-safeReadPointer;
6837 handle->xrun[1] = true;
6839 // - large adjustments: we've probably run out of CPU cycles, so just resync exactly,
6840 // and perform fine adjustments later.
6841 // - small adjustments: back off by twice as much.
6842 if ( adjustment >= 2*bufferBytes )
6843 nextReadPointer = safeReadPointer-2*bufferBytes;
6845 nextReadPointer = safeReadPointer-bufferBytes-adjustment;
6847 if ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;
6851 // In pre=roll time. Just do it.
6852 nextReadPointer = safeReadPointer - bufferBytes;
6853 while ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;
6855 endRead = nextReadPointer + bufferBytes;
6858 else { // mode == INPUT
6859 while ( safeReadPointer < endRead && stream_.callbackInfo.isRunning ) {
6860 // See comments for playback.
6861 double millis = (endRead - safeReadPointer) * 1000.0;
6862 millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate);
6863 if ( millis < 1.0 ) millis = 1.0;
6864 Sleep( (DWORD) millis );
6866 // Wake up and find out where we are now.
6867 result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer );
6868 if ( FAILED( result ) ) {
6869 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
6870 errorText_ = errorStream_.str();
6871 MUTEX_UNLOCK( &stream_.mutex );
6872 error( RtAudioError::SYSTEM_ERROR );
6876 if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset
6880 // Lock free space in the buffer
6881 result = dsBuffer->Lock( nextReadPointer, bufferBytes, &buffer1,
6882 &bufferSize1, &buffer2, &bufferSize2, 0 );
6883 if ( FAILED( result ) ) {
6884 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking capture buffer!";
6885 errorText_ = errorStream_.str();
6886 MUTEX_UNLOCK( &stream_.mutex );
6887 error( RtAudioError::SYSTEM_ERROR );
6891 if ( duplexPrerollBytes <= 0 ) {
6892 // Copy our buffer into the DS buffer
6893 CopyMemory( buffer, buffer1, bufferSize1 );
6894 if ( buffer2 != NULL ) CopyMemory( buffer+bufferSize1, buffer2, bufferSize2 );
6897 memset( buffer, 0, bufferSize1 );
6898 if ( buffer2 != NULL ) memset( buffer + bufferSize1, 0, bufferSize2 );
6899 duplexPrerollBytes -= bufferSize1 + bufferSize2;
6902 // Update our buffer offset and unlock sound buffer
6903 nextReadPointer = ( nextReadPointer + bufferSize1 + bufferSize2 ) % dsBufferSize;
6904 dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
6905 if ( FAILED( result ) ) {
6906 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking capture buffer!";
6907 errorText_ = errorStream_.str();
6908 MUTEX_UNLOCK( &stream_.mutex );
6909 error( RtAudioError::SYSTEM_ERROR );
6912 handle->bufferPointer[1] = nextReadPointer;
6914 // No byte swapping necessary in DirectSound implementation.
6916 // If necessary, convert 8-bit data from unsigned to signed.
6917 if ( stream_.deviceFormat[1] == RTAUDIO_SINT8 )
6918 for ( int j=0; j<bufferBytes; j++ ) buffer[j] = (signed char) ( buffer[j] - 128 );
6920 // Do buffer conversion if necessary.
6921 if ( stream_.doConvertBuffer[1] )
6922 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
6926 MUTEX_UNLOCK( &stream_.mutex );
6927 RtApi::tickStreamTime();
6930 // Definitions for utility functions and callbacks
6931 // specific to the DirectSound implementation.
6933 static unsigned __stdcall callbackHandler( void *ptr )
6935 CallbackInfo *info = (CallbackInfo *) ptr;
6936 RtApiDs *object = (RtApiDs *) info->object;
6937 bool* isRunning = &info->isRunning;
6939 while ( *isRunning == true ) {
6940 object->callbackEvent();
6947 static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
6948 LPCTSTR description,
6952 struct DsProbeData& probeInfo = *(struct DsProbeData*) lpContext;
6953 std::vector<struct DsDevice>& dsDevices = *probeInfo.dsDevices;
6956 bool validDevice = false;
6957 if ( probeInfo.isInput == true ) {
6959 LPDIRECTSOUNDCAPTURE object;
6961 hr = DirectSoundCaptureCreate( lpguid, &object, NULL );
6962 if ( hr != DS_OK ) return TRUE;
6964 caps.dwSize = sizeof(caps);
6965 hr = object->GetCaps( &caps );
6966 if ( hr == DS_OK ) {
6967 if ( caps.dwChannels > 0 && caps.dwFormats > 0 )
6974 LPDIRECTSOUND object;
6975 hr = DirectSoundCreate( lpguid, &object, NULL );
6976 if ( hr != DS_OK ) return TRUE;
6978 caps.dwSize = sizeof(caps);
6979 hr = object->GetCaps( &caps );
6980 if ( hr == DS_OK ) {
6981 if ( caps.dwFlags & DSCAPS_PRIMARYMONO || caps.dwFlags & DSCAPS_PRIMARYSTEREO )
6987 // If good device, then save its name and guid.
6988 std::string name = convertCharPointerToStdString( description );
6989 //if ( name == "Primary Sound Driver" || name == "Primary Sound Capture Driver" )
6990 if ( lpguid == NULL )
6991 name = "Default Device";
6992 if ( validDevice ) {
6993 for ( unsigned int i=0; i<dsDevices.size(); i++ ) {
6994 if ( dsDevices[i].name == name ) {
6995 dsDevices[i].found = true;
6996 if ( probeInfo.isInput ) {
6997 dsDevices[i].id[1] = lpguid;
6998 dsDevices[i].validId[1] = true;
7001 dsDevices[i].id[0] = lpguid;
7002 dsDevices[i].validId[0] = true;
7010 device.found = true;
7011 if ( probeInfo.isInput ) {
7012 device.id[1] = lpguid;
7013 device.validId[1] = true;
7016 device.id[0] = lpguid;
7017 device.validId[0] = true;
7019 dsDevices.push_back( device );
7025 static const char* getErrorString( int code )
7029 case DSERR_ALLOCATED:
7030 return "Already allocated";
7032 case DSERR_CONTROLUNAVAIL:
7033 return "Control unavailable";
7035 case DSERR_INVALIDPARAM:
7036 return "Invalid parameter";
7038 case DSERR_INVALIDCALL:
7039 return "Invalid call";
7042 return "Generic error";
7044 case DSERR_PRIOLEVELNEEDED:
7045 return "Priority level needed";
7047 case DSERR_OUTOFMEMORY:
7048 return "Out of memory";
7050 case DSERR_BADFORMAT:
7051 return "The sample rate or the channel format is not supported";
7053 case DSERR_UNSUPPORTED:
7054 return "Not supported";
7056 case DSERR_NODRIVER:
7059 case DSERR_ALREADYINITIALIZED:
7060 return "Already initialized";
7062 case DSERR_NOAGGREGATION:
7063 return "No aggregation";
7065 case DSERR_BUFFERLOST:
7066 return "Buffer lost";
7068 case DSERR_OTHERAPPHASPRIO:
7069 return "Another application already has priority";
7071 case DSERR_UNINITIALIZED:
7072 return "Uninitialized";
7075 return "DirectSound unknown error";
7078 //******************** End of __WINDOWS_DS__ *********************//
7082 #if defined(__LINUX_ALSA__)
7084 #include <alsa/asoundlib.h>
7087 // A structure to hold various information related to the ALSA API
7090 snd_pcm_t *handles[2];
7093 pthread_cond_t runnable_cv;
7097 :synchronized(false), runnable(false) { xrun[0] = false; xrun[1] = false; }
7100 static void *alsaCallbackHandler( void * ptr );
7102 RtApiAlsa :: RtApiAlsa()
7104 // Nothing to do here.
7107 RtApiAlsa :: ~RtApiAlsa()
7109 if ( stream_.state != STREAM_CLOSED ) closeStream();
7112 unsigned int RtApiAlsa :: getDeviceCount( void )
7114 unsigned nDevices = 0;
7115 int result, subdevice, card;
7119 // Count cards and devices
7121 snd_card_next( &card );
7122 while ( card >= 0 ) {
7123 sprintf( name, "hw:%d", card );
7124 result = snd_ctl_open( &handle, name, 0 );
7126 errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << ".";
7127 errorText_ = errorStream_.str();
7128 error( RtAudioError::WARNING );
7133 result = snd_ctl_pcm_next_device( handle, &subdevice );
7135 errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
7136 errorText_ = errorStream_.str();
7137 error( RtAudioError::WARNING );
7140 if ( subdevice < 0 )
7145 snd_ctl_close( handle );
7146 snd_card_next( &card );
7149 result = snd_ctl_open( &handle, "default", 0 );
7152 snd_ctl_close( handle );
7158 RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device )
7160 RtAudio::DeviceInfo info;
7161 info.probed = false;
7163 unsigned nDevices = 0;
7164 int result, subdevice, card;
7168 // Count cards and devices
7171 snd_card_next( &card );
7172 while ( card >= 0 ) {
7173 sprintf( name, "hw:%d", card );
7174 result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
7176 errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << ".";
7177 errorText_ = errorStream_.str();
7178 error( RtAudioError::WARNING );
7183 result = snd_ctl_pcm_next_device( chandle, &subdevice );
7185 errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
7186 errorText_ = errorStream_.str();
7187 error( RtAudioError::WARNING );
7190 if ( subdevice < 0 ) break;
7191 if ( nDevices == device ) {
7192 sprintf( name, "hw:%d,%d", card, subdevice );
7198 snd_ctl_close( chandle );
7199 snd_card_next( &card );
7202 result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
7203 if ( result == 0 ) {
7204 if ( nDevices == device ) {
7205 strcpy( name, "default" );
7211 if ( nDevices == 0 ) {
7212 errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!";
7213 error( RtAudioError::INVALID_USE );
7217 if ( device >= nDevices ) {
7218 errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!";
7219 error( RtAudioError::INVALID_USE );
7225 // If a stream is already open, we cannot probe the stream devices.
7226 // Thus, use the saved results.
7227 if ( stream_.state != STREAM_CLOSED &&
7228 ( stream_.device[0] == device || stream_.device[1] == device ) ) {
7229 snd_ctl_close( chandle );
7230 if ( device >= devices_.size() ) {
7231 errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened.";
7232 error( RtAudioError::WARNING );
7235 return devices_[ device ];
7238 int openMode = SND_PCM_ASYNC;
7239 snd_pcm_stream_t stream;
7240 snd_pcm_info_t *pcminfo;
7241 snd_pcm_info_alloca( &pcminfo );
7243 snd_pcm_hw_params_t *params;
7244 snd_pcm_hw_params_alloca( ¶ms );
7246 // First try for playback unless default device (which has subdev -1)
7247 stream = SND_PCM_STREAM_PLAYBACK;
7248 snd_pcm_info_set_stream( pcminfo, stream );
7249 if ( subdevice != -1 ) {
7250 snd_pcm_info_set_device( pcminfo, subdevice );
7251 snd_pcm_info_set_subdevice( pcminfo, 0 );
7253 result = snd_ctl_pcm_info( chandle, pcminfo );
7255 // Device probably doesn't support playback.
7260 result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK );
7262 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
7263 errorText_ = errorStream_.str();
7264 error( RtAudioError::WARNING );
7268 // The device is open ... fill the parameter structure.
7269 result = snd_pcm_hw_params_any( phandle, params );
7271 snd_pcm_close( phandle );
7272 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
7273 errorText_ = errorStream_.str();
7274 error( RtAudioError::WARNING );
7278 // Get output channel information.
7280 result = snd_pcm_hw_params_get_channels_max( params, &value );
7282 snd_pcm_close( phandle );
7283 errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << ".";
7284 errorText_ = errorStream_.str();
7285 error( RtAudioError::WARNING );
7288 info.outputChannels = value;
7289 snd_pcm_close( phandle );
7292 stream = SND_PCM_STREAM_CAPTURE;
7293 snd_pcm_info_set_stream( pcminfo, stream );
7295 // Now try for capture unless default device (with subdev = -1)
7296 if ( subdevice != -1 ) {
7297 result = snd_ctl_pcm_info( chandle, pcminfo );
7298 snd_ctl_close( chandle );
7300 // Device probably doesn't support capture.
7301 if ( info.outputChannels == 0 ) return info;
7302 goto probeParameters;
7306 snd_ctl_close( chandle );
7308 result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
7310 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
7311 errorText_ = errorStream_.str();
7312 error( RtAudioError::WARNING );
7313 if ( info.outputChannels == 0 ) return info;
7314 goto probeParameters;
7317 // The device is open ... fill the parameter structure.
7318 result = snd_pcm_hw_params_any( phandle, params );
7320 snd_pcm_close( phandle );
7321 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
7322 errorText_ = errorStream_.str();
7323 error( RtAudioError::WARNING );
7324 if ( info.outputChannels == 0 ) return info;
7325 goto probeParameters;
7328 result = snd_pcm_hw_params_get_channels_max( params, &value );
7330 snd_pcm_close( phandle );
7331 errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << ".";
7332 errorText_ = errorStream_.str();
7333 error( RtAudioError::WARNING );
7334 if ( info.outputChannels == 0 ) return info;
7335 goto probeParameters;
7337 info.inputChannels = value;
7338 snd_pcm_close( phandle );
7340 // If device opens for both playback and capture, we determine the channels.
7341 if ( info.outputChannels > 0 && info.inputChannels > 0 )
7342 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
7344 // ALSA doesn't provide default devices so we'll use the first available one.
7345 if ( device == 0 && info.outputChannels > 0 )
7346 info.isDefaultOutput = true;
7347 if ( device == 0 && info.inputChannels > 0 )
7348 info.isDefaultInput = true;
7351 // At this point, we just need to figure out the supported data
7352 // formats and sample rates. We'll proceed by opening the device in
7353 // the direction with the maximum number of channels, or playback if
7354 // they are equal. This might limit our sample rate options, but so
7357 if ( info.outputChannels >= info.inputChannels )
7358 stream = SND_PCM_STREAM_PLAYBACK;
7360 stream = SND_PCM_STREAM_CAPTURE;
7361 snd_pcm_info_set_stream( pcminfo, stream );
7363 result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
7365 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
7366 errorText_ = errorStream_.str();
7367 error( RtAudioError::WARNING );
7371 // The device is open ... fill the parameter structure.
7372 result = snd_pcm_hw_params_any( phandle, params );
7374 snd_pcm_close( phandle );
7375 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
7376 errorText_ = errorStream_.str();
7377 error( RtAudioError::WARNING );
7381 // Test our discrete set of sample rate values.
7382 info.sampleRates.clear();
7383 for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {
7384 if ( snd_pcm_hw_params_test_rate( phandle, params, SAMPLE_RATES[i], 0 ) == 0 ) {
7385 info.sampleRates.push_back( SAMPLE_RATES[i] );
7387 if ( !info.preferredSampleRate || ( SAMPLE_RATES[i] <= 48000 && SAMPLE_RATES[i] > info.preferredSampleRate ) )
7388 info.preferredSampleRate = SAMPLE_RATES[i];
7391 if ( info.sampleRates.size() == 0 ) {
7392 snd_pcm_close( phandle );
7393 errorStream_ << "RtApiAlsa::getDeviceInfo: no supported sample rates found for device (" << name << ").";
7394 errorText_ = errorStream_.str();
7395 error( RtAudioError::WARNING );
7399 // Probe the supported data formats ... we don't care about endian-ness just yet
7400 snd_pcm_format_t format;
7401 info.nativeFormats = 0;
7402 format = SND_PCM_FORMAT_S8;
7403 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7404 info.nativeFormats |= RTAUDIO_SINT8;
7405 format = SND_PCM_FORMAT_S16;
7406 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7407 info.nativeFormats |= RTAUDIO_SINT16;
7408 format = SND_PCM_FORMAT_S24;
7409 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7410 info.nativeFormats |= RTAUDIO_SINT24;
7411 format = SND_PCM_FORMAT_S32;
7412 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7413 info.nativeFormats |= RTAUDIO_SINT32;
7414 format = SND_PCM_FORMAT_FLOAT;
7415 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7416 info.nativeFormats |= RTAUDIO_FLOAT32;
7417 format = SND_PCM_FORMAT_FLOAT64;
7418 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7419 info.nativeFormats |= RTAUDIO_FLOAT64;
7421 // Check that we have at least one supported format
7422 if ( info.nativeFormats == 0 ) {
7423 snd_pcm_close( phandle );
7424 errorStream_ << "RtApiAlsa::getDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio.";
7425 errorText_ = errorStream_.str();
7426 error( RtAudioError::WARNING );
7430 // Get the device name
7432 result = snd_card_get_name( card, &cardname );
7433 if ( result >= 0 ) {
7434 sprintf( name, "hw:%s,%d", cardname, subdevice );
7439 // That's all ... close the device and return
7440 snd_pcm_close( phandle );
7445 void RtApiAlsa :: saveDeviceInfo( void )
7449 unsigned int nDevices = getDeviceCount();
7450 devices_.resize( nDevices );
7451 for ( unsigned int i=0; i<nDevices; i++ )
7452 devices_[i] = getDeviceInfo( i );
7455 bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
7456 unsigned int firstChannel, unsigned int sampleRate,
7457 RtAudioFormat format, unsigned int *bufferSize,
7458 RtAudio::StreamOptions *options )
7461 #if defined(__RTAUDIO_DEBUG__)
7463 snd_output_stdio_attach(&out, stderr, 0);
7466 // I'm not using the "plug" interface ... too much inconsistent behavior.
7468 unsigned nDevices = 0;
7469 int result, subdevice, card;
7473 if ( options && options->flags & RTAUDIO_ALSA_USE_DEFAULT )
7474 snprintf(name, sizeof(name), "%s", "default");
7476 // Count cards and devices
7478 snd_card_next( &card );
7479 while ( card >= 0 ) {
7480 sprintf( name, "hw:%d", card );
7481 result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
7483 errorStream_ << "RtApiAlsa::probeDeviceOpen: control open, card = " << card << ", " << snd_strerror( result ) << ".";
7484 errorText_ = errorStream_.str();
7489 result = snd_ctl_pcm_next_device( chandle, &subdevice );
7490 if ( result < 0 ) break;
7491 if ( subdevice < 0 ) break;
7492 if ( nDevices == device ) {
7493 sprintf( name, "hw:%d,%d", card, subdevice );
7494 snd_ctl_close( chandle );
7499 snd_ctl_close( chandle );
7500 snd_card_next( &card );
7503 result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
7504 if ( result == 0 ) {
7505 if ( nDevices == device ) {
7506 strcpy( name, "default" );
7507 snd_ctl_close( chandle );
7512 snd_ctl_close( chandle );
7514 if ( nDevices == 0 ) {
7515 // This should not happen because a check is made before this function is called.
7516 errorText_ = "RtApiAlsa::probeDeviceOpen: no devices found!";
7520 if ( device >= nDevices ) {
7521 // This should not happen because a check is made before this function is called.
7522 errorText_ = "RtApiAlsa::probeDeviceOpen: device ID is invalid!";
7529 // The getDeviceInfo() function will not work for a device that is
7530 // already open. Thus, we'll probe the system before opening a
7531 // stream and save the results for use by getDeviceInfo().
7532 if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) // only do once
7533 this->saveDeviceInfo();
7535 snd_pcm_stream_t stream;
7536 if ( mode == OUTPUT )
7537 stream = SND_PCM_STREAM_PLAYBACK;
7539 stream = SND_PCM_STREAM_CAPTURE;
7542 int openMode = SND_PCM_ASYNC;
7543 result = snd_pcm_open( &phandle, name, stream, openMode );
7545 if ( mode == OUTPUT )
7546 errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for output.";
7548 errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for input.";
7549 errorText_ = errorStream_.str();
7553 // Fill the parameter structure.
7554 snd_pcm_hw_params_t *hw_params;
7555 snd_pcm_hw_params_alloca( &hw_params );
7556 result = snd_pcm_hw_params_any( phandle, hw_params );
7558 snd_pcm_close( phandle );
7559 errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") parameters, " << snd_strerror( result ) << ".";
7560 errorText_ = errorStream_.str();
7564 #if defined(__RTAUDIO_DEBUG__)
7565 fprintf( stderr, "\nRtApiAlsa: dump hardware params just after device open:\n\n" );
7566 snd_pcm_hw_params_dump( hw_params, out );
7569 // Set access ... check user preference.
7570 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) {
7571 stream_.userInterleaved = false;
7572 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
7574 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
7575 stream_.deviceInterleaved[mode] = true;
7578 stream_.deviceInterleaved[mode] = false;
7581 stream_.userInterleaved = true;
7582 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
7584 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
7585 stream_.deviceInterleaved[mode] = false;
7588 stream_.deviceInterleaved[mode] = true;
7592 snd_pcm_close( phandle );
7593 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") access, " << snd_strerror( result ) << ".";
7594 errorText_ = errorStream_.str();
7598 // Determine how to set the device format.
7599 stream_.userFormat = format;
7600 snd_pcm_format_t deviceFormat = SND_PCM_FORMAT_UNKNOWN;
7602 if ( format == RTAUDIO_SINT8 )
7603 deviceFormat = SND_PCM_FORMAT_S8;
7604 else if ( format == RTAUDIO_SINT16 )
7605 deviceFormat = SND_PCM_FORMAT_S16;
7606 else if ( format == RTAUDIO_SINT24 )
7607 deviceFormat = SND_PCM_FORMAT_S24;
7608 else if ( format == RTAUDIO_SINT32 )
7609 deviceFormat = SND_PCM_FORMAT_S32;
7610 else if ( format == RTAUDIO_FLOAT32 )
7611 deviceFormat = SND_PCM_FORMAT_FLOAT;
7612 else if ( format == RTAUDIO_FLOAT64 )
7613 deviceFormat = SND_PCM_FORMAT_FLOAT64;
7615 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat) == 0) {
7616 stream_.deviceFormat[mode] = format;
7620 // The user requested format is not natively supported by the device.
7621 deviceFormat = SND_PCM_FORMAT_FLOAT64;
7622 if ( snd_pcm_hw_params_test_format( phandle, hw_params, deviceFormat ) == 0 ) {
7623 stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
7627 deviceFormat = SND_PCM_FORMAT_FLOAT;
7628 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7629 stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
7633 deviceFormat = SND_PCM_FORMAT_S32;
7634 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7635 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
7639 deviceFormat = SND_PCM_FORMAT_S24;
7640 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7641 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
7645 deviceFormat = SND_PCM_FORMAT_S16;
7646 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7647 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
7651 deviceFormat = SND_PCM_FORMAT_S8;
7652 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7653 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
7657 // If we get here, no supported format was found.
7658 snd_pcm_close( phandle );
7659 errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device " << device << " data format not supported by RtAudio.";
7660 errorText_ = errorStream_.str();
7664 result = snd_pcm_hw_params_set_format( phandle, hw_params, deviceFormat );
7666 snd_pcm_close( phandle );
7667 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") data format, " << snd_strerror( result ) << ".";
7668 errorText_ = errorStream_.str();
7672 // Determine whether byte-swaping is necessary.
7673 stream_.doByteSwap[mode] = false;
7674 if ( deviceFormat != SND_PCM_FORMAT_S8 ) {
7675 result = snd_pcm_format_cpu_endian( deviceFormat );
7677 stream_.doByteSwap[mode] = true;
7678 else if (result < 0) {
7679 snd_pcm_close( phandle );
7680 errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") endian-ness, " << snd_strerror( result ) << ".";
7681 errorText_ = errorStream_.str();
7686 // Set the sample rate.
7687 result = snd_pcm_hw_params_set_rate_near( phandle, hw_params, (unsigned int*) &sampleRate, 0 );
7689 snd_pcm_close( phandle );
7690 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting sample rate on device (" << name << "), " << snd_strerror( result ) << ".";
7691 errorText_ = errorStream_.str();
7695 // Determine the number of channels for this device. We support a possible
7696 // minimum device channel number > than the value requested by the user.
7697 stream_.nUserChannels[mode] = channels;
7699 result = snd_pcm_hw_params_get_channels_max( hw_params, &value );
7700 unsigned int deviceChannels = value;
7701 if ( result < 0 || deviceChannels < channels + firstChannel ) {
7702 snd_pcm_close( phandle );
7703 errorStream_ << "RtApiAlsa::probeDeviceOpen: requested channel parameters not supported by device (" << name << "), " << snd_strerror( result ) << ".";
7704 errorText_ = errorStream_.str();
7708 result = snd_pcm_hw_params_get_channels_min( hw_params, &value );
7710 snd_pcm_close( phandle );
7711 errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting minimum channels for device (" << name << "), " << snd_strerror( result ) << ".";
7712 errorText_ = errorStream_.str();
7715 deviceChannels = value;
7716 if ( deviceChannels < channels + firstChannel ) deviceChannels = channels + firstChannel;
7717 stream_.nDeviceChannels[mode] = deviceChannels;
7719 // Set the device channels.
7720 result = snd_pcm_hw_params_set_channels( phandle, hw_params, deviceChannels );
7722 snd_pcm_close( phandle );
7723 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting channels for device (" << name << "), " << snd_strerror( result ) << ".";
7724 errorText_ = errorStream_.str();
7728 // Set the buffer (or period) size.
7730 snd_pcm_uframes_t periodSize = *bufferSize;
7731 result = snd_pcm_hw_params_set_period_size_near( phandle, hw_params, &periodSize, &dir );
7733 snd_pcm_close( phandle );
7734 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting period size for device (" << name << "), " << snd_strerror( result ) << ".";
7735 errorText_ = errorStream_.str();
7738 *bufferSize = periodSize;
7740 // Set the buffer number, which in ALSA is referred to as the "period".
7741 unsigned int periods = 0;
7742 if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2;
7743 if ( options && options->numberOfBuffers > 0 ) periods = options->numberOfBuffers;
7744 if ( periods < 2 ) periods = 4; // a fairly safe default value
7745 result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir );
7747 snd_pcm_close( phandle );
7748 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << ".";
7749 errorText_ = errorStream_.str();
7753 // If attempting to setup a duplex stream, the bufferSize parameter
7754 // MUST be the same in both directions!
7755 if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {
7756 snd_pcm_close( phandle );
7757 errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ").";
7758 errorText_ = errorStream_.str();
7762 stream_.bufferSize = *bufferSize;
7764 // Install the hardware configuration
7765 result = snd_pcm_hw_params( phandle, hw_params );
7767 snd_pcm_close( phandle );
7768 errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing hardware configuration on device (" << name << "), " << snd_strerror( result ) << ".";
7769 errorText_ = errorStream_.str();
7773 #if defined(__RTAUDIO_DEBUG__)
7774 fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n");
7775 snd_pcm_hw_params_dump( hw_params, out );
7778 // Set the software configuration to fill buffers with zeros and prevent device stopping on xruns.
7779 snd_pcm_sw_params_t *sw_params = NULL;
7780 snd_pcm_sw_params_alloca( &sw_params );
7781 snd_pcm_sw_params_current( phandle, sw_params );
7782 snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize );
7783 snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, ULONG_MAX );
7784 snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 );
7786 // The following two settings were suggested by Theo Veenker
7787 //snd_pcm_sw_params_set_avail_min( phandle, sw_params, *bufferSize );
7788 //snd_pcm_sw_params_set_xfer_align( phandle, sw_params, 1 );
7790 // here are two options for a fix
7791 //snd_pcm_sw_params_set_silence_size( phandle, sw_params, ULONG_MAX );
7792 snd_pcm_uframes_t val;
7793 snd_pcm_sw_params_get_boundary( sw_params, &val );
7794 snd_pcm_sw_params_set_silence_size( phandle, sw_params, val );
7796 result = snd_pcm_sw_params( phandle, sw_params );
7798 snd_pcm_close( phandle );
7799 errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing software configuration on device (" << name << "), " << snd_strerror( result ) << ".";
7800 errorText_ = errorStream_.str();
7804 #if defined(__RTAUDIO_DEBUG__)
7805 fprintf(stderr, "\nRtApiAlsa: dump software params after installation:\n\n");
7806 snd_pcm_sw_params_dump( sw_params, out );
7809 // Set flags for buffer conversion
7810 stream_.doConvertBuffer[mode] = false;
7811 if ( stream_.userFormat != stream_.deviceFormat[mode] )
7812 stream_.doConvertBuffer[mode] = true;
7813 if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
7814 stream_.doConvertBuffer[mode] = true;
7815 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
7816 stream_.nUserChannels[mode] > 1 )
7817 stream_.doConvertBuffer[mode] = true;
7819 // Allocate the ApiHandle if necessary and then save.
7820 AlsaHandle *apiInfo = 0;
7821 if ( stream_.apiHandle == 0 ) {
7823 apiInfo = (AlsaHandle *) new AlsaHandle;
7825 catch ( std::bad_alloc& ) {
7826 errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating AlsaHandle memory.";
7830 if ( pthread_cond_init( &apiInfo->runnable_cv, NULL ) ) {
7831 errorText_ = "RtApiAlsa::probeDeviceOpen: error initializing pthread condition variable.";
7835 stream_.apiHandle = (void *) apiInfo;
7836 apiInfo->handles[0] = 0;
7837 apiInfo->handles[1] = 0;
7840 apiInfo = (AlsaHandle *) stream_.apiHandle;
7842 apiInfo->handles[mode] = phandle;
7845 // Allocate necessary internal buffers.
7846 unsigned long bufferBytes;
7847 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
7848 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
7849 if ( stream_.userBuffer[mode] == NULL ) {
7850 errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating user buffer memory.";
7854 if ( stream_.doConvertBuffer[mode] ) {
7856 bool makeBuffer = true;
7857 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
7858 if ( mode == INPUT ) {
7859 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
7860 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
7861 if ( bufferBytes <= bytesOut ) makeBuffer = false;
7866 bufferBytes *= *bufferSize;
7867 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
7868 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
7869 if ( stream_.deviceBuffer == NULL ) {
7870 errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating device buffer memory.";
7876 stream_.sampleRate = sampleRate;
7877 stream_.nBuffers = periods;
7878 stream_.device[mode] = device;
7879 stream_.state = STREAM_STOPPED;
7881 // Setup the buffer conversion information structure.
7882 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
7884 // Setup thread if necessary.
7885 if ( stream_.mode == OUTPUT && mode == INPUT ) {
7886 // We had already set up an output stream.
7887 stream_.mode = DUPLEX;
7888 // Link the streams if possible.
7889 apiInfo->synchronized = false;
7890 if ( snd_pcm_link( apiInfo->handles[0], apiInfo->handles[1] ) == 0 )
7891 apiInfo->synchronized = true;
7893 errorText_ = "RtApiAlsa::probeDeviceOpen: unable to synchronize input and output devices.";
7894 error( RtAudioError::WARNING );
7898 stream_.mode = mode;
7900 // Setup callback thread.
7901 stream_.callbackInfo.object = (void *) this;
7903 // Set the thread attributes for joinable and realtime scheduling
7904 // priority (optional). The higher priority will only take affect
7905 // if the program is run as root or suid. Note, under Linux
7906 // processes with CAP_SYS_NICE privilege, a user can change
7907 // scheduling policy and priority (thus need not be root). See
7908 // POSIX "capabilities".
7909 pthread_attr_t attr;
7910 pthread_attr_init( &attr );
7911 pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
7912 #ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
7913 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
7914 stream_.callbackInfo.doRealtime = true;
7915 struct sched_param param;
7916 int priority = options->priority;
7917 int min = sched_get_priority_min( SCHED_RR );
7918 int max = sched_get_priority_max( SCHED_RR );
7919 if ( priority < min ) priority = min;
7920 else if ( priority > max ) priority = max;
7921 param.sched_priority = priority;
7923 // Set the policy BEFORE the priority. Otherwise it fails.
7924 pthread_attr_setschedpolicy(&attr, SCHED_RR);
7925 pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM);
7926 // This is definitely required. Otherwise it fails.
7927 pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED);
7928 pthread_attr_setschedparam(&attr, ¶m);
7931 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
7933 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
7936 stream_.callbackInfo.isRunning = true;
7937 result = pthread_create( &stream_.callbackInfo.thread, &attr, alsaCallbackHandler, &stream_.callbackInfo );
7938 pthread_attr_destroy( &attr );
7940 // Failed. Try instead with default attributes.
7941 result = pthread_create( &stream_.callbackInfo.thread, NULL, alsaCallbackHandler, &stream_.callbackInfo );
7943 stream_.callbackInfo.isRunning = false;
7944 errorText_ = "RtApiAlsa::error creating callback thread!";
7954 pthread_cond_destroy( &apiInfo->runnable_cv );
7955 if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
7956 if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
7958 stream_.apiHandle = 0;
7961 if ( phandle) snd_pcm_close( phandle );
7963 for ( int i=0; i<2; i++ ) {
7964 if ( stream_.userBuffer[i] ) {
7965 free( stream_.userBuffer[i] );
7966 stream_.userBuffer[i] = 0;
7970 if ( stream_.deviceBuffer ) {
7971 free( stream_.deviceBuffer );
7972 stream_.deviceBuffer = 0;
7975 stream_.state = STREAM_CLOSED;
7979 void RtApiAlsa :: closeStream()
7981 if ( stream_.state == STREAM_CLOSED ) {
7982 errorText_ = "RtApiAlsa::closeStream(): no open stream to close!";
7983 error( RtAudioError::WARNING );
7987 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
7988 stream_.callbackInfo.isRunning = false;
7989 MUTEX_LOCK( &stream_.mutex );
7990 if ( stream_.state == STREAM_STOPPED ) {
7991 apiInfo->runnable = true;
7992 pthread_cond_signal( &apiInfo->runnable_cv );
7994 MUTEX_UNLOCK( &stream_.mutex );
7995 pthread_join( stream_.callbackInfo.thread, NULL );
7997 if ( stream_.state == STREAM_RUNNING ) {
7998 stream_.state = STREAM_STOPPED;
7999 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
8000 snd_pcm_drop( apiInfo->handles[0] );
8001 if ( stream_.mode == INPUT || stream_.mode == DUPLEX )
8002 snd_pcm_drop( apiInfo->handles[1] );
8006 pthread_cond_destroy( &apiInfo->runnable_cv );
8007 if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
8008 if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
8010 stream_.apiHandle = 0;
8013 for ( int i=0; i<2; i++ ) {
8014 if ( stream_.userBuffer[i] ) {
8015 free( stream_.userBuffer[i] );
8016 stream_.userBuffer[i] = 0;
8020 if ( stream_.deviceBuffer ) {
8021 free( stream_.deviceBuffer );
8022 stream_.deviceBuffer = 0;
8025 stream_.mode = UNINITIALIZED;
8026 stream_.state = STREAM_CLOSED;
8029 void RtApiAlsa :: startStream()
8031 // This method calls snd_pcm_prepare if the device isn't already in that state.
8034 if ( stream_.state == STREAM_RUNNING ) {
8035 errorText_ = "RtApiAlsa::startStream(): the stream is already running!";
8036 error( RtAudioError::WARNING );
8040 MUTEX_LOCK( &stream_.mutex );
8043 snd_pcm_state_t state;
8044 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
8045 snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
8046 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
8047 state = snd_pcm_state( handle[0] );
8048 if ( state != SND_PCM_STATE_PREPARED ) {
8049 result = snd_pcm_prepare( handle[0] );
8051 errorStream_ << "RtApiAlsa::startStream: error preparing output pcm device, " << snd_strerror( result ) << ".";
8052 errorText_ = errorStream_.str();
8058 if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
8059 result = snd_pcm_drop(handle[1]); // fix to remove stale data received since device has been open
8060 state = snd_pcm_state( handle[1] );
8061 if ( state != SND_PCM_STATE_PREPARED ) {
8062 result = snd_pcm_prepare( handle[1] );
8064 errorStream_ << "RtApiAlsa::startStream: error preparing input pcm device, " << snd_strerror( result ) << ".";
8065 errorText_ = errorStream_.str();
8071 stream_.state = STREAM_RUNNING;
8074 apiInfo->runnable = true;
8075 pthread_cond_signal( &apiInfo->runnable_cv );
8076 MUTEX_UNLOCK( &stream_.mutex );
8078 if ( result >= 0 ) return;
8079 error( RtAudioError::SYSTEM_ERROR );
8082 void RtApiAlsa :: stopStream()
8085 if ( stream_.state == STREAM_STOPPED ) {
8086 errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!";
8087 error( RtAudioError::WARNING );
8091 stream_.state = STREAM_STOPPED;
8092 MUTEX_LOCK( &stream_.mutex );
8095 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
8096 snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
8097 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
8098 if ( apiInfo->synchronized )
8099 result = snd_pcm_drop( handle[0] );
8101 result = snd_pcm_drain( handle[0] );
8103 errorStream_ << "RtApiAlsa::stopStream: error draining output pcm device, " << snd_strerror( result ) << ".";
8104 errorText_ = errorStream_.str();
8109 if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
8110 result = snd_pcm_drop( handle[1] );
8112 errorStream_ << "RtApiAlsa::stopStream: error stopping input pcm device, " << snd_strerror( result ) << ".";
8113 errorText_ = errorStream_.str();
8119 apiInfo->runnable = false; // fixes high CPU usage when stopped
8120 MUTEX_UNLOCK( &stream_.mutex );
8122 if ( result >= 0 ) return;
8123 error( RtAudioError::SYSTEM_ERROR );
8126 void RtApiAlsa :: abortStream()
8129 if ( stream_.state == STREAM_STOPPED ) {
8130 errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!";
8131 error( RtAudioError::WARNING );
8135 stream_.state = STREAM_STOPPED;
8136 MUTEX_LOCK( &stream_.mutex );
8139 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
8140 snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
8141 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
8142 result = snd_pcm_drop( handle[0] );
8144 errorStream_ << "RtApiAlsa::abortStream: error aborting output pcm device, " << snd_strerror( result ) << ".";
8145 errorText_ = errorStream_.str();
8150 if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
8151 result = snd_pcm_drop( handle[1] );
8153 errorStream_ << "RtApiAlsa::abortStream: error aborting input pcm device, " << snd_strerror( result ) << ".";
8154 errorText_ = errorStream_.str();
8160 apiInfo->runnable = false; // fixes high CPU usage when stopped
8161 MUTEX_UNLOCK( &stream_.mutex );
8163 if ( result >= 0 ) return;
8164 error( RtAudioError::SYSTEM_ERROR );
8167 void RtApiAlsa :: callbackEvent()
8169 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
8170 if ( stream_.state == STREAM_STOPPED ) {
8171 MUTEX_LOCK( &stream_.mutex );
8172 while ( !apiInfo->runnable )
8173 pthread_cond_wait( &apiInfo->runnable_cv, &stream_.mutex );
8175 if ( stream_.state != STREAM_RUNNING ) {
8176 MUTEX_UNLOCK( &stream_.mutex );
8179 MUTEX_UNLOCK( &stream_.mutex );
8182 if ( stream_.state == STREAM_CLOSED ) {
8183 errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!";
8184 error( RtAudioError::WARNING );
8188 int doStopStream = 0;
8189 RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
8190 double streamTime = getStreamTime();
8191 RtAudioStreamStatus status = 0;
8192 if ( stream_.mode != INPUT && apiInfo->xrun[0] == true ) {
8193 status |= RTAUDIO_OUTPUT_UNDERFLOW;
8194 apiInfo->xrun[0] = false;
8196 if ( stream_.mode != OUTPUT && apiInfo->xrun[1] == true ) {
8197 status |= RTAUDIO_INPUT_OVERFLOW;
8198 apiInfo->xrun[1] = false;
8200 doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
8201 stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
8203 if ( doStopStream == 2 ) {
8208 MUTEX_LOCK( &stream_.mutex );
8210 // The state might change while waiting on a mutex.
8211 if ( stream_.state == STREAM_STOPPED ) goto unlock;
8217 snd_pcm_sframes_t frames;
8218 RtAudioFormat format;
8219 handle = (snd_pcm_t **) apiInfo->handles;
8221 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
8223 // Setup parameters.
8224 if ( stream_.doConvertBuffer[1] ) {
8225 buffer = stream_.deviceBuffer;
8226 channels = stream_.nDeviceChannels[1];
8227 format = stream_.deviceFormat[1];
8230 buffer = stream_.userBuffer[1];
8231 channels = stream_.nUserChannels[1];
8232 format = stream_.userFormat;
8235 // Read samples from device in interleaved/non-interleaved format.
8236 if ( stream_.deviceInterleaved[1] )
8237 result = snd_pcm_readi( handle[1], buffer, stream_.bufferSize );
8239 void *bufs[channels];
8240 size_t offset = stream_.bufferSize * formatBytes( format );
8241 for ( int i=0; i<channels; i++ )
8242 bufs[i] = (void *) (buffer + (i * offset));
8243 result = snd_pcm_readn( handle[1], bufs, stream_.bufferSize );
8246 if ( result < (int) stream_.bufferSize ) {
8247 // Either an error or overrun occured.
8248 if ( result == -EPIPE ) {
8249 snd_pcm_state_t state = snd_pcm_state( handle[1] );
8250 if ( state == SND_PCM_STATE_XRUN ) {
8251 apiInfo->xrun[1] = true;
8252 result = snd_pcm_prepare( handle[1] );
8254 errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after overrun, " << snd_strerror( result ) << ".";
8255 errorText_ = errorStream_.str();
8259 errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
8260 errorText_ = errorStream_.str();
8264 errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << ".";
8265 errorText_ = errorStream_.str();
8267 error( RtAudioError::WARNING );
8271 // Do byte swapping if necessary.
8272 if ( stream_.doByteSwap[1] )
8273 byteSwapBuffer( buffer, stream_.bufferSize * channels, format );
8275 // Do buffer conversion if necessary.
8276 if ( stream_.doConvertBuffer[1] )
8277 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
8279 // Check stream latency
8280 result = snd_pcm_delay( handle[1], &frames );
8281 if ( result == 0 && frames > 0 ) stream_.latency[1] = frames;
8286 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
8288 // Setup parameters and do buffer conversion if necessary.
8289 if ( stream_.doConvertBuffer[0] ) {
8290 buffer = stream_.deviceBuffer;
8291 convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
8292 channels = stream_.nDeviceChannels[0];
8293 format = stream_.deviceFormat[0];
8296 buffer = stream_.userBuffer[0];
8297 channels = stream_.nUserChannels[0];
8298 format = stream_.userFormat;
8301 // Do byte swapping if necessary.
8302 if ( stream_.doByteSwap[0] )
8303 byteSwapBuffer(buffer, stream_.bufferSize * channels, format);
8305 // Write samples to device in interleaved/non-interleaved format.
8306 if ( stream_.deviceInterleaved[0] )
8307 result = snd_pcm_writei( handle[0], buffer, stream_.bufferSize );
8309 void *bufs[channels];
8310 size_t offset = stream_.bufferSize * formatBytes( format );
8311 for ( int i=0; i<channels; i++ )
8312 bufs[i] = (void *) (buffer + (i * offset));
8313 result = snd_pcm_writen( handle[0], bufs, stream_.bufferSize );
8316 if ( result < (int) stream_.bufferSize ) {
8317 // Either an error or underrun occured.
8318 if ( result == -EPIPE ) {
8319 snd_pcm_state_t state = snd_pcm_state( handle[0] );
8320 if ( state == SND_PCM_STATE_XRUN ) {
8321 apiInfo->xrun[0] = true;
8322 result = snd_pcm_prepare( handle[0] );
8324 errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after underrun, " << snd_strerror( result ) << ".";
8325 errorText_ = errorStream_.str();
8328 errorText_ = "RtApiAlsa::callbackEvent: audio write error, underrun.";
8331 errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
8332 errorText_ = errorStream_.str();
8336 errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << ".";
8337 errorText_ = errorStream_.str();
8339 error( RtAudioError::WARNING );
8343 // Check stream latency
8344 result = snd_pcm_delay( handle[0], &frames );
8345 if ( result == 0 && frames > 0 ) stream_.latency[0] = frames;
8349 MUTEX_UNLOCK( &stream_.mutex );
8351 RtApi::tickStreamTime();
8352 if ( doStopStream == 1 ) this->stopStream();
8355 static void *alsaCallbackHandler( void *ptr )
8357 CallbackInfo *info = (CallbackInfo *) ptr;
8358 RtApiAlsa *object = (RtApiAlsa *) info->object;
8359 bool *isRunning = &info->isRunning;
8361 #ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
8362 if ( info->doRealtime ) {
8363 std::cerr << "RtAudio alsa: " <<
8364 (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") <<
8365 "running realtime scheduling" << std::endl;
8369 while ( *isRunning == true ) {
8370 pthread_testcancel();
8371 object->callbackEvent();
8374 pthread_exit( NULL );
8377 //******************** End of __LINUX_ALSA__ *********************//
8380 #if defined(__LINUX_PULSE__)
8382 // Code written by Peter Meerwald, pmeerw@pmeerw.net
8383 // and Tristan Matthews.
8385 #include <pulse/error.h>
8386 #include <pulse/simple.h>
8389 static const unsigned int SUPPORTED_SAMPLERATES[] = { 8000, 16000, 22050, 32000,
8390 44100, 48000, 96000, 0};
8392 struct rtaudio_pa_format_mapping_t {
8393 RtAudioFormat rtaudio_format;
8394 pa_sample_format_t pa_format;
8397 static const rtaudio_pa_format_mapping_t supported_sampleformats[] = {
8398 {RTAUDIO_SINT16, PA_SAMPLE_S16LE},
8399 {RTAUDIO_SINT32, PA_SAMPLE_S32LE},
8400 {RTAUDIO_FLOAT32, PA_SAMPLE_FLOAT32LE},
8401 {0, PA_SAMPLE_INVALID}};
8403 struct PulseAudioHandle {
8407 pthread_cond_t runnable_cv;
8409 PulseAudioHandle() : s_play(0), s_rec(0), runnable(false) { }
8412 RtApiPulse::~RtApiPulse()
8414 if ( stream_.state != STREAM_CLOSED )
8418 unsigned int RtApiPulse::getDeviceCount( void )
8423 RtAudio::DeviceInfo RtApiPulse::getDeviceInfo( unsigned int /*device*/ )
8425 RtAudio::DeviceInfo info;
8427 info.name = "PulseAudio";
8428 info.outputChannels = 2;
8429 info.inputChannels = 2;
8430 info.duplexChannels = 2;
8431 info.isDefaultOutput = true;
8432 info.isDefaultInput = true;
8434 for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr )
8435 info.sampleRates.push_back( *sr );
8437 info.preferredSampleRate = 48000;
8438 info.nativeFormats = RTAUDIO_SINT16 | RTAUDIO_SINT32 | RTAUDIO_FLOAT32;
8443 static void *pulseaudio_callback( void * user )
8445 CallbackInfo *cbi = static_cast<CallbackInfo *>( user );
8446 RtApiPulse *context = static_cast<RtApiPulse *>( cbi->object );
8447 volatile bool *isRunning = &cbi->isRunning;
8449 #ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
8450 if (cbi->doRealtime) {
8451 std::cerr << "RtAudio pulse: " <<
8452 (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") <<
8453 "running realtime scheduling" << std::endl;
8457 while ( *isRunning ) {
8458 pthread_testcancel();
8459 context->callbackEvent();
8462 pthread_exit( NULL );
8465 void RtApiPulse::closeStream( void )
8467 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8469 stream_.callbackInfo.isRunning = false;
8471 MUTEX_LOCK( &stream_.mutex );
8472 if ( stream_.state == STREAM_STOPPED ) {
8473 pah->runnable = true;
8474 pthread_cond_signal( &pah->runnable_cv );
8476 MUTEX_UNLOCK( &stream_.mutex );
8478 pthread_join( pah->thread, 0 );
8479 if ( pah->s_play ) {
8480 pa_simple_flush( pah->s_play, NULL );
8481 pa_simple_free( pah->s_play );
8484 pa_simple_free( pah->s_rec );
8486 pthread_cond_destroy( &pah->runnable_cv );
8488 stream_.apiHandle = 0;
8491 if ( stream_.userBuffer[0] ) {
8492 free( stream_.userBuffer[0] );
8493 stream_.userBuffer[0] = 0;
8495 if ( stream_.userBuffer[1] ) {
8496 free( stream_.userBuffer[1] );
8497 stream_.userBuffer[1] = 0;
8500 stream_.state = STREAM_CLOSED;
8501 stream_.mode = UNINITIALIZED;
8504 void RtApiPulse::callbackEvent( void )
8506 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8508 if ( stream_.state == STREAM_STOPPED ) {
8509 MUTEX_LOCK( &stream_.mutex );
8510 while ( !pah->runnable )
8511 pthread_cond_wait( &pah->runnable_cv, &stream_.mutex );
8513 if ( stream_.state != STREAM_RUNNING ) {
8514 MUTEX_UNLOCK( &stream_.mutex );
8517 MUTEX_UNLOCK( &stream_.mutex );
8520 if ( stream_.state == STREAM_CLOSED ) {
8521 errorText_ = "RtApiPulse::callbackEvent(): the stream is closed ... "
8522 "this shouldn't happen!";
8523 error( RtAudioError::WARNING );
8527 RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
8528 double streamTime = getStreamTime();
8529 RtAudioStreamStatus status = 0;
8530 int doStopStream = callback( stream_.userBuffer[OUTPUT], stream_.userBuffer[INPUT],
8531 stream_.bufferSize, streamTime, status,
8532 stream_.callbackInfo.userData );
8534 if ( doStopStream == 2 ) {
8539 MUTEX_LOCK( &stream_.mutex );
8540 void *pulse_in = stream_.doConvertBuffer[INPUT] ? stream_.deviceBuffer : stream_.userBuffer[INPUT];
8541 void *pulse_out = stream_.doConvertBuffer[OUTPUT] ? stream_.deviceBuffer : stream_.userBuffer[OUTPUT];
8543 if ( stream_.state != STREAM_RUNNING )
8548 if (stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
8549 if ( stream_.doConvertBuffer[OUTPUT] ) {
8550 convertBuffer( stream_.deviceBuffer,
8551 stream_.userBuffer[OUTPUT],
8552 stream_.convertInfo[OUTPUT] );
8553 bytes = stream_.nDeviceChannels[OUTPUT] * stream_.bufferSize *
8554 formatBytes( stream_.deviceFormat[OUTPUT] );
8556 bytes = stream_.nUserChannels[OUTPUT] * stream_.bufferSize *
8557 formatBytes( stream_.userFormat );
8559 if ( pa_simple_write( pah->s_play, pulse_out, bytes, &pa_error ) < 0 ) {
8560 errorStream_ << "RtApiPulse::callbackEvent: audio write error, " <<
8561 pa_strerror( pa_error ) << ".";
8562 errorText_ = errorStream_.str();
8563 error( RtAudioError::WARNING );
8567 if ( stream_.mode == INPUT || stream_.mode == DUPLEX) {
8568 if ( stream_.doConvertBuffer[INPUT] )
8569 bytes = stream_.nDeviceChannels[INPUT] * stream_.bufferSize *
8570 formatBytes( stream_.deviceFormat[INPUT] );
8572 bytes = stream_.nUserChannels[INPUT] * stream_.bufferSize *
8573 formatBytes( stream_.userFormat );
8575 if ( pa_simple_read( pah->s_rec, pulse_in, bytes, &pa_error ) < 0 ) {
8576 errorStream_ << "RtApiPulse::callbackEvent: audio read error, " <<
8577 pa_strerror( pa_error ) << ".";
8578 errorText_ = errorStream_.str();
8579 error( RtAudioError::WARNING );
8581 if ( stream_.doConvertBuffer[INPUT] ) {
8582 convertBuffer( stream_.userBuffer[INPUT],
8583 stream_.deviceBuffer,
8584 stream_.convertInfo[INPUT] );
8589 MUTEX_UNLOCK( &stream_.mutex );
8590 RtApi::tickStreamTime();
8592 if ( doStopStream == 1 )
8596 void RtApiPulse::startStream( void )
8598 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8600 if ( stream_.state == STREAM_CLOSED ) {
8601 errorText_ = "RtApiPulse::startStream(): the stream is not open!";
8602 error( RtAudioError::INVALID_USE );
8605 if ( stream_.state == STREAM_RUNNING ) {
8606 errorText_ = "RtApiPulse::startStream(): the stream is already running!";
8607 error( RtAudioError::WARNING );
8611 MUTEX_LOCK( &stream_.mutex );
8613 stream_.state = STREAM_RUNNING;
8615 pah->runnable = true;
8616 pthread_cond_signal( &pah->runnable_cv );
8617 MUTEX_UNLOCK( &stream_.mutex );
8620 void RtApiPulse::stopStream( void )
8622 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8624 if ( stream_.state == STREAM_CLOSED ) {
8625 errorText_ = "RtApiPulse::stopStream(): the stream is not open!";
8626 error( RtAudioError::INVALID_USE );
8629 if ( stream_.state == STREAM_STOPPED ) {
8630 errorText_ = "RtApiPulse::stopStream(): the stream is already stopped!";
8631 error( RtAudioError::WARNING );
8635 stream_.state = STREAM_STOPPED;
8636 MUTEX_LOCK( &stream_.mutex );
8638 if ( pah && pah->s_play ) {
8640 if ( pa_simple_drain( pah->s_play, &pa_error ) < 0 ) {
8641 errorStream_ << "RtApiPulse::stopStream: error draining output device, " <<
8642 pa_strerror( pa_error ) << ".";
8643 errorText_ = errorStream_.str();
8644 MUTEX_UNLOCK( &stream_.mutex );
8645 error( RtAudioError::SYSTEM_ERROR );
8650 stream_.state = STREAM_STOPPED;
8651 MUTEX_UNLOCK( &stream_.mutex );
8654 void RtApiPulse::abortStream( void )
8656 PulseAudioHandle *pah = static_cast<PulseAudioHandle*>( stream_.apiHandle );
8658 if ( stream_.state == STREAM_CLOSED ) {
8659 errorText_ = "RtApiPulse::abortStream(): the stream is not open!";
8660 error( RtAudioError::INVALID_USE );
8663 if ( stream_.state == STREAM_STOPPED ) {
8664 errorText_ = "RtApiPulse::abortStream(): the stream is already stopped!";
8665 error( RtAudioError::WARNING );
8669 stream_.state = STREAM_STOPPED;
8670 MUTEX_LOCK( &stream_.mutex );
8672 if ( pah && pah->s_play ) {
8674 if ( pa_simple_flush( pah->s_play, &pa_error ) < 0 ) {
8675 errorStream_ << "RtApiPulse::abortStream: error flushing output device, " <<
8676 pa_strerror( pa_error ) << ".";
8677 errorText_ = errorStream_.str();
8678 MUTEX_UNLOCK( &stream_.mutex );
8679 error( RtAudioError::SYSTEM_ERROR );
8684 stream_.state = STREAM_STOPPED;
8685 MUTEX_UNLOCK( &stream_.mutex );
8688 bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode,
8689 unsigned int channels, unsigned int firstChannel,
8690 unsigned int sampleRate, RtAudioFormat format,
8691 unsigned int *bufferSize, RtAudio::StreamOptions *options )
8693 PulseAudioHandle *pah = 0;
8694 unsigned long bufferBytes = 0;
8697 if ( device != 0 ) return false;
8698 if ( mode != INPUT && mode != OUTPUT ) return false;
8699 if ( channels != 1 && channels != 2 ) {
8700 errorText_ = "RtApiPulse::probeDeviceOpen: unsupported number of channels.";
8703 ss.channels = channels;
8705 if ( firstChannel != 0 ) return false;
8707 bool sr_found = false;
8708 for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr ) {
8709 if ( sampleRate == *sr ) {
8711 stream_.sampleRate = sampleRate;
8712 ss.rate = sampleRate;
8717 errorText_ = "RtApiPulse::probeDeviceOpen: unsupported sample rate.";
8722 for ( const rtaudio_pa_format_mapping_t *sf = supported_sampleformats;
8723 sf->rtaudio_format && sf->pa_format != PA_SAMPLE_INVALID; ++sf ) {
8724 if ( format == sf->rtaudio_format ) {
8726 stream_.userFormat = sf->rtaudio_format;
8727 stream_.deviceFormat[mode] = stream_.userFormat;
8728 ss.format = sf->pa_format;
8732 if ( !sf_found ) { // Use internal data format conversion.
8733 stream_.userFormat = format;
8734 stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
8735 ss.format = PA_SAMPLE_FLOAT32LE;
8738 // Set other stream parameters.
8739 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
8740 else stream_.userInterleaved = true;
8741 stream_.deviceInterleaved[mode] = true;
8742 stream_.nBuffers = 1;
8743 stream_.doByteSwap[mode] = false;
8744 stream_.nUserChannels[mode] = channels;
8745 stream_.nDeviceChannels[mode] = channels + firstChannel;
8746 stream_.channelOffset[mode] = 0;
8747 std::string streamName = "RtAudio";
8749 // Set flags for buffer conversion.
8750 stream_.doConvertBuffer[mode] = false;
8751 if ( stream_.userFormat != stream_.deviceFormat[mode] )
8752 stream_.doConvertBuffer[mode] = true;
8753 if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
8754 stream_.doConvertBuffer[mode] = true;
8756 // Allocate necessary internal buffers.
8757 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
8758 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
8759 if ( stream_.userBuffer[mode] == NULL ) {
8760 errorText_ = "RtApiPulse::probeDeviceOpen: error allocating user buffer memory.";
8763 stream_.bufferSize = *bufferSize;
8765 if ( stream_.doConvertBuffer[mode] ) {
8767 bool makeBuffer = true;
8768 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
8769 if ( mode == INPUT ) {
8770 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
8771 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
8772 if ( bufferBytes <= bytesOut ) makeBuffer = false;
8777 bufferBytes *= *bufferSize;
8778 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
8779 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
8780 if ( stream_.deviceBuffer == NULL ) {
8781 errorText_ = "RtApiPulse::probeDeviceOpen: error allocating device buffer memory.";
8787 stream_.device[mode] = device;
8789 // Setup the buffer conversion information structure.
8790 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
8792 if ( !stream_.apiHandle ) {
8793 PulseAudioHandle *pah = new PulseAudioHandle;
8795 errorText_ = "RtApiPulse::probeDeviceOpen: error allocating memory for handle.";
8799 stream_.apiHandle = pah;
8800 if ( pthread_cond_init( &pah->runnable_cv, NULL ) != 0 ) {
8801 errorText_ = "RtApiPulse::probeDeviceOpen: error creating condition variable.";
8805 pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8808 if ( options && !options->streamName.empty() ) streamName = options->streamName;
8811 pa_buffer_attr buffer_attr;
8812 buffer_attr.fragsize = bufferBytes;
8813 buffer_attr.maxlength = -1;
8815 pah->s_rec = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_RECORD, NULL, "Record", &ss, NULL, &buffer_attr, &error );
8816 if ( !pah->s_rec ) {
8817 errorText_ = "RtApiPulse::probeDeviceOpen: error connecting input to PulseAudio server.";
8822 pah->s_play = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_PLAYBACK, NULL, "Playback", &ss, NULL, NULL, &error );
8823 if ( !pah->s_play ) {
8824 errorText_ = "RtApiPulse::probeDeviceOpen: error connecting output to PulseAudio server.";
8832 if ( stream_.mode == UNINITIALIZED )
8833 stream_.mode = mode;
8834 else if ( stream_.mode == mode )
8837 stream_.mode = DUPLEX;
8839 if ( !stream_.callbackInfo.isRunning ) {
8840 stream_.callbackInfo.object = this;
8842 stream_.state = STREAM_STOPPED;
8843 // Set the thread attributes for joinable and realtime scheduling
8844 // priority (optional). The higher priority will only take affect
8845 // if the program is run as root or suid. Note, under Linux
8846 // processes with CAP_SYS_NICE privilege, a user can change
8847 // scheduling policy and priority (thus need not be root). See
8848 // POSIX "capabilities".
8849 pthread_attr_t attr;
8850 pthread_attr_init( &attr );
8851 pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
8852 #ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
8853 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
8854 stream_.callbackInfo.doRealtime = true;
8855 struct sched_param param;
8856 int priority = options->priority;
8857 int min = sched_get_priority_min( SCHED_RR );
8858 int max = sched_get_priority_max( SCHED_RR );
8859 if ( priority < min ) priority = min;
8860 else if ( priority > max ) priority = max;
8861 param.sched_priority = priority;
8863 // Set the policy BEFORE the priority. Otherwise it fails.
8864 pthread_attr_setschedpolicy(&attr, SCHED_RR);
8865 pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM);
8866 // This is definitely required. Otherwise it fails.
8867 pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED);
8868 pthread_attr_setschedparam(&attr, ¶m);
8871 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
8873 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
8876 stream_.callbackInfo.isRunning = true;
8877 int result = pthread_create( &pah->thread, &attr, pulseaudio_callback, (void *)&stream_.callbackInfo);
8878 pthread_attr_destroy(&attr);
8880 // Failed. Try instead with default attributes.
8881 result = pthread_create( &pah->thread, NULL, pulseaudio_callback, (void *)&stream_.callbackInfo);
8883 stream_.callbackInfo.isRunning = false;
8884 errorText_ = "RtApiPulse::probeDeviceOpen: error creating thread.";
8893 if ( pah && stream_.callbackInfo.isRunning ) {
8894 pthread_cond_destroy( &pah->runnable_cv );
8896 stream_.apiHandle = 0;
8899 for ( int i=0; i<2; i++ ) {
8900 if ( stream_.userBuffer[i] ) {
8901 free( stream_.userBuffer[i] );
8902 stream_.userBuffer[i] = 0;
8906 if ( stream_.deviceBuffer ) {
8907 free( stream_.deviceBuffer );
8908 stream_.deviceBuffer = 0;
8911 stream_.state = STREAM_CLOSED;
8915 //******************** End of __LINUX_PULSE__ *********************//
8918 #if defined(__LINUX_OSS__)
8921 #include <sys/ioctl.h>
8924 #include <sys/soundcard.h>
8928 static void *ossCallbackHandler(void * ptr);
8930 // A structure to hold various information related to the OSS API
8933 int id[2]; // device ids
8936 pthread_cond_t runnable;
8939 :triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
8942 RtApiOss :: RtApiOss()
8944 // Nothing to do here.
8947 RtApiOss :: ~RtApiOss()
8949 if ( stream_.state != STREAM_CLOSED ) closeStream();
8952 unsigned int RtApiOss :: getDeviceCount( void )
8954 int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
8955 if ( mixerfd == -1 ) {
8956 errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'.";
8957 error( RtAudioError::WARNING );
8961 oss_sysinfo sysinfo;
8962 if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) {
8964 errorText_ = "RtApiOss::getDeviceCount: error getting sysinfo, OSS version >= 4.0 is required.";
8965 error( RtAudioError::WARNING );
8970 return sysinfo.numaudios;
8973 RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device )
8975 RtAudio::DeviceInfo info;
8976 info.probed = false;
8978 int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
8979 if ( mixerfd == -1 ) {
8980 errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'.";
8981 error( RtAudioError::WARNING );
8985 oss_sysinfo sysinfo;
8986 int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
8987 if ( result == -1 ) {
8989 errorText_ = "RtApiOss::getDeviceInfo: error getting sysinfo, OSS version >= 4.0 is required.";
8990 error( RtAudioError::WARNING );
8994 unsigned nDevices = sysinfo.numaudios;
8995 if ( nDevices == 0 ) {
8997 errorText_ = "RtApiOss::getDeviceInfo: no devices found!";
8998 error( RtAudioError::INVALID_USE );
9002 if ( device >= nDevices ) {
9004 errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!";
9005 error( RtAudioError::INVALID_USE );
9009 oss_audioinfo ainfo;
9011 result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
9013 if ( result == -1 ) {
9014 errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
9015 errorText_ = errorStream_.str();
9016 error( RtAudioError::WARNING );
9021 if ( ainfo.caps & PCM_CAP_OUTPUT ) info.outputChannels = ainfo.max_channels;
9022 if ( ainfo.caps & PCM_CAP_INPUT ) info.inputChannels = ainfo.max_channels;
9023 if ( ainfo.caps & PCM_CAP_DUPLEX ) {
9024 if ( info.outputChannels > 0 && info.inputChannels > 0 && ainfo.caps & PCM_CAP_DUPLEX )
9025 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
9028 // Probe data formats ... do for input
9029 unsigned long mask = ainfo.iformats;
9030 if ( mask & AFMT_S16_LE || mask & AFMT_S16_BE )
9031 info.nativeFormats |= RTAUDIO_SINT16;
9032 if ( mask & AFMT_S8 )
9033 info.nativeFormats |= RTAUDIO_SINT8;
9034 if ( mask & AFMT_S32_LE || mask & AFMT_S32_BE )
9035 info.nativeFormats |= RTAUDIO_SINT32;
9037 if ( mask & AFMT_FLOAT )
9038 info.nativeFormats |= RTAUDIO_FLOAT32;
9040 if ( mask & AFMT_S24_LE || mask & AFMT_S24_BE )
9041 info.nativeFormats |= RTAUDIO_SINT24;
9043 // Check that we have at least one supported format
9044 if ( info.nativeFormats == 0 ) {
9045 errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio.";
9046 errorText_ = errorStream_.str();
9047 error( RtAudioError::WARNING );
9051 // Probe the supported sample rates.
9052 info.sampleRates.clear();
9053 if ( ainfo.nrates ) {
9054 for ( unsigned int i=0; i<ainfo.nrates; i++ ) {
9055 for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
9056 if ( ainfo.rates[i] == SAMPLE_RATES[k] ) {
9057 info.sampleRates.push_back( SAMPLE_RATES[k] );
9059 if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
9060 info.preferredSampleRate = SAMPLE_RATES[k];
9068 // Check min and max rate values;
9069 for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
9070 if ( ainfo.min_rate <= (int) SAMPLE_RATES[k] && ainfo.max_rate >= (int) SAMPLE_RATES[k] ) {
9071 info.sampleRates.push_back( SAMPLE_RATES[k] );
9073 if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
9074 info.preferredSampleRate = SAMPLE_RATES[k];
9079 if ( info.sampleRates.size() == 0 ) {
9080 errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ").";
9081 errorText_ = errorStream_.str();
9082 error( RtAudioError::WARNING );
9086 info.name = ainfo.name;
9093 bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
9094 unsigned int firstChannel, unsigned int sampleRate,
9095 RtAudioFormat format, unsigned int *bufferSize,
9096 RtAudio::StreamOptions *options )
9098 int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
9099 if ( mixerfd == -1 ) {
9100 errorText_ = "RtApiOss::probeDeviceOpen: error opening '/dev/mixer'.";
9104 oss_sysinfo sysinfo;
9105 int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
9106 if ( result == -1 ) {
9108 errorText_ = "RtApiOss::probeDeviceOpen: error getting sysinfo, OSS version >= 4.0 is required.";
9112 unsigned nDevices = sysinfo.numaudios;
9113 if ( nDevices == 0 ) {
9114 // This should not happen because a check is made before this function is called.
9116 errorText_ = "RtApiOss::probeDeviceOpen: no devices found!";
9120 if ( device >= nDevices ) {
9121 // This should not happen because a check is made before this function is called.
9123 errorText_ = "RtApiOss::probeDeviceOpen: device ID is invalid!";
9127 oss_audioinfo ainfo;
9129 result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
9131 if ( result == -1 ) {
9132 errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
9133 errorText_ = errorStream_.str();
9137 // Check if device supports input or output
9138 if ( ( mode == OUTPUT && !( ainfo.caps & PCM_CAP_OUTPUT ) ) ||
9139 ( mode == INPUT && !( ainfo.caps & PCM_CAP_INPUT ) ) ) {
9140 if ( mode == OUTPUT )
9141 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support output.";
9143 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support input.";
9144 errorText_ = errorStream_.str();
9149 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9150 if ( mode == OUTPUT )
9152 else { // mode == INPUT
9153 if (stream_.mode == OUTPUT && stream_.device[0] == device) {
9154 // We just set the same device for playback ... close and reopen for duplex (OSS only).
9155 close( handle->id[0] );
9157 if ( !( ainfo.caps & PCM_CAP_DUPLEX ) ) {
9158 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support duplex mode.";
9159 errorText_ = errorStream_.str();
9162 // Check that the number previously set channels is the same.
9163 if ( stream_.nUserChannels[0] != channels ) {
9164 errorStream_ << "RtApiOss::probeDeviceOpen: input/output channels must be equal for OSS duplex device (" << ainfo.name << ").";
9165 errorText_ = errorStream_.str();
9174 // Set exclusive access if specified.
9175 if ( options && options->flags & RTAUDIO_HOG_DEVICE ) flags |= O_EXCL;
9177 // Try to open the device.
9179 fd = open( ainfo.devnode, flags, 0 );
9181 if ( errno == EBUSY )
9182 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") is busy.";
9184 errorStream_ << "RtApiOss::probeDeviceOpen: error opening device (" << ainfo.name << ").";
9185 errorText_ = errorStream_.str();
9189 // For duplex operation, specifically set this mode (this doesn't seem to work).
9191 if ( flags | O_RDWR ) {
9192 result = ioctl( fd, SNDCTL_DSP_SETDUPLEX, NULL );
9193 if ( result == -1) {
9194 errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ").";
9195 errorText_ = errorStream_.str();
9201 // Check the device channel support.
9202 stream_.nUserChannels[mode] = channels;
9203 if ( ainfo.max_channels < (int)(channels + firstChannel) ) {
9205 errorStream_ << "RtApiOss::probeDeviceOpen: the device (" << ainfo.name << ") does not support requested channel parameters.";
9206 errorText_ = errorStream_.str();
9210 // Set the number of channels.
9211 int deviceChannels = channels + firstChannel;
9212 result = ioctl( fd, SNDCTL_DSP_CHANNELS, &deviceChannels );
9213 if ( result == -1 || deviceChannels < (int)(channels + firstChannel) ) {
9215 errorStream_ << "RtApiOss::probeDeviceOpen: error setting channel parameters on device (" << ainfo.name << ").";
9216 errorText_ = errorStream_.str();
9219 stream_.nDeviceChannels[mode] = deviceChannels;
9221 // Get the data format mask
9223 result = ioctl( fd, SNDCTL_DSP_GETFMTS, &mask );
9224 if ( result == -1 ) {
9226 errorStream_ << "RtApiOss::probeDeviceOpen: error getting device (" << ainfo.name << ") data formats.";
9227 errorText_ = errorStream_.str();
9231 // Determine how to set the device format.
9232 stream_.userFormat = format;
9233 int deviceFormat = -1;
9234 stream_.doByteSwap[mode] = false;
9235 if ( format == RTAUDIO_SINT8 ) {
9236 if ( mask & AFMT_S8 ) {
9237 deviceFormat = AFMT_S8;
9238 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
9241 else if ( format == RTAUDIO_SINT16 ) {
9242 if ( mask & AFMT_S16_NE ) {
9243 deviceFormat = AFMT_S16_NE;
9244 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
9246 else if ( mask & AFMT_S16_OE ) {
9247 deviceFormat = AFMT_S16_OE;
9248 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
9249 stream_.doByteSwap[mode] = true;
9252 else if ( format == RTAUDIO_SINT24 ) {
9253 if ( mask & AFMT_S24_NE ) {
9254 deviceFormat = AFMT_S24_NE;
9255 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
9257 else if ( mask & AFMT_S24_OE ) {
9258 deviceFormat = AFMT_S24_OE;
9259 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
9260 stream_.doByteSwap[mode] = true;
9263 else if ( format == RTAUDIO_SINT32 ) {
9264 if ( mask & AFMT_S32_NE ) {
9265 deviceFormat = AFMT_S32_NE;
9266 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
9268 else if ( mask & AFMT_S32_OE ) {
9269 deviceFormat = AFMT_S32_OE;
9270 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
9271 stream_.doByteSwap[mode] = true;
9275 if ( deviceFormat == -1 ) {
9276 // The user requested format is not natively supported by the device.
9277 if ( mask & AFMT_S16_NE ) {
9278 deviceFormat = AFMT_S16_NE;
9279 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
9281 else if ( mask & AFMT_S32_NE ) {
9282 deviceFormat = AFMT_S32_NE;
9283 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
9285 else if ( mask & AFMT_S24_NE ) {
9286 deviceFormat = AFMT_S24_NE;
9287 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
9289 else if ( mask & AFMT_S16_OE ) {
9290 deviceFormat = AFMT_S16_OE;
9291 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
9292 stream_.doByteSwap[mode] = true;
9294 else if ( mask & AFMT_S32_OE ) {
9295 deviceFormat = AFMT_S32_OE;
9296 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
9297 stream_.doByteSwap[mode] = true;
9299 else if ( mask & AFMT_S24_OE ) {
9300 deviceFormat = AFMT_S24_OE;
9301 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
9302 stream_.doByteSwap[mode] = true;
9304 else if ( mask & AFMT_S8) {
9305 deviceFormat = AFMT_S8;
9306 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
9310 if ( stream_.deviceFormat[mode] == 0 ) {
9311 // This really shouldn't happen ...
9313 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") data format not supported by RtAudio.";
9314 errorText_ = errorStream_.str();
9318 // Set the data format.
9319 int temp = deviceFormat;
9320 result = ioctl( fd, SNDCTL_DSP_SETFMT, &deviceFormat );
9321 if ( result == -1 || deviceFormat != temp ) {
9323 errorStream_ << "RtApiOss::probeDeviceOpen: error setting data format on device (" << ainfo.name << ").";
9324 errorText_ = errorStream_.str();
9328 // Attempt to set the buffer size. According to OSS, the minimum
9329 // number of buffers is two. The supposed minimum buffer size is 16
9330 // bytes, so that will be our lower bound. The argument to this
9331 // call is in the form 0xMMMMSSSS (hex), where the buffer size (in
9332 // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM.
9333 // We'll check the actual value used near the end of the setup
9335 int ossBufferBytes = *bufferSize * formatBytes( stream_.deviceFormat[mode] ) * deviceChannels;
9336 if ( ossBufferBytes < 16 ) ossBufferBytes = 16;
9338 if ( options ) buffers = options->numberOfBuffers;
9339 if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) buffers = 2;
9340 if ( buffers < 2 ) buffers = 3;
9341 temp = ((int) buffers << 16) + (int)( log10( (double)ossBufferBytes ) / log10( 2.0 ) );
9342 result = ioctl( fd, SNDCTL_DSP_SETFRAGMENT, &temp );
9343 if ( result == -1 ) {
9345 errorStream_ << "RtApiOss::probeDeviceOpen: error setting buffer size on device (" << ainfo.name << ").";
9346 errorText_ = errorStream_.str();
9349 stream_.nBuffers = buffers;
9351 // Save buffer size (in sample frames).
9352 *bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels );
9353 stream_.bufferSize = *bufferSize;
9355 // Set the sample rate.
9356 int srate = sampleRate;
9357 result = ioctl( fd, SNDCTL_DSP_SPEED, &srate );
9358 if ( result == -1 ) {
9360 errorStream_ << "RtApiOss::probeDeviceOpen: error setting sample rate (" << sampleRate << ") on device (" << ainfo.name << ").";
9361 errorText_ = errorStream_.str();
9365 // Verify the sample rate setup worked.
9366 if ( abs( srate - (int)sampleRate ) > 100 ) {
9368 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ").";
9369 errorText_ = errorStream_.str();
9372 stream_.sampleRate = sampleRate;
9374 if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device) {
9375 // We're doing duplex setup here.
9376 stream_.deviceFormat[0] = stream_.deviceFormat[1];
9377 stream_.nDeviceChannels[0] = deviceChannels;
9380 // Set interleaving parameters.
9381 stream_.userInterleaved = true;
9382 stream_.deviceInterleaved[mode] = true;
9383 if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
9384 stream_.userInterleaved = false;
9386 // Set flags for buffer conversion
9387 stream_.doConvertBuffer[mode] = false;
9388 if ( stream_.userFormat != stream_.deviceFormat[mode] )
9389 stream_.doConvertBuffer[mode] = true;
9390 if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
9391 stream_.doConvertBuffer[mode] = true;
9392 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
9393 stream_.nUserChannels[mode] > 1 )
9394 stream_.doConvertBuffer[mode] = true;
9396 // Allocate the stream handles if necessary and then save.
9397 if ( stream_.apiHandle == 0 ) {
9399 handle = new OssHandle;
9401 catch ( std::bad_alloc& ) {
9402 errorText_ = "RtApiOss::probeDeviceOpen: error allocating OssHandle memory.";
9406 if ( pthread_cond_init( &handle->runnable, NULL ) ) {
9407 errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable.";
9411 stream_.apiHandle = (void *) handle;
9414 handle = (OssHandle *) stream_.apiHandle;
9416 handle->id[mode] = fd;
9418 // Allocate necessary internal buffers.
9419 unsigned long bufferBytes;
9420 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
9421 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
9422 if ( stream_.userBuffer[mode] == NULL ) {
9423 errorText_ = "RtApiOss::probeDeviceOpen: error allocating user buffer memory.";
9427 if ( stream_.doConvertBuffer[mode] ) {
9429 bool makeBuffer = true;
9430 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
9431 if ( mode == INPUT ) {
9432 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
9433 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
9434 if ( bufferBytes <= bytesOut ) makeBuffer = false;
9439 bufferBytes *= *bufferSize;
9440 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
9441 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
9442 if ( stream_.deviceBuffer == NULL ) {
9443 errorText_ = "RtApiOss::probeDeviceOpen: error allocating device buffer memory.";
9449 stream_.device[mode] = device;
9450 stream_.state = STREAM_STOPPED;
9452 // Setup the buffer conversion information structure.
9453 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
9455 // Setup thread if necessary.
9456 if ( stream_.mode == OUTPUT && mode == INPUT ) {
9457 // We had already set up an output stream.
9458 stream_.mode = DUPLEX;
9459 if ( stream_.device[0] == device ) handle->id[0] = fd;
9462 stream_.mode = mode;
9464 // Setup callback thread.
9465 stream_.callbackInfo.object = (void *) this;
9467 // Set the thread attributes for joinable and realtime scheduling
9468 // priority. The higher priority will only take affect if the
9469 // program is run as root or suid.
9470 pthread_attr_t attr;
9471 pthread_attr_init( &attr );
9472 pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
9473 #ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
9474 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
9475 stream_.callbackInfo.doRealtime = true;
9476 struct sched_param param;
9477 int priority = options->priority;
9478 int min = sched_get_priority_min( SCHED_RR );
9479 int max = sched_get_priority_max( SCHED_RR );
9480 if ( priority < min ) priority = min;
9481 else if ( priority > max ) priority = max;
9482 param.sched_priority = priority;
9484 // Set the policy BEFORE the priority. Otherwise it fails.
9485 pthread_attr_setschedpolicy(&attr, SCHED_RR);
9486 pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM);
9487 // This is definitely required. Otherwise it fails.
9488 pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED);
9489 pthread_attr_setschedparam(&attr, ¶m);
9492 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
9494 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
9497 stream_.callbackInfo.isRunning = true;
9498 result = pthread_create( &stream_.callbackInfo.thread, &attr, ossCallbackHandler, &stream_.callbackInfo );
9499 pthread_attr_destroy( &attr );
9501 // Failed. Try instead with default attributes.
9502 result = pthread_create( &stream_.callbackInfo.thread, NULL, ossCallbackHandler, &stream_.callbackInfo );
9504 stream_.callbackInfo.isRunning = false;
9505 errorText_ = "RtApiOss::error creating callback thread!";
9515 pthread_cond_destroy( &handle->runnable );
9516 if ( handle->id[0] ) close( handle->id[0] );
9517 if ( handle->id[1] ) close( handle->id[1] );
9519 stream_.apiHandle = 0;
9522 for ( int i=0; i<2; i++ ) {
9523 if ( stream_.userBuffer[i] ) {
9524 free( stream_.userBuffer[i] );
9525 stream_.userBuffer[i] = 0;
9529 if ( stream_.deviceBuffer ) {
9530 free( stream_.deviceBuffer );
9531 stream_.deviceBuffer = 0;
9534 stream_.state = STREAM_CLOSED;
9538 void RtApiOss :: closeStream()
9540 if ( stream_.state == STREAM_CLOSED ) {
9541 errorText_ = "RtApiOss::closeStream(): no open stream to close!";
9542 error( RtAudioError::WARNING );
9546 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9547 stream_.callbackInfo.isRunning = false;
9548 MUTEX_LOCK( &stream_.mutex );
9549 if ( stream_.state == STREAM_STOPPED )
9550 pthread_cond_signal( &handle->runnable );
9551 MUTEX_UNLOCK( &stream_.mutex );
9552 pthread_join( stream_.callbackInfo.thread, NULL );
9554 if ( stream_.state == STREAM_RUNNING ) {
9555 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
9556 ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
9558 ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
9559 stream_.state = STREAM_STOPPED;
9563 pthread_cond_destroy( &handle->runnable );
9564 if ( handle->id[0] ) close( handle->id[0] );
9565 if ( handle->id[1] ) close( handle->id[1] );
9567 stream_.apiHandle = 0;
9570 for ( int i=0; i<2; i++ ) {
9571 if ( stream_.userBuffer[i] ) {
9572 free( stream_.userBuffer[i] );
9573 stream_.userBuffer[i] = 0;
9577 if ( stream_.deviceBuffer ) {
9578 free( stream_.deviceBuffer );
9579 stream_.deviceBuffer = 0;
9582 stream_.mode = UNINITIALIZED;
9583 stream_.state = STREAM_CLOSED;
9586 void RtApiOss :: startStream()
9589 if ( stream_.state == STREAM_RUNNING ) {
9590 errorText_ = "RtApiOss::startStream(): the stream is already running!";
9591 error( RtAudioError::WARNING );
9595 MUTEX_LOCK( &stream_.mutex );
9597 stream_.state = STREAM_RUNNING;
9599 // No need to do anything else here ... OSS automatically starts
9600 // when fed samples.
9602 MUTEX_UNLOCK( &stream_.mutex );
9604 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9605 pthread_cond_signal( &handle->runnable );
9608 void RtApiOss :: stopStream()
9611 if ( stream_.state == STREAM_STOPPED ) {
9612 errorText_ = "RtApiOss::stopStream(): the stream is already stopped!";
9613 error( RtAudioError::WARNING );
9617 MUTEX_LOCK( &stream_.mutex );
9619 // The state might change while waiting on a mutex.
9620 if ( stream_.state == STREAM_STOPPED ) {
9621 MUTEX_UNLOCK( &stream_.mutex );
9626 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9627 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
9629 // Flush the output with zeros a few times.
9632 RtAudioFormat format;
9634 if ( stream_.doConvertBuffer[0] ) {
9635 buffer = stream_.deviceBuffer;
9636 samples = stream_.bufferSize * stream_.nDeviceChannels[0];
9637 format = stream_.deviceFormat[0];
9640 buffer = stream_.userBuffer[0];
9641 samples = stream_.bufferSize * stream_.nUserChannels[0];
9642 format = stream_.userFormat;
9645 memset( buffer, 0, samples * formatBytes(format) );
9646 for ( unsigned int i=0; i<stream_.nBuffers+1; i++ ) {
9647 result = write( handle->id[0], buffer, samples * formatBytes(format) );
9648 if ( result == -1 ) {
9649 errorText_ = "RtApiOss::stopStream: audio write error.";
9650 error( RtAudioError::WARNING );
9654 result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
9655 if ( result == -1 ) {
9656 errorStream_ << "RtApiOss::stopStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
9657 errorText_ = errorStream_.str();
9660 handle->triggered = false;
9663 if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
9664 result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
9665 if ( result == -1 ) {
9666 errorStream_ << "RtApiOss::stopStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
9667 errorText_ = errorStream_.str();
9673 stream_.state = STREAM_STOPPED;
9674 MUTEX_UNLOCK( &stream_.mutex );
9676 if ( result != -1 ) return;
9677 error( RtAudioError::SYSTEM_ERROR );
9680 void RtApiOss :: abortStream()
9683 if ( stream_.state == STREAM_STOPPED ) {
9684 errorText_ = "RtApiOss::abortStream(): the stream is already stopped!";
9685 error( RtAudioError::WARNING );
9689 MUTEX_LOCK( &stream_.mutex );
9691 // The state might change while waiting on a mutex.
9692 if ( stream_.state == STREAM_STOPPED ) {
9693 MUTEX_UNLOCK( &stream_.mutex );
9698 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9699 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
9700 result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
9701 if ( result == -1 ) {
9702 errorStream_ << "RtApiOss::abortStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
9703 errorText_ = errorStream_.str();
9706 handle->triggered = false;
9709 if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
9710 result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
9711 if ( result == -1 ) {
9712 errorStream_ << "RtApiOss::abortStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
9713 errorText_ = errorStream_.str();
9719 stream_.state = STREAM_STOPPED;
9720 MUTEX_UNLOCK( &stream_.mutex );
9722 if ( result != -1 ) return;
9723 error( RtAudioError::SYSTEM_ERROR );
9726 void RtApiOss :: callbackEvent()
9728 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9729 if ( stream_.state == STREAM_STOPPED ) {
9730 MUTEX_LOCK( &stream_.mutex );
9731 pthread_cond_wait( &handle->runnable, &stream_.mutex );
9732 if ( stream_.state != STREAM_RUNNING ) {
9733 MUTEX_UNLOCK( &stream_.mutex );
9736 MUTEX_UNLOCK( &stream_.mutex );
9739 if ( stream_.state == STREAM_CLOSED ) {
9740 errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!";
9741 error( RtAudioError::WARNING );
9745 // Invoke user callback to get fresh output data.
9746 int doStopStream = 0;
9747 RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
9748 double streamTime = getStreamTime();
9749 RtAudioStreamStatus status = 0;
9750 if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
9751 status |= RTAUDIO_OUTPUT_UNDERFLOW;
9752 handle->xrun[0] = false;
9754 if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
9755 status |= RTAUDIO_INPUT_OVERFLOW;
9756 handle->xrun[1] = false;
9758 doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
9759 stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
9760 if ( doStopStream == 2 ) {
9761 this->abortStream();
9765 MUTEX_LOCK( &stream_.mutex );
9767 // The state might change while waiting on a mutex.
9768 if ( stream_.state == STREAM_STOPPED ) goto unlock;
9773 RtAudioFormat format;
9775 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
9777 // Setup parameters and do buffer conversion if necessary.
9778 if ( stream_.doConvertBuffer[0] ) {
9779 buffer = stream_.deviceBuffer;
9780 convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
9781 samples = stream_.bufferSize * stream_.nDeviceChannels[0];
9782 format = stream_.deviceFormat[0];
9785 buffer = stream_.userBuffer[0];
9786 samples = stream_.bufferSize * stream_.nUserChannels[0];
9787 format = stream_.userFormat;
9790 // Do byte swapping if necessary.
9791 if ( stream_.doByteSwap[0] )
9792 byteSwapBuffer( buffer, samples, format );
9794 if ( stream_.mode == DUPLEX && handle->triggered == false ) {
9796 ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
9797 result = write( handle->id[0], buffer, samples * formatBytes(format) );
9798 trig = PCM_ENABLE_INPUT|PCM_ENABLE_OUTPUT;
9799 ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
9800 handle->triggered = true;
9803 // Write samples to device.
9804 result = write( handle->id[0], buffer, samples * formatBytes(format) );
9806 if ( result == -1 ) {
9807 // We'll assume this is an underrun, though there isn't a
9808 // specific means for determining that.
9809 handle->xrun[0] = true;
9810 errorText_ = "RtApiOss::callbackEvent: audio write error.";
9811 error( RtAudioError::WARNING );
9812 // Continue on to input section.
9816 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
9818 // Setup parameters.
9819 if ( stream_.doConvertBuffer[1] ) {
9820 buffer = stream_.deviceBuffer;
9821 samples = stream_.bufferSize * stream_.nDeviceChannels[1];
9822 format = stream_.deviceFormat[1];
9825 buffer = stream_.userBuffer[1];
9826 samples = stream_.bufferSize * stream_.nUserChannels[1];
9827 format = stream_.userFormat;
9830 // Read samples from device.
9831 result = read( handle->id[1], buffer, samples * formatBytes(format) );
9833 if ( result == -1 ) {
9834 // We'll assume this is an overrun, though there isn't a
9835 // specific means for determining that.
9836 handle->xrun[1] = true;
9837 errorText_ = "RtApiOss::callbackEvent: audio read error.";
9838 error( RtAudioError::WARNING );
9842 // Do byte swapping if necessary.
9843 if ( stream_.doByteSwap[1] )
9844 byteSwapBuffer( buffer, samples, format );
9846 // Do buffer conversion if necessary.
9847 if ( stream_.doConvertBuffer[1] )
9848 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
9852 MUTEX_UNLOCK( &stream_.mutex );
9854 RtApi::tickStreamTime();
9855 if ( doStopStream == 1 ) this->stopStream();
9858 static void *ossCallbackHandler( void *ptr )
9860 CallbackInfo *info = (CallbackInfo *) ptr;
9861 RtApiOss *object = (RtApiOss *) info->object;
9862 bool *isRunning = &info->isRunning;
9864 #ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
9865 if (info->doRealtime) {
9866 std::cerr << "RtAudio oss: " <<
9867 (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") <<
9868 "running realtime scheduling" << std::endl;
9872 while ( *isRunning == true ) {
9873 pthread_testcancel();
9874 object->callbackEvent();
9877 pthread_exit( NULL );
9880 //******************** End of __LINUX_OSS__ *********************//
9884 // *************************************************** //
9886 // Protected common (OS-independent) RtAudio methods.
9888 // *************************************************** //
9890 // This method can be modified to control the behavior of error
9891 // message printing.
9892 void RtApi :: error( RtAudioError::Type type )
9894 errorStream_.str(""); // clear the ostringstream
9896 RtAudioErrorCallback errorCallback = (RtAudioErrorCallback) stream_.callbackInfo.errorCallback;
9897 if ( errorCallback ) {
9898 // abortStream() can generate new error messages. Ignore them. Just keep original one.
9900 if ( firstErrorOccurred_ )
9903 firstErrorOccurred_ = true;
9904 const std::string errorMessage = errorText_;
9906 if ( type != RtAudioError::WARNING && stream_.state != STREAM_STOPPED) {
9907 stream_.callbackInfo.isRunning = false; // exit from the thread
9911 errorCallback( type, errorMessage );
9912 firstErrorOccurred_ = false;
9916 if ( type == RtAudioError::WARNING && showWarnings_ == true )
9917 std::cerr << '\n' << errorText_ << "\n\n";
9918 else if ( type != RtAudioError::WARNING )
9919 throw( RtAudioError( errorText_, type ) );
9922 void RtApi :: verifyStream()
9924 if ( stream_.state == STREAM_CLOSED ) {
9925 errorText_ = "RtApi:: a stream is not open!";
9926 error( RtAudioError::INVALID_USE );
9930 void RtApi :: clearStreamInfo()
9932 stream_.mode = UNINITIALIZED;
9933 stream_.state = STREAM_CLOSED;
9934 stream_.sampleRate = 0;
9935 stream_.bufferSize = 0;
9936 stream_.nBuffers = 0;
9937 stream_.userFormat = 0;
9938 stream_.userInterleaved = true;
9939 stream_.streamTime = 0.0;
9940 stream_.apiHandle = 0;
9941 stream_.deviceBuffer = 0;
9942 stream_.callbackInfo.callback = 0;
9943 stream_.callbackInfo.userData = 0;
9944 stream_.callbackInfo.isRunning = false;
9945 stream_.callbackInfo.errorCallback = 0;
9946 for ( int i=0; i<2; i++ ) {
9947 stream_.device[i] = 11111;
9948 stream_.doConvertBuffer[i] = false;
9949 stream_.deviceInterleaved[i] = true;
9950 stream_.doByteSwap[i] = false;
9951 stream_.nUserChannels[i] = 0;
9952 stream_.nDeviceChannels[i] = 0;
9953 stream_.channelOffset[i] = 0;
9954 stream_.deviceFormat[i] = 0;
9955 stream_.latency[i] = 0;
9956 stream_.userBuffer[i] = 0;
9957 stream_.convertInfo[i].channels = 0;
9958 stream_.convertInfo[i].inJump = 0;
9959 stream_.convertInfo[i].outJump = 0;
9960 stream_.convertInfo[i].inFormat = 0;
9961 stream_.convertInfo[i].outFormat = 0;
9962 stream_.convertInfo[i].inOffset.clear();
9963 stream_.convertInfo[i].outOffset.clear();
9967 unsigned int RtApi :: formatBytes( RtAudioFormat format )
9969 if ( format == RTAUDIO_SINT16 )
9971 else if ( format == RTAUDIO_SINT32 || format == RTAUDIO_FLOAT32 )
9973 else if ( format == RTAUDIO_FLOAT64 )
9975 else if ( format == RTAUDIO_SINT24 )
9977 else if ( format == RTAUDIO_SINT8 )
9980 errorText_ = "RtApi::formatBytes: undefined format.";
9981 error( RtAudioError::WARNING );
9986 void RtApi :: setConvertInfo( StreamMode mode, unsigned int firstChannel )
9988 if ( mode == INPUT ) { // convert device to user buffer
9989 stream_.convertInfo[mode].inJump = stream_.nDeviceChannels[1];
9990 stream_.convertInfo[mode].outJump = stream_.nUserChannels[1];
9991 stream_.convertInfo[mode].inFormat = stream_.deviceFormat[1];
9992 stream_.convertInfo[mode].outFormat = stream_.userFormat;
9994 else { // convert user to device buffer
9995 stream_.convertInfo[mode].inJump = stream_.nUserChannels[0];
9996 stream_.convertInfo[mode].outJump = stream_.nDeviceChannels[0];
9997 stream_.convertInfo[mode].inFormat = stream_.userFormat;
9998 stream_.convertInfo[mode].outFormat = stream_.deviceFormat[0];
10001 if ( stream_.convertInfo[mode].inJump < stream_.convertInfo[mode].outJump )
10002 stream_.convertInfo[mode].channels = stream_.convertInfo[mode].inJump;
10004 stream_.convertInfo[mode].channels = stream_.convertInfo[mode].outJump;
10006 // Set up the interleave/deinterleave offsets.
10007 if ( stream_.deviceInterleaved[mode] != stream_.userInterleaved ) {
10008 if ( ( mode == OUTPUT && stream_.deviceInterleaved[mode] ) ||
10009 ( mode == INPUT && stream_.userInterleaved ) ) {
10010 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
10011 stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
10012 stream_.convertInfo[mode].outOffset.push_back( k );
10013 stream_.convertInfo[mode].inJump = 1;
10017 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
10018 stream_.convertInfo[mode].inOffset.push_back( k );
10019 stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
10020 stream_.convertInfo[mode].outJump = 1;
10024 else { // no (de)interleaving
10025 if ( stream_.userInterleaved ) {
10026 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
10027 stream_.convertInfo[mode].inOffset.push_back( k );
10028 stream_.convertInfo[mode].outOffset.push_back( k );
10032 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
10033 stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
10034 stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
10035 stream_.convertInfo[mode].inJump = 1;
10036 stream_.convertInfo[mode].outJump = 1;
10041 // Add channel offset.
10042 if ( firstChannel > 0 ) {
10043 if ( stream_.deviceInterleaved[mode] ) {
10044 if ( mode == OUTPUT ) {
10045 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
10046 stream_.convertInfo[mode].outOffset[k] += firstChannel;
10049 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
10050 stream_.convertInfo[mode].inOffset[k] += firstChannel;
10054 if ( mode == OUTPUT ) {
10055 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
10056 stream_.convertInfo[mode].outOffset[k] += ( firstChannel * stream_.bufferSize );
10059 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
10060 stream_.convertInfo[mode].inOffset[k] += ( firstChannel * stream_.bufferSize );
10066 void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info )
10068 // This function does format conversion, input/output channel compensation, and
10069 // data interleaving/deinterleaving. 24-bit integers are assumed to occupy
10070 // the lower three bytes of a 32-bit integer.
10072 // Clear our device buffer when in/out duplex device channels are different
10073 if ( outBuffer == stream_.deviceBuffer && stream_.mode == DUPLEX &&
10074 ( stream_.nDeviceChannels[0] < stream_.nDeviceChannels[1] ) )
10075 memset( outBuffer, 0, stream_.bufferSize * info.outJump * formatBytes( info.outFormat ) );
10078 if (info.outFormat == RTAUDIO_FLOAT64) {
10080 Float64 *out = (Float64 *)outBuffer;
10082 if (info.inFormat == RTAUDIO_SINT8) {
10083 signed char *in = (signed char *)inBuffer;
10084 scale = 1.0 / 127.5;
10085 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10086 for (j=0; j<info.channels; j++) {
10087 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
10088 out[info.outOffset[j]] += 0.5;
10089 out[info.outOffset[j]] *= scale;
10092 out += info.outJump;
10095 else if (info.inFormat == RTAUDIO_SINT16) {
10096 Int16 *in = (Int16 *)inBuffer;
10097 scale = 1.0 / 32767.5;
10098 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10099 for (j=0; j<info.channels; j++) {
10100 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
10101 out[info.outOffset[j]] += 0.5;
10102 out[info.outOffset[j]] *= scale;
10105 out += info.outJump;
10108 else if (info.inFormat == RTAUDIO_SINT24) {
10109 Int24 *in = (Int24 *)inBuffer;
10110 scale = 1.0 / 8388607.5;
10111 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10112 for (j=0; j<info.channels; j++) {
10113 out[info.outOffset[j]] = (Float64) (in[info.inOffset[j]].asInt());
10114 out[info.outOffset[j]] += 0.5;
10115 out[info.outOffset[j]] *= scale;
10118 out += info.outJump;
10121 else if (info.inFormat == RTAUDIO_SINT32) {
10122 Int32 *in = (Int32 *)inBuffer;
10123 scale = 1.0 / 2147483647.5;
10124 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10125 for (j=0; j<info.channels; j++) {
10126 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
10127 out[info.outOffset[j]] += 0.5;
10128 out[info.outOffset[j]] *= scale;
10131 out += info.outJump;
10134 else if (info.inFormat == RTAUDIO_FLOAT32) {
10135 Float32 *in = (Float32 *)inBuffer;
10136 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10137 for (j=0; j<info.channels; j++) {
10138 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
10141 out += info.outJump;
10144 else if (info.inFormat == RTAUDIO_FLOAT64) {
10145 // Channel compensation and/or (de)interleaving only.
10146 Float64 *in = (Float64 *)inBuffer;
10147 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10148 for (j=0; j<info.channels; j++) {
10149 out[info.outOffset[j]] = in[info.inOffset[j]];
10152 out += info.outJump;
10156 else if (info.outFormat == RTAUDIO_FLOAT32) {
10158 Float32 *out = (Float32 *)outBuffer;
10160 if (info.inFormat == RTAUDIO_SINT8) {
10161 signed char *in = (signed char *)inBuffer;
10162 scale = (Float32) ( 1.0 / 127.5 );
10163 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10164 for (j=0; j<info.channels; j++) {
10165 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
10166 out[info.outOffset[j]] += 0.5;
10167 out[info.outOffset[j]] *= scale;
10170 out += info.outJump;
10173 else if (info.inFormat == RTAUDIO_SINT16) {
10174 Int16 *in = (Int16 *)inBuffer;
10175 scale = (Float32) ( 1.0 / 32767.5 );
10176 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10177 for (j=0; j<info.channels; j++) {
10178 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
10179 out[info.outOffset[j]] += 0.5;
10180 out[info.outOffset[j]] *= scale;
10183 out += info.outJump;
10186 else if (info.inFormat == RTAUDIO_SINT24) {
10187 Int24 *in = (Int24 *)inBuffer;
10188 scale = (Float32) ( 1.0 / 8388607.5 );
10189 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10190 for (j=0; j<info.channels; j++) {
10191 out[info.outOffset[j]] = (Float32) (in[info.inOffset[j]].asInt());
10192 out[info.outOffset[j]] += 0.5;
10193 out[info.outOffset[j]] *= scale;
10196 out += info.outJump;
10199 else if (info.inFormat == RTAUDIO_SINT32) {
10200 Int32 *in = (Int32 *)inBuffer;
10201 scale = (Float32) ( 1.0 / 2147483647.5 );
10202 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10203 for (j=0; j<info.channels; j++) {
10204 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
10205 out[info.outOffset[j]] += 0.5;
10206 out[info.outOffset[j]] *= scale;
10209 out += info.outJump;
10212 else if (info.inFormat == RTAUDIO_FLOAT32) {
10213 // Channel compensation and/or (de)interleaving only.
10214 Float32 *in = (Float32 *)inBuffer;
10215 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10216 for (j=0; j<info.channels; j++) {
10217 out[info.outOffset[j]] = in[info.inOffset[j]];
10220 out += info.outJump;
10223 else if (info.inFormat == RTAUDIO_FLOAT64) {
10224 Float64 *in = (Float64 *)inBuffer;
10225 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10226 for (j=0; j<info.channels; j++) {
10227 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
10230 out += info.outJump;
10234 else if (info.outFormat == RTAUDIO_SINT32) {
10235 Int32 *out = (Int32 *)outBuffer;
10236 if (info.inFormat == RTAUDIO_SINT8) {
10237 signed char *in = (signed char *)inBuffer;
10238 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10239 for (j=0; j<info.channels; j++) {
10240 out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
10241 out[info.outOffset[j]] <<= 24;
10244 out += info.outJump;
10247 else if (info.inFormat == RTAUDIO_SINT16) {
10248 Int16 *in = (Int16 *)inBuffer;
10249 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10250 for (j=0; j<info.channels; j++) {
10251 out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
10252 out[info.outOffset[j]] <<= 16;
10255 out += info.outJump;
10258 else if (info.inFormat == RTAUDIO_SINT24) {
10259 Int24 *in = (Int24 *)inBuffer;
10260 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10261 for (j=0; j<info.channels; j++) {
10262 out[info.outOffset[j]] = (Int32) in[info.inOffset[j]].asInt();
10263 out[info.outOffset[j]] <<= 8;
10266 out += info.outJump;
10269 else if (info.inFormat == RTAUDIO_SINT32) {
10270 // Channel compensation and/or (de)interleaving only.
10271 Int32 *in = (Int32 *)inBuffer;
10272 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10273 for (j=0; j<info.channels; j++) {
10274 out[info.outOffset[j]] = in[info.inOffset[j]];
10277 out += info.outJump;
10280 else if (info.inFormat == RTAUDIO_FLOAT32) {
10281 Float32 *in = (Float32 *)inBuffer;
10282 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10283 for (j=0; j<info.channels; j++) {
10284 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
10287 out += info.outJump;
10290 else if (info.inFormat == RTAUDIO_FLOAT64) {
10291 Float64 *in = (Float64 *)inBuffer;
10292 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10293 for (j=0; j<info.channels; j++) {
10294 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
10297 out += info.outJump;
10301 else if (info.outFormat == RTAUDIO_SINT24) {
10302 Int24 *out = (Int24 *)outBuffer;
10303 if (info.inFormat == RTAUDIO_SINT8) {
10304 signed char *in = (signed char *)inBuffer;
10305 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10306 for (j=0; j<info.channels; j++) {
10307 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] << 16);
10308 //out[info.outOffset[j]] <<= 16;
10311 out += info.outJump;
10314 else if (info.inFormat == RTAUDIO_SINT16) {
10315 Int16 *in = (Int16 *)inBuffer;
10316 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10317 for (j=0; j<info.channels; j++) {
10318 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] << 8);
10319 //out[info.outOffset[j]] <<= 8;
10322 out += info.outJump;
10325 else if (info.inFormat == RTAUDIO_SINT24) {
10326 // Channel compensation and/or (de)interleaving only.
10327 Int24 *in = (Int24 *)inBuffer;
10328 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10329 for (j=0; j<info.channels; j++) {
10330 out[info.outOffset[j]] = in[info.inOffset[j]];
10333 out += info.outJump;
10336 else if (info.inFormat == RTAUDIO_SINT32) {
10337 Int32 *in = (Int32 *)inBuffer;
10338 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10339 for (j=0; j<info.channels; j++) {
10340 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] >> 8);
10341 //out[info.outOffset[j]] >>= 8;
10344 out += info.outJump;
10347 else if (info.inFormat == RTAUDIO_FLOAT32) {
10348 Float32 *in = (Float32 *)inBuffer;
10349 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10350 for (j=0; j<info.channels; j++) {
10351 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
10354 out += info.outJump;
10357 else if (info.inFormat == RTAUDIO_FLOAT64) {
10358 Float64 *in = (Float64 *)inBuffer;
10359 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10360 for (j=0; j<info.channels; j++) {
10361 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
10364 out += info.outJump;
10368 else if (info.outFormat == RTAUDIO_SINT16) {
10369 Int16 *out = (Int16 *)outBuffer;
10370 if (info.inFormat == RTAUDIO_SINT8) {
10371 signed char *in = (signed char *)inBuffer;
10372 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10373 for (j=0; j<info.channels; j++) {
10374 out[info.outOffset[j]] = (Int16) in[info.inOffset[j]];
10375 out[info.outOffset[j]] <<= 8;
10378 out += info.outJump;
10381 else if (info.inFormat == RTAUDIO_SINT16) {
10382 // Channel compensation and/or (de)interleaving only.
10383 Int16 *in = (Int16 *)inBuffer;
10384 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10385 for (j=0; j<info.channels; j++) {
10386 out[info.outOffset[j]] = in[info.inOffset[j]];
10389 out += info.outJump;
10392 else if (info.inFormat == RTAUDIO_SINT24) {
10393 Int24 *in = (Int24 *)inBuffer;
10394 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10395 for (j=0; j<info.channels; j++) {
10396 out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]].asInt() >> 8);
10399 out += info.outJump;
10402 else if (info.inFormat == RTAUDIO_SINT32) {
10403 Int32 *in = (Int32 *)inBuffer;
10404 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10405 for (j=0; j<info.channels; j++) {
10406 out[info.outOffset[j]] = (Int16) ((in[info.inOffset[j]] >> 16) & 0x0000ffff);
10409 out += info.outJump;
10412 else if (info.inFormat == RTAUDIO_FLOAT32) {
10413 Float32 *in = (Float32 *)inBuffer;
10414 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10415 for (j=0; j<info.channels; j++) {
10416 out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
10419 out += info.outJump;
10422 else if (info.inFormat == RTAUDIO_FLOAT64) {
10423 Float64 *in = (Float64 *)inBuffer;
10424 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10425 for (j=0; j<info.channels; j++) {
10426 out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
10429 out += info.outJump;
10433 else if (info.outFormat == RTAUDIO_SINT8) {
10434 signed char *out = (signed char *)outBuffer;
10435 if (info.inFormat == RTAUDIO_SINT8) {
10436 // Channel compensation and/or (de)interleaving only.
10437 signed char *in = (signed char *)inBuffer;
10438 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10439 for (j=0; j<info.channels; j++) {
10440 out[info.outOffset[j]] = in[info.inOffset[j]];
10443 out += info.outJump;
10446 if (info.inFormat == RTAUDIO_SINT16) {
10447 Int16 *in = (Int16 *)inBuffer;
10448 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10449 for (j=0; j<info.channels; j++) {
10450 out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 8) & 0x00ff);
10453 out += info.outJump;
10456 else if (info.inFormat == RTAUDIO_SINT24) {
10457 Int24 *in = (Int24 *)inBuffer;
10458 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10459 for (j=0; j<info.channels; j++) {
10460 out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]].asInt() >> 16);
10463 out += info.outJump;
10466 else if (info.inFormat == RTAUDIO_SINT32) {
10467 Int32 *in = (Int32 *)inBuffer;
10468 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10469 for (j=0; j<info.channels; j++) {
10470 out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 24) & 0x000000ff);
10473 out += info.outJump;
10476 else if (info.inFormat == RTAUDIO_FLOAT32) {
10477 Float32 *in = (Float32 *)inBuffer;
10478 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10479 for (j=0; j<info.channels; j++) {
10480 out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
10483 out += info.outJump;
10486 else if (info.inFormat == RTAUDIO_FLOAT64) {
10487 Float64 *in = (Float64 *)inBuffer;
10488 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10489 for (j=0; j<info.channels; j++) {
10490 out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
10493 out += info.outJump;
10499 //static inline uint16_t bswap_16(uint16_t x) { return (x>>8) | (x<<8); }
10500 //static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); }
10501 //static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); }
10503 void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format )
10509 if ( format == RTAUDIO_SINT16 ) {
10510 for ( unsigned int i=0; i<samples; i++ ) {
10511 // Swap 1st and 2nd bytes.
10516 // Increment 2 bytes.
10520 else if ( format == RTAUDIO_SINT32 ||
10521 format == RTAUDIO_FLOAT32 ) {
10522 for ( unsigned int i=0; i<samples; i++ ) {
10523 // Swap 1st and 4th bytes.
10528 // Swap 2nd and 3rd bytes.
10534 // Increment 3 more bytes.
10538 else if ( format == RTAUDIO_SINT24 ) {
10539 for ( unsigned int i=0; i<samples; i++ ) {
10540 // Swap 1st and 3rd bytes.
10545 // Increment 2 more bytes.
10549 else if ( format == RTAUDIO_FLOAT64 ) {
10550 for ( unsigned int i=0; i<samples; i++ ) {
10551 // Swap 1st and 8th bytes
10556 // Swap 2nd and 7th bytes
10562 // Swap 3rd and 6th bytes
10568 // Swap 4th and 5th bytes
10574 // Increment 5 more bytes.
10580 // Indentation settings for Vim and Emacs
10582 // Local Variables:
10583 // c-basic-offset: 2
10584 // indent-tabs-mode: nil
10587 // vim: et sts=2 sw=2