1 /************************************************************************/
3 \brief Realtime audio i/o C++ classes.
5 RtAudio provides a common API (Application Programming Interface)
6 for realtime audio input/output across Linux (native ALSA, Jack,
7 and OSS), Macintosh OS X (CoreAudio and Jack), and Windows
8 (DirectSound, ASIO and WASAPI) operating systems.
10 RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/
12 RtAudio: realtime audio i/o C++ classes
13 Copyright (c) 2001-2017 Gary P. Scavone
15 Permission is hereby granted, free of charge, to any person
16 obtaining a copy of this software and associated documentation files
17 (the "Software"), to deal in the Software without restriction,
18 including without limitation the rights to use, copy, modify, merge,
19 publish, distribute, sublicense, and/or sell copies of the Software,
20 and to permit persons to whom the Software is furnished to do so,
21 subject to the following conditions:
23 The above copyright notice and this permission notice shall be
24 included in all copies or substantial portions of the Software.
26 Any person wishing to distribute modifications to the Software is
27 asked to send the modifications to the original developer so that
28 they can be incorporated into the canonical version. This is,
29 however, not a binding provision of this license.
31 THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
32 EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
33 MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
34 IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
35 ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
36 CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
37 WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
39 /************************************************************************/
41 // RtAudio: Version 5.0.0
51 // Static variable definitions.
52 const unsigned int RtApi::MAX_SAMPLE_RATES = 14;
53 const unsigned int RtApi::SAMPLE_RATES[] = {
54 4000, 5512, 8000, 9600, 11025, 16000, 22050,
55 32000, 44100, 48000, 88200, 96000, 176400, 192000
58 #if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) || defined(__WINDOWS_WASAPI__)
59 #define MUTEX_INITIALIZE(A) InitializeCriticalSection(A)
60 #define MUTEX_DESTROY(A) DeleteCriticalSection(A)
61 #define MUTEX_LOCK(A) EnterCriticalSection(A)
62 #define MUTEX_UNLOCK(A) LeaveCriticalSection(A)
66 static std::string convertCharPointerToStdString(const char *text)
68 return std::string(text);
71 static std::string convertCharPointerToStdString(const wchar_t *text)
73 int length = WideCharToMultiByte(CP_UTF8, 0, text, -1, NULL, 0, NULL, NULL);
74 std::string s( length-1, '\0' );
75 WideCharToMultiByte(CP_UTF8, 0, text, -1, &s[0], length, NULL, NULL);
79 #elif defined(__LINUX_ALSA__) || defined(__LINUX_PULSE__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__)
81 #define MUTEX_INITIALIZE(A) pthread_mutex_init(A, NULL)
82 #define MUTEX_DESTROY(A) pthread_mutex_destroy(A)
83 #define MUTEX_LOCK(A) pthread_mutex_lock(A)
84 #define MUTEX_UNLOCK(A) pthread_mutex_unlock(A)
86 #define MUTEX_INITIALIZE(A) abs(*A) // dummy definitions
87 #define MUTEX_DESTROY(A) abs(*A) // dummy definitions
90 // *************************************************** //
92 // RtAudio definitions.
94 // *************************************************** //
96 std::string RtAudio :: getVersion( void )
98 return RTAUDIO_VERSION;
101 // Define API names and display names.
102 // Must be in same order as API enum.
104 const char* rtaudio_api_names[][2] = {
105 { "unspecified" , "Unknown" },
107 { "pulse" , "Pulse" },
108 { "oss" , "OpenSoundSystem" },
110 { "core" , "CoreAudio" },
111 { "wasapi" , "WASAPI" },
113 { "ds" , "DirectSound" },
114 { "dummy" , "Dummy" },
116 const unsigned int rtaudio_num_api_names =
117 sizeof(rtaudio_api_names)/sizeof(rtaudio_api_names[0]);
119 // The order here will control the order of RtAudio's API search in
121 extern "C" const RtAudio::Api rtaudio_compiled_apis[] = {
122 #if defined(__UNIX_JACK__)
125 #if defined(__LINUX_PULSE__)
126 RtAudio::LINUX_PULSE,
128 #if defined(__LINUX_ALSA__)
131 #if defined(__LINUX_OSS__)
134 #if defined(__WINDOWS_ASIO__)
135 RtAudio::WINDOWS_ASIO,
137 #if defined(__WINDOWS_WASAPI__)
138 RtAudio::WINDOWS_WASAPI,
140 #if defined(__WINDOWS_DS__)
143 #if defined(__MACOSX_CORE__)
144 RtAudio::MACOSX_CORE,
146 #if defined(__RTAUDIO_DUMMY__)
147 RtAudio::RTAUDIO_DUMMY,
149 RtAudio::UNSPECIFIED,
151 extern "C" const unsigned int rtaudio_num_compiled_apis =
152 sizeof(rtaudio_compiled_apis)/sizeof(rtaudio_compiled_apis[0])-1;
155 // This is a compile-time check that rtaudio_num_api_names == RtAudio::NUM_APIS.
156 // If the build breaks here, check that they match.
157 template<bool b> class StaticAssert { private: StaticAssert() {} };
158 template<> class StaticAssert<true>{ public: StaticAssert() {} };
159 class StaticAssertions { StaticAssertions() {
160 StaticAssert<rtaudio_num_api_names == RtAudio::NUM_APIS>();
163 void RtAudio :: getCompiledApi( std::vector<RtAudio::Api> &apis )
165 apis = std::vector<RtAudio::Api>(rtaudio_compiled_apis,
166 rtaudio_compiled_apis + rtaudio_num_compiled_apis);
169 std::string RtAudio :: getApiName( RtAudio::Api api )
171 if (api < 0 || api >= RtAudio::NUM_APIS)
173 return rtaudio_api_names[api][0];
176 std::string RtAudio :: getApiDisplayName( RtAudio::Api api )
178 if (api < 0 || api >= RtAudio::NUM_APIS)
180 return rtaudio_api_names[api][1];
183 RtAudio::Api RtAudio :: getCompiledApiByName( const std::string &name )
186 for (i = 0; i < rtaudio_num_compiled_apis; ++i)
187 if (name == rtaudio_api_names[rtaudio_compiled_apis[i]][0])
188 return rtaudio_compiled_apis[i];
189 return RtAudio::UNSPECIFIED;
192 void RtAudio :: openRtApi( RtAudio::Api api )
198 #if defined(__UNIX_JACK__)
199 if ( api == UNIX_JACK )
200 rtapi_ = new RtApiJack();
202 #if defined(__LINUX_ALSA__)
203 if ( api == LINUX_ALSA )
204 rtapi_ = new RtApiAlsa();
206 #if defined(__LINUX_PULSE__)
207 if ( api == LINUX_PULSE )
208 rtapi_ = new RtApiPulse();
210 #if defined(__LINUX_OSS__)
211 if ( api == LINUX_OSS )
212 rtapi_ = new RtApiOss();
214 #if defined(__WINDOWS_ASIO__)
215 if ( api == WINDOWS_ASIO )
216 rtapi_ = new RtApiAsio();
218 #if defined(__WINDOWS_WASAPI__)
219 if ( api == WINDOWS_WASAPI )
220 rtapi_ = new RtApiWasapi();
222 #if defined(__WINDOWS_DS__)
223 if ( api == WINDOWS_DS )
224 rtapi_ = new RtApiDs();
226 #if defined(__MACOSX_CORE__)
227 if ( api == MACOSX_CORE )
228 rtapi_ = new RtApiCore();
230 #if defined(__RTAUDIO_DUMMY__)
231 if ( api == RTAUDIO_DUMMY )
232 rtapi_ = new RtApiDummy();
236 RtAudio :: RtAudio( RtAudio::Api api )
240 if ( api != UNSPECIFIED ) {
241 // Attempt to open the specified API.
243 if ( rtapi_ ) return;
245 // No compiled support for specified API value. Issue a debug
246 // warning and continue as if no API was specified.
247 std::cerr << "\nRtAudio: no compiled support for specified API argument!\n" << std::endl;
250 // Iterate through the compiled APIs and return as soon as we find
251 // one with at least one device or we reach the end of the list.
252 std::vector< RtAudio::Api > apis;
253 getCompiledApi( apis );
254 for ( unsigned int i=0; i<apis.size(); i++ ) {
255 openRtApi( apis[i] );
256 if ( rtapi_ && rtapi_->getDeviceCount() ) break;
259 if ( rtapi_ ) return;
261 // It should not be possible to get here because the preprocessor
262 // definition __RTAUDIO_DUMMY__ is automatically defined if no
263 // API-specific definitions are passed to the compiler. But just in
264 // case something weird happens, we'll thow an error.
265 std::string errorText = "\nRtAudio: no compiled API support found ... critical error!!\n\n";
266 throw( RtAudioError( errorText, RtAudioError::UNSPECIFIED ) );
269 RtAudio :: ~RtAudio()
275 void RtAudio :: openStream( RtAudio::StreamParameters *outputParameters,
276 RtAudio::StreamParameters *inputParameters,
277 RtAudioFormat format, unsigned int sampleRate,
278 unsigned int *bufferFrames,
279 RtAudioCallback callback, void *userData,
280 RtAudio::StreamOptions *options,
281 RtAudioErrorCallback errorCallback )
283 return rtapi_->openStream( outputParameters, inputParameters, format,
284 sampleRate, bufferFrames, callback,
285 userData, options, errorCallback );
288 // *************************************************** //
290 // Public RtApi definitions (see end of file for
291 // private or protected utility functions).
293 // *************************************************** //
297 stream_.state = STREAM_CLOSED;
298 stream_.mode = UNINITIALIZED;
299 stream_.apiHandle = 0;
300 stream_.userBuffer[0] = 0;
301 stream_.userBuffer[1] = 0;
302 MUTEX_INITIALIZE( &stream_.mutex );
303 showWarnings_ = true;
304 firstErrorOccurred_ = false;
309 MUTEX_DESTROY( &stream_.mutex );
312 void RtApi :: openStream( RtAudio::StreamParameters *oParams,
313 RtAudio::StreamParameters *iParams,
314 RtAudioFormat format, unsigned int sampleRate,
315 unsigned int *bufferFrames,
316 RtAudioCallback callback, void *userData,
317 RtAudio::StreamOptions *options,
318 RtAudioErrorCallback errorCallback )
320 if ( stream_.state != STREAM_CLOSED ) {
321 errorText_ = "RtApi::openStream: a stream is already open!";
322 error( RtAudioError::INVALID_USE );
326 // Clear stream information potentially left from a previously open stream.
329 if ( oParams && oParams->nChannels < 1 ) {
330 errorText_ = "RtApi::openStream: a non-NULL output StreamParameters structure cannot have an nChannels value less than one.";
331 error( RtAudioError::INVALID_USE );
335 if ( iParams && iParams->nChannels < 1 ) {
336 errorText_ = "RtApi::openStream: a non-NULL input StreamParameters structure cannot have an nChannels value less than one.";
337 error( RtAudioError::INVALID_USE );
341 if ( oParams == NULL && iParams == NULL ) {
342 errorText_ = "RtApi::openStream: input and output StreamParameters structures are both NULL!";
343 error( RtAudioError::INVALID_USE );
347 if ( formatBytes(format) == 0 ) {
348 errorText_ = "RtApi::openStream: 'format' parameter value is undefined.";
349 error( RtAudioError::INVALID_USE );
353 unsigned int nDevices = getDeviceCount();
354 unsigned int oChannels = 0;
356 oChannels = oParams->nChannels;
357 if ( oParams->deviceId >= nDevices ) {
358 errorText_ = "RtApi::openStream: output device parameter value is invalid.";
359 error( RtAudioError::INVALID_USE );
364 unsigned int iChannels = 0;
366 iChannels = iParams->nChannels;
367 if ( iParams->deviceId >= nDevices ) {
368 errorText_ = "RtApi::openStream: input device parameter value is invalid.";
369 error( RtAudioError::INVALID_USE );
376 if ( oChannels > 0 ) {
378 result = probeDeviceOpen( oParams->deviceId, OUTPUT, oChannels, oParams->firstChannel,
379 sampleRate, format, bufferFrames, options );
380 if ( result == false ) {
381 error( RtAudioError::SYSTEM_ERROR );
386 if ( iChannels > 0 ) {
388 result = probeDeviceOpen( iParams->deviceId, INPUT, iChannels, iParams->firstChannel,
389 sampleRate, format, bufferFrames, options );
390 if ( result == false ) {
391 if ( oChannels > 0 ) closeStream();
392 error( RtAudioError::SYSTEM_ERROR );
397 stream_.callbackInfo.callback = (void *) callback;
398 stream_.callbackInfo.userData = userData;
399 stream_.callbackInfo.errorCallback = (void *) errorCallback;
401 if ( options ) options->numberOfBuffers = stream_.nBuffers;
402 stream_.state = STREAM_STOPPED;
405 unsigned int RtApi :: getDefaultInputDevice( void )
407 // Should be implemented in subclasses if possible.
411 unsigned int RtApi :: getDefaultOutputDevice( void )
413 // Should be implemented in subclasses if possible.
417 void RtApi :: closeStream( void )
419 // MUST be implemented in subclasses!
423 bool RtApi :: probeDeviceOpen( unsigned int /*device*/, StreamMode /*mode*/, unsigned int /*channels*/,
424 unsigned int /*firstChannel*/, unsigned int /*sampleRate*/,
425 RtAudioFormat /*format*/, unsigned int * /*bufferSize*/,
426 RtAudio::StreamOptions * /*options*/ )
428 // MUST be implemented in subclasses!
432 void RtApi :: tickStreamTime( void )
434 // Subclasses that do not provide their own implementation of
435 // getStreamTime should call this function once per buffer I/O to
436 // provide basic stream time support.
438 stream_.streamTime += ( stream_.bufferSize * 1.0 / stream_.sampleRate );
440 #if defined( HAVE_GETTIMEOFDAY )
441 gettimeofday( &stream_.lastTickTimestamp, NULL );
445 long RtApi :: getStreamLatency( void )
449 long totalLatency = 0;
450 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
451 totalLatency = stream_.latency[0];
452 if ( stream_.mode == INPUT || stream_.mode == DUPLEX )
453 totalLatency += stream_.latency[1];
458 double RtApi :: getStreamTime( void )
462 #if defined( HAVE_GETTIMEOFDAY )
463 // Return a very accurate estimate of the stream time by
464 // adding in the elapsed time since the last tick.
468 if ( stream_.state != STREAM_RUNNING || stream_.streamTime == 0.0 )
469 return stream_.streamTime;
471 gettimeofday( &now, NULL );
472 then = stream_.lastTickTimestamp;
473 return stream_.streamTime +
474 ((now.tv_sec + 0.000001 * now.tv_usec) -
475 (then.tv_sec + 0.000001 * then.tv_usec));
477 return stream_.streamTime;
481 void RtApi :: setStreamTime( double time )
486 stream_.streamTime = time;
487 #if defined( HAVE_GETTIMEOFDAY )
488 gettimeofday( &stream_.lastTickTimestamp, NULL );
492 unsigned int RtApi :: getStreamSampleRate( void )
496 return stream_.sampleRate;
500 // *************************************************** //
502 // OS/API-specific methods.
504 // *************************************************** //
506 #if defined(__MACOSX_CORE__)
508 // The OS X CoreAudio API is designed to use a separate callback
509 // procedure for each of its audio devices. A single RtAudio duplex
510 // stream using two different devices is supported here, though it
511 // cannot be guaranteed to always behave correctly because we cannot
512 // synchronize these two callbacks.
514 // A property listener is installed for over/underrun information.
515 // However, no functionality is currently provided to allow property
516 // listeners to trigger user handlers because it is unclear what could
517 // be done if a critical stream parameter (buffer size, sample rate,
518 // device disconnect) notification arrived. The listeners entail
519 // quite a bit of extra code and most likely, a user program wouldn't
520 // be prepared for the result anyway. However, we do provide a flag
521 // to the client callback function to inform of an over/underrun.
523 // A structure to hold various information related to the CoreAudio API
526 AudioDeviceID id[2]; // device ids
527 #if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
528 AudioDeviceIOProcID procId[2];
530 UInt32 iStream[2]; // device stream index (or first if using multiple)
531 UInt32 nStreams[2]; // number of streams to use
534 pthread_cond_t condition;
535 int drainCounter; // Tracks callback counts when draining
536 bool internalDrain; // Indicates if stop is initiated from callback or not.
539 :deviceBuffer(0), drainCounter(0), internalDrain(false) { nStreams[0] = 1; nStreams[1] = 1; id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
542 RtApiCore:: RtApiCore()
544 #if defined( AVAILABLE_MAC_OS_X_VERSION_10_6_AND_LATER )
545 // This is a largely undocumented but absolutely necessary
546 // requirement starting with OS-X 10.6. If not called, queries and
547 // updates to various audio device properties are not handled
549 CFRunLoopRef theRunLoop = NULL;
550 AudioObjectPropertyAddress property = { kAudioHardwarePropertyRunLoop,
551 kAudioObjectPropertyScopeGlobal,
552 kAudioObjectPropertyElementMaster };
553 OSStatus result = AudioObjectSetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, sizeof(CFRunLoopRef), &theRunLoop);
554 if ( result != noErr ) {
555 errorText_ = "RtApiCore::RtApiCore: error setting run loop property!";
556 error( RtAudioError::WARNING );
561 RtApiCore :: ~RtApiCore()
563 // The subclass destructor gets called before the base class
564 // destructor, so close an existing stream before deallocating
565 // apiDeviceId memory.
566 if ( stream_.state != STREAM_CLOSED ) closeStream();
569 unsigned int RtApiCore :: getDeviceCount( void )
571 // Find out how many audio devices there are, if any.
573 AudioObjectPropertyAddress propertyAddress = { kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
574 OSStatus result = AudioObjectGetPropertyDataSize( kAudioObjectSystemObject, &propertyAddress, 0, NULL, &dataSize );
575 if ( result != noErr ) {
576 errorText_ = "RtApiCore::getDeviceCount: OS-X error getting device info!";
577 error( RtAudioError::WARNING );
581 return dataSize / sizeof( AudioDeviceID );
584 unsigned int RtApiCore :: getDefaultInputDevice( void )
586 unsigned int nDevices = getDeviceCount();
587 if ( nDevices <= 1 ) return 0;
590 UInt32 dataSize = sizeof( AudioDeviceID );
591 AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultInputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
592 OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id );
593 if ( result != noErr ) {
594 errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device.";
595 error( RtAudioError::WARNING );
599 dataSize *= nDevices;
600 AudioDeviceID deviceList[ nDevices ];
601 property.mSelector = kAudioHardwarePropertyDevices;
602 result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList );
603 if ( result != noErr ) {
604 errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device IDs.";
605 error( RtAudioError::WARNING );
609 for ( unsigned int i=0; i<nDevices; i++ )
610 if ( id == deviceList[i] ) return i;
612 errorText_ = "RtApiCore::getDefaultInputDevice: No default device found!";
613 error( RtAudioError::WARNING );
617 unsigned int RtApiCore :: getDefaultOutputDevice( void )
619 unsigned int nDevices = getDeviceCount();
620 if ( nDevices <= 1 ) return 0;
623 UInt32 dataSize = sizeof( AudioDeviceID );
624 AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultOutputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
625 OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id );
626 if ( result != noErr ) {
627 errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device.";
628 error( RtAudioError::WARNING );
632 dataSize = sizeof( AudioDeviceID ) * nDevices;
633 AudioDeviceID deviceList[ nDevices ];
634 property.mSelector = kAudioHardwarePropertyDevices;
635 result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList );
636 if ( result != noErr ) {
637 errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device IDs.";
638 error( RtAudioError::WARNING );
642 for ( unsigned int i=0; i<nDevices; i++ )
643 if ( id == deviceList[i] ) return i;
645 errorText_ = "RtApiCore::getDefaultOutputDevice: No default device found!";
646 error( RtAudioError::WARNING );
650 RtAudio::DeviceInfo RtApiCore :: getDeviceInfo( unsigned int device )
652 RtAudio::DeviceInfo info;
656 unsigned int nDevices = getDeviceCount();
657 if ( nDevices == 0 ) {
658 errorText_ = "RtApiCore::getDeviceInfo: no devices found!";
659 error( RtAudioError::INVALID_USE );
663 if ( device >= nDevices ) {
664 errorText_ = "RtApiCore::getDeviceInfo: device ID is invalid!";
665 error( RtAudioError::INVALID_USE );
669 AudioDeviceID deviceList[ nDevices ];
670 UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;
671 AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
672 kAudioObjectPropertyScopeGlobal,
673 kAudioObjectPropertyElementMaster };
674 OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property,
675 0, NULL, &dataSize, (void *) &deviceList );
676 if ( result != noErr ) {
677 errorText_ = "RtApiCore::getDeviceInfo: OS-X system error getting device IDs.";
678 error( RtAudioError::WARNING );
682 AudioDeviceID id = deviceList[ device ];
684 // Get the device name.
687 dataSize = sizeof( CFStringRef );
688 property.mSelector = kAudioObjectPropertyManufacturer;
689 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname );
690 if ( result != noErr ) {
691 errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device manufacturer.";
692 errorText_ = errorStream_.str();
693 error( RtAudioError::WARNING );
697 //const char *mname = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() );
698 int length = CFStringGetLength(cfname);
699 char *mname = (char *)malloc(length * 3 + 1);
700 #if defined( UNICODE ) || defined( _UNICODE )
701 CFStringGetCString(cfname, mname, length * 3 + 1, kCFStringEncodingUTF8);
703 CFStringGetCString(cfname, mname, length * 3 + 1, CFStringGetSystemEncoding());
705 info.name.append( (const char *)mname, strlen(mname) );
706 info.name.append( ": " );
710 property.mSelector = kAudioObjectPropertyName;
711 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname );
712 if ( result != noErr ) {
713 errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device name.";
714 errorText_ = errorStream_.str();
715 error( RtAudioError::WARNING );
719 //const char *name = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() );
720 length = CFStringGetLength(cfname);
721 char *name = (char *)malloc(length * 3 + 1);
722 #if defined( UNICODE ) || defined( _UNICODE )
723 CFStringGetCString(cfname, name, length * 3 + 1, kCFStringEncodingUTF8);
725 CFStringGetCString(cfname, name, length * 3 + 1, CFStringGetSystemEncoding());
727 info.name.append( (const char *)name, strlen(name) );
731 // Get the output stream "configuration".
732 AudioBufferList *bufferList = nil;
733 property.mSelector = kAudioDevicePropertyStreamConfiguration;
734 property.mScope = kAudioDevicePropertyScopeOutput;
735 // property.mElement = kAudioObjectPropertyElementWildcard;
737 result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
738 if ( result != noErr || dataSize == 0 ) {
739 errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration info for device (" << device << ").";
740 errorText_ = errorStream_.str();
741 error( RtAudioError::WARNING );
745 // Allocate the AudioBufferList.
746 bufferList = (AudioBufferList *) malloc( dataSize );
747 if ( bufferList == NULL ) {
748 errorText_ = "RtApiCore::getDeviceInfo: memory error allocating output AudioBufferList.";
749 error( RtAudioError::WARNING );
753 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
754 if ( result != noErr || dataSize == 0 ) {
756 errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration for device (" << device << ").";
757 errorText_ = errorStream_.str();
758 error( RtAudioError::WARNING );
762 // Get output channel information.
763 unsigned int i, nStreams = bufferList->mNumberBuffers;
764 for ( i=0; i<nStreams; i++ )
765 info.outputChannels += bufferList->mBuffers[i].mNumberChannels;
768 // Get the input stream "configuration".
769 property.mScope = kAudioDevicePropertyScopeInput;
770 result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
771 if ( result != noErr || dataSize == 0 ) {
772 errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration info for device (" << device << ").";
773 errorText_ = errorStream_.str();
774 error( RtAudioError::WARNING );
778 // Allocate the AudioBufferList.
779 bufferList = (AudioBufferList *) malloc( dataSize );
780 if ( bufferList == NULL ) {
781 errorText_ = "RtApiCore::getDeviceInfo: memory error allocating input AudioBufferList.";
782 error( RtAudioError::WARNING );
786 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
787 if (result != noErr || dataSize == 0) {
789 errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration for device (" << device << ").";
790 errorText_ = errorStream_.str();
791 error( RtAudioError::WARNING );
795 // Get input channel information.
796 nStreams = bufferList->mNumberBuffers;
797 for ( i=0; i<nStreams; i++ )
798 info.inputChannels += bufferList->mBuffers[i].mNumberChannels;
801 // If device opens for both playback and capture, we determine the channels.
802 if ( info.outputChannels > 0 && info.inputChannels > 0 )
803 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
805 // Probe the device sample rates.
806 bool isInput = false;
807 if ( info.outputChannels == 0 ) isInput = true;
809 // Determine the supported sample rates.
810 property.mSelector = kAudioDevicePropertyAvailableNominalSampleRates;
811 if ( isInput == false ) property.mScope = kAudioDevicePropertyScopeOutput;
812 result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
813 if ( result != kAudioHardwareNoError || dataSize == 0 ) {
814 errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rate info.";
815 errorText_ = errorStream_.str();
816 error( RtAudioError::WARNING );
820 UInt32 nRanges = dataSize / sizeof( AudioValueRange );
821 AudioValueRange rangeList[ nRanges ];
822 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &rangeList );
823 if ( result != kAudioHardwareNoError ) {
824 errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rates.";
825 errorText_ = errorStream_.str();
826 error( RtAudioError::WARNING );
830 // The sample rate reporting mechanism is a bit of a mystery. It
831 // seems that it can either return individual rates or a range of
832 // rates. I assume that if the min / max range values are the same,
833 // then that represents a single supported rate and if the min / max
834 // range values are different, the device supports an arbitrary
835 // range of values (though there might be multiple ranges, so we'll
836 // use the most conservative range).
837 Float64 minimumRate = 1.0, maximumRate = 10000000000.0;
838 bool haveValueRange = false;
839 info.sampleRates.clear();
840 for ( UInt32 i=0; i<nRanges; i++ ) {
841 if ( rangeList[i].mMinimum == rangeList[i].mMaximum ) {
842 unsigned int tmpSr = (unsigned int) rangeList[i].mMinimum;
843 info.sampleRates.push_back( tmpSr );
845 if ( !info.preferredSampleRate || ( tmpSr <= 48000 && tmpSr > info.preferredSampleRate ) )
846 info.preferredSampleRate = tmpSr;
849 haveValueRange = true;
850 if ( rangeList[i].mMinimum > minimumRate ) minimumRate = rangeList[i].mMinimum;
851 if ( rangeList[i].mMaximum < maximumRate ) maximumRate = rangeList[i].mMaximum;
855 if ( haveValueRange ) {
856 for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
857 if ( SAMPLE_RATES[k] >= (unsigned int) minimumRate && SAMPLE_RATES[k] <= (unsigned int) maximumRate ) {
858 info.sampleRates.push_back( SAMPLE_RATES[k] );
860 if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
861 info.preferredSampleRate = SAMPLE_RATES[k];
866 // Sort and remove any redundant values
867 std::sort( info.sampleRates.begin(), info.sampleRates.end() );
868 info.sampleRates.erase( unique( info.sampleRates.begin(), info.sampleRates.end() ), info.sampleRates.end() );
870 if ( info.sampleRates.size() == 0 ) {
871 errorStream_ << "RtApiCore::probeDeviceInfo: No supported sample rates found for device (" << device << ").";
872 errorText_ = errorStream_.str();
873 error( RtAudioError::WARNING );
877 // CoreAudio always uses 32-bit floating point data for PCM streams.
878 // Thus, any other "physical" formats supported by the device are of
879 // no interest to the client.
880 info.nativeFormats = RTAUDIO_FLOAT32;
882 if ( info.outputChannels > 0 )
883 if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true;
884 if ( info.inputChannels > 0 )
885 if ( getDefaultInputDevice() == device ) info.isDefaultInput = true;
891 static OSStatus callbackHandler( AudioDeviceID inDevice,
892 const AudioTimeStamp* /*inNow*/,
893 const AudioBufferList* inInputData,
894 const AudioTimeStamp* /*inInputTime*/,
895 AudioBufferList* outOutputData,
896 const AudioTimeStamp* /*inOutputTime*/,
899 CallbackInfo *info = (CallbackInfo *) infoPointer;
901 RtApiCore *object = (RtApiCore *) info->object;
902 if ( object->callbackEvent( inDevice, inInputData, outOutputData ) == false )
903 return kAudioHardwareUnspecifiedError;
905 return kAudioHardwareNoError;
908 static OSStatus xrunListener( AudioObjectID /*inDevice*/,
910 const AudioObjectPropertyAddress properties[],
911 void* handlePointer )
913 CoreHandle *handle = (CoreHandle *) handlePointer;
914 for ( UInt32 i=0; i<nAddresses; i++ ) {
915 if ( properties[i].mSelector == kAudioDeviceProcessorOverload ) {
916 if ( properties[i].mScope == kAudioDevicePropertyScopeInput )
917 handle->xrun[1] = true;
919 handle->xrun[0] = true;
923 return kAudioHardwareNoError;
926 static OSStatus rateListener( AudioObjectID inDevice,
927 UInt32 /*nAddresses*/,
928 const AudioObjectPropertyAddress /*properties*/[],
931 Float64 *rate = (Float64 *) ratePointer;
932 UInt32 dataSize = sizeof( Float64 );
933 AudioObjectPropertyAddress property = { kAudioDevicePropertyNominalSampleRate,
934 kAudioObjectPropertyScopeGlobal,
935 kAudioObjectPropertyElementMaster };
936 AudioObjectGetPropertyData( inDevice, &property, 0, NULL, &dataSize, rate );
937 return kAudioHardwareNoError;
940 bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
941 unsigned int firstChannel, unsigned int sampleRate,
942 RtAudioFormat format, unsigned int *bufferSize,
943 RtAudio::StreamOptions *options )
946 unsigned int nDevices = getDeviceCount();
947 if ( nDevices == 0 ) {
948 // This should not happen because a check is made before this function is called.
949 errorText_ = "RtApiCore::probeDeviceOpen: no devices found!";
953 if ( device >= nDevices ) {
954 // This should not happen because a check is made before this function is called.
955 errorText_ = "RtApiCore::probeDeviceOpen: device ID is invalid!";
959 AudioDeviceID deviceList[ nDevices ];
960 UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;
961 AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
962 kAudioObjectPropertyScopeGlobal,
963 kAudioObjectPropertyElementMaster };
964 OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property,
965 0, NULL, &dataSize, (void *) &deviceList );
966 if ( result != noErr ) {
967 errorText_ = "RtApiCore::probeDeviceOpen: OS-X system error getting device IDs.";
971 AudioDeviceID id = deviceList[ device ];
973 // Setup for stream mode.
974 bool isInput = false;
975 if ( mode == INPUT ) {
977 property.mScope = kAudioDevicePropertyScopeInput;
980 property.mScope = kAudioDevicePropertyScopeOutput;
982 // Get the stream "configuration".
983 AudioBufferList *bufferList = nil;
985 property.mSelector = kAudioDevicePropertyStreamConfiguration;
986 result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
987 if ( result != noErr || dataSize == 0 ) {
988 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration info for device (" << device << ").";
989 errorText_ = errorStream_.str();
993 // Allocate the AudioBufferList.
994 bufferList = (AudioBufferList *) malloc( dataSize );
995 if ( bufferList == NULL ) {
996 errorText_ = "RtApiCore::probeDeviceOpen: memory error allocating AudioBufferList.";
1000 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
1001 if (result != noErr || dataSize == 0) {
1003 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration for device (" << device << ").";
1004 errorText_ = errorStream_.str();
1008 // Search for one or more streams that contain the desired number of
1009 // channels. CoreAudio devices can have an arbitrary number of
1010 // streams and each stream can have an arbitrary number of channels.
1011 // For each stream, a single buffer of interleaved samples is
1012 // provided. RtAudio prefers the use of one stream of interleaved
1013 // data or multiple consecutive single-channel streams. However, we
1014 // now support multiple consecutive multi-channel streams of
1015 // interleaved data as well.
1016 UInt32 iStream, offsetCounter = firstChannel;
1017 UInt32 nStreams = bufferList->mNumberBuffers;
1018 bool monoMode = false;
1019 bool foundStream = false;
1021 // First check that the device supports the requested number of
1023 UInt32 deviceChannels = 0;
1024 for ( iStream=0; iStream<nStreams; iStream++ )
1025 deviceChannels += bufferList->mBuffers[iStream].mNumberChannels;
1027 if ( deviceChannels < ( channels + firstChannel ) ) {
1029 errorStream_ << "RtApiCore::probeDeviceOpen: the device (" << device << ") does not support the requested channel count.";
1030 errorText_ = errorStream_.str();
1034 // Look for a single stream meeting our needs.
1035 UInt32 firstStream, streamCount = 1, streamChannels = 0, channelOffset = 0;
1036 for ( iStream=0; iStream<nStreams; iStream++ ) {
1037 streamChannels = bufferList->mBuffers[iStream].mNumberChannels;
1038 if ( streamChannels >= channels + offsetCounter ) {
1039 firstStream = iStream;
1040 channelOffset = offsetCounter;
1044 if ( streamChannels > offsetCounter ) break;
1045 offsetCounter -= streamChannels;
1048 // If we didn't find a single stream above, then we should be able
1049 // to meet the channel specification with multiple streams.
1050 if ( foundStream == false ) {
1052 offsetCounter = firstChannel;
1053 for ( iStream=0; iStream<nStreams; iStream++ ) {
1054 streamChannels = bufferList->mBuffers[iStream].mNumberChannels;
1055 if ( streamChannels > offsetCounter ) break;
1056 offsetCounter -= streamChannels;
1059 firstStream = iStream;
1060 channelOffset = offsetCounter;
1061 Int32 channelCounter = channels + offsetCounter - streamChannels;
1063 if ( streamChannels > 1 ) monoMode = false;
1064 while ( channelCounter > 0 ) {
1065 streamChannels = bufferList->mBuffers[++iStream].mNumberChannels;
1066 if ( streamChannels > 1 ) monoMode = false;
1067 channelCounter -= streamChannels;
1074 // Determine the buffer size.
1075 AudioValueRange bufferRange;
1076 dataSize = sizeof( AudioValueRange );
1077 property.mSelector = kAudioDevicePropertyBufferFrameSizeRange;
1078 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &bufferRange );
1080 if ( result != noErr ) {
1081 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting buffer size range for device (" << device << ").";
1082 errorText_ = errorStream_.str();
1086 if ( bufferRange.mMinimum > *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMinimum;
1087 else if ( bufferRange.mMaximum < *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMaximum;
1088 if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) *bufferSize = (unsigned long) bufferRange.mMinimum;
1090 // Set the buffer size. For multiple streams, I'm assuming we only
1091 // need to make this setting for the master channel.
1092 UInt32 theSize = (UInt32) *bufferSize;
1093 dataSize = sizeof( UInt32 );
1094 property.mSelector = kAudioDevicePropertyBufferFrameSize;
1095 result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &theSize );
1097 if ( result != noErr ) {
1098 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting the buffer size for device (" << device << ").";
1099 errorText_ = errorStream_.str();
1103 // If attempting to setup a duplex stream, the bufferSize parameter
1104 // MUST be the same in both directions!
1105 *bufferSize = theSize;
1106 if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {
1107 errorStream_ << "RtApiCore::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << device << ").";
1108 errorText_ = errorStream_.str();
1112 stream_.bufferSize = *bufferSize;
1113 stream_.nBuffers = 1;
1115 // Try to set "hog" mode ... it's not clear to me this is working.
1116 if ( options && options->flags & RTAUDIO_HOG_DEVICE ) {
1118 dataSize = sizeof( hog_pid );
1119 property.mSelector = kAudioDevicePropertyHogMode;
1120 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &hog_pid );
1121 if ( result != noErr ) {
1122 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting 'hog' state!";
1123 errorText_ = errorStream_.str();
1127 if ( hog_pid != getpid() ) {
1129 result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &hog_pid );
1130 if ( result != noErr ) {
1131 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting 'hog' state!";
1132 errorText_ = errorStream_.str();
1138 // Check and if necessary, change the sample rate for the device.
1139 Float64 nominalRate;
1140 dataSize = sizeof( Float64 );
1141 property.mSelector = kAudioDevicePropertyNominalSampleRate;
1142 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &nominalRate );
1143 if ( result != noErr ) {
1144 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting current sample rate.";
1145 errorText_ = errorStream_.str();
1149 // Only change the sample rate if off by more than 1 Hz.
1150 if ( fabs( nominalRate - (double)sampleRate ) > 1.0 ) {
1152 // Set a property listener for the sample rate change
1153 Float64 reportedRate = 0.0;
1154 AudioObjectPropertyAddress tmp = { kAudioDevicePropertyNominalSampleRate, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
1155 result = AudioObjectAddPropertyListener( id, &tmp, rateListener, (void *) &reportedRate );
1156 if ( result != noErr ) {
1157 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate property listener for device (" << device << ").";
1158 errorText_ = errorStream_.str();
1162 nominalRate = (Float64) sampleRate;
1163 result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &nominalRate );
1164 if ( result != noErr ) {
1165 AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate );
1166 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate for device (" << device << ").";
1167 errorText_ = errorStream_.str();
1171 // Now wait until the reported nominal rate is what we just set.
1172 UInt32 microCounter = 0;
1173 while ( reportedRate != nominalRate ) {
1174 microCounter += 5000;
1175 if ( microCounter > 5000000 ) break;
1179 // Remove the property listener.
1180 AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate );
1182 if ( microCounter > 5000000 ) {
1183 errorStream_ << "RtApiCore::probeDeviceOpen: timeout waiting for sample rate update for device (" << device << ").";
1184 errorText_ = errorStream_.str();
1189 // Now set the stream format for all streams. Also, check the
1190 // physical format of the device and change that if necessary.
1191 AudioStreamBasicDescription description;
1192 dataSize = sizeof( AudioStreamBasicDescription );
1193 property.mSelector = kAudioStreamPropertyVirtualFormat;
1194 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &description );
1195 if ( result != noErr ) {
1196 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream format for device (" << device << ").";
1197 errorText_ = errorStream_.str();
1201 // Set the sample rate and data format id. However, only make the
1202 // change if the sample rate is not within 1.0 of the desired
1203 // rate and the format is not linear pcm.
1204 bool updateFormat = false;
1205 if ( fabs( description.mSampleRate - (Float64)sampleRate ) > 1.0 ) {
1206 description.mSampleRate = (Float64) sampleRate;
1207 updateFormat = true;
1210 if ( description.mFormatID != kAudioFormatLinearPCM ) {
1211 description.mFormatID = kAudioFormatLinearPCM;
1212 updateFormat = true;
1215 if ( updateFormat ) {
1216 result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &description );
1217 if ( result != noErr ) {
1218 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate or data format for device (" << device << ").";
1219 errorText_ = errorStream_.str();
1224 // Now check the physical format.
1225 property.mSelector = kAudioStreamPropertyPhysicalFormat;
1226 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &description );
1227 if ( result != noErr ) {
1228 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream physical format for device (" << device << ").";
1229 errorText_ = errorStream_.str();
1233 //std::cout << "Current physical stream format:" << std::endl;
1234 //std::cout << " mBitsPerChan = " << description.mBitsPerChannel << std::endl;
1235 //std::cout << " aligned high = " << (description.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (description.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl;
1236 //std::cout << " bytesPerFrame = " << description.mBytesPerFrame << std::endl;
1237 //std::cout << " sample rate = " << description.mSampleRate << std::endl;
1239 if ( description.mFormatID != kAudioFormatLinearPCM || description.mBitsPerChannel < 16 ) {
1240 description.mFormatID = kAudioFormatLinearPCM;
1241 //description.mSampleRate = (Float64) sampleRate;
1242 AudioStreamBasicDescription testDescription = description;
1245 // We'll try higher bit rates first and then work our way down.
1246 std::vector< std::pair<UInt32, UInt32> > physicalFormats;
1247 formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsFloat) & ~kLinearPCMFormatFlagIsSignedInteger;
1248 physicalFormats.push_back( std::pair<Float32, UInt32>( 32, formatFlags ) );
1249 formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat;
1250 physicalFormats.push_back( std::pair<Float32, UInt32>( 32, formatFlags ) );
1251 physicalFormats.push_back( std::pair<Float32, UInt32>( 24, formatFlags ) ); // 24-bit packed
1252 formatFlags &= ~( kAudioFormatFlagIsPacked | kAudioFormatFlagIsAlignedHigh );
1253 physicalFormats.push_back( std::pair<Float32, UInt32>( 24.2, formatFlags ) ); // 24-bit in 4 bytes, aligned low
1254 formatFlags |= kAudioFormatFlagIsAlignedHigh;
1255 physicalFormats.push_back( std::pair<Float32, UInt32>( 24.4, formatFlags ) ); // 24-bit in 4 bytes, aligned high
1256 formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat;
1257 physicalFormats.push_back( std::pair<Float32, UInt32>( 16, formatFlags ) );
1258 physicalFormats.push_back( std::pair<Float32, UInt32>( 8, formatFlags ) );
1260 bool setPhysicalFormat = false;
1261 for( unsigned int i=0; i<physicalFormats.size(); i++ ) {
1262 testDescription = description;
1263 testDescription.mBitsPerChannel = (UInt32) physicalFormats[i].first;
1264 testDescription.mFormatFlags = physicalFormats[i].second;
1265 if ( (24 == (UInt32)physicalFormats[i].first) && ~( physicalFormats[i].second & kAudioFormatFlagIsPacked ) )
1266 testDescription.mBytesPerFrame = 4 * testDescription.mChannelsPerFrame;
1268 testDescription.mBytesPerFrame = testDescription.mBitsPerChannel/8 * testDescription.mChannelsPerFrame;
1269 testDescription.mBytesPerPacket = testDescription.mBytesPerFrame * testDescription.mFramesPerPacket;
1270 result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &testDescription );
1271 if ( result == noErr ) {
1272 setPhysicalFormat = true;
1273 //std::cout << "Updated physical stream format:" << std::endl;
1274 //std::cout << " mBitsPerChan = " << testDescription.mBitsPerChannel << std::endl;
1275 //std::cout << " aligned high = " << (testDescription.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (testDescription.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl;
1276 //std::cout << " bytesPerFrame = " << testDescription.mBytesPerFrame << std::endl;
1277 //std::cout << " sample rate = " << testDescription.mSampleRate << std::endl;
1282 if ( !setPhysicalFormat ) {
1283 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting physical data format for device (" << device << ").";
1284 errorText_ = errorStream_.str();
1287 } // done setting virtual/physical formats.
1289 // Get the stream / device latency.
1291 dataSize = sizeof( UInt32 );
1292 property.mSelector = kAudioDevicePropertyLatency;
1293 if ( AudioObjectHasProperty( id, &property ) == true ) {
1294 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &latency );
1295 if ( result == kAudioHardwareNoError ) stream_.latency[ mode ] = latency;
1297 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting device latency for device (" << device << ").";
1298 errorText_ = errorStream_.str();
1299 error( RtAudioError::WARNING );
1303 // Byte-swapping: According to AudioHardware.h, the stream data will
1304 // always be presented in native-endian format, so we should never
1305 // need to byte swap.
1306 stream_.doByteSwap[mode] = false;
1308 // From the CoreAudio documentation, PCM data must be supplied as
1310 stream_.userFormat = format;
1311 stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
1313 if ( streamCount == 1 )
1314 stream_.nDeviceChannels[mode] = description.mChannelsPerFrame;
1315 else // multiple streams
1316 stream_.nDeviceChannels[mode] = channels;
1317 stream_.nUserChannels[mode] = channels;
1318 stream_.channelOffset[mode] = channelOffset; // offset within a CoreAudio stream
1319 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
1320 else stream_.userInterleaved = true;
1321 stream_.deviceInterleaved[mode] = true;
1322 if ( monoMode == true ) stream_.deviceInterleaved[mode] = false;
1324 // Set flags for buffer conversion.
1325 stream_.doConvertBuffer[mode] = false;
1326 if ( stream_.userFormat != stream_.deviceFormat[mode] )
1327 stream_.doConvertBuffer[mode] = true;
1328 if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
1329 stream_.doConvertBuffer[mode] = true;
1330 if ( streamCount == 1 ) {
1331 if ( stream_.nUserChannels[mode] > 1 &&
1332 stream_.userInterleaved != stream_.deviceInterleaved[mode] )
1333 stream_.doConvertBuffer[mode] = true;
1335 else if ( monoMode && stream_.userInterleaved )
1336 stream_.doConvertBuffer[mode] = true;
1338 // Allocate our CoreHandle structure for the stream.
1339 CoreHandle *handle = 0;
1340 if ( stream_.apiHandle == 0 ) {
1342 handle = new CoreHandle;
1344 catch ( std::bad_alloc& ) {
1345 errorText_ = "RtApiCore::probeDeviceOpen: error allocating CoreHandle memory.";
1349 if ( pthread_cond_init( &handle->condition, NULL ) ) {
1350 errorText_ = "RtApiCore::probeDeviceOpen: error initializing pthread condition variable.";
1353 stream_.apiHandle = (void *) handle;
1356 handle = (CoreHandle *) stream_.apiHandle;
1357 handle->iStream[mode] = firstStream;
1358 handle->nStreams[mode] = streamCount;
1359 handle->id[mode] = id;
1361 // Allocate necessary internal buffers.
1362 unsigned long bufferBytes;
1363 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
1364 // stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
1365 stream_.userBuffer[mode] = (char *) malloc( bufferBytes * sizeof(char) );
1366 memset( stream_.userBuffer[mode], 0, bufferBytes * sizeof(char) );
1367 if ( stream_.userBuffer[mode] == NULL ) {
1368 errorText_ = "RtApiCore::probeDeviceOpen: error allocating user buffer memory.";
1372 // If possible, we will make use of the CoreAudio stream buffers as
1373 // "device buffers". However, we can't do this if using multiple
1375 if ( stream_.doConvertBuffer[mode] && handle->nStreams[mode] > 1 ) {
1377 bool makeBuffer = true;
1378 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
1379 if ( mode == INPUT ) {
1380 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
1381 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
1382 if ( bufferBytes <= bytesOut ) makeBuffer = false;
1387 bufferBytes *= *bufferSize;
1388 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
1389 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
1390 if ( stream_.deviceBuffer == NULL ) {
1391 errorText_ = "RtApiCore::probeDeviceOpen: error allocating device buffer memory.";
1397 stream_.sampleRate = sampleRate;
1398 stream_.device[mode] = device;
1399 stream_.state = STREAM_STOPPED;
1400 stream_.callbackInfo.object = (void *) this;
1402 // Setup the buffer conversion information structure.
1403 if ( stream_.doConvertBuffer[mode] ) {
1404 if ( streamCount > 1 ) setConvertInfo( mode, 0 );
1405 else setConvertInfo( mode, channelOffset );
1408 if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device )
1409 // Only one callback procedure per device.
1410 stream_.mode = DUPLEX;
1412 #if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
1413 result = AudioDeviceCreateIOProcID( id, callbackHandler, (void *) &stream_.callbackInfo, &handle->procId[mode] );
1415 // deprecated in favor of AudioDeviceCreateIOProcID()
1416 result = AudioDeviceAddIOProc( id, callbackHandler, (void *) &stream_.callbackInfo );
1418 if ( result != noErr ) {
1419 errorStream_ << "RtApiCore::probeDeviceOpen: system error setting callback for device (" << device << ").";
1420 errorText_ = errorStream_.str();
1423 if ( stream_.mode == OUTPUT && mode == INPUT )
1424 stream_.mode = DUPLEX;
1426 stream_.mode = mode;
1429 // Setup the device property listener for over/underload.
1430 property.mSelector = kAudioDeviceProcessorOverload;
1431 property.mScope = kAudioObjectPropertyScopeGlobal;
1432 result = AudioObjectAddPropertyListener( id, &property, xrunListener, (void *) handle );
1438 pthread_cond_destroy( &handle->condition );
1440 stream_.apiHandle = 0;
1443 for ( int i=0; i<2; i++ ) {
1444 if ( stream_.userBuffer[i] ) {
1445 free( stream_.userBuffer[i] );
1446 stream_.userBuffer[i] = 0;
1450 if ( stream_.deviceBuffer ) {
1451 free( stream_.deviceBuffer );
1452 stream_.deviceBuffer = 0;
1455 stream_.state = STREAM_CLOSED;
1459 void RtApiCore :: closeStream( void )
1461 if ( stream_.state == STREAM_CLOSED ) {
1462 errorText_ = "RtApiCore::closeStream(): no open stream to close!";
1463 error( RtAudioError::WARNING );
1467 CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
1468 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
1470 AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
1471 kAudioObjectPropertyScopeGlobal,
1472 kAudioObjectPropertyElementMaster };
1474 property.mSelector = kAudioDeviceProcessorOverload;
1475 property.mScope = kAudioObjectPropertyScopeGlobal;
1476 if (AudioObjectRemovePropertyListener( handle->id[0], &property, xrunListener, (void *) handle ) != noErr) {
1477 errorText_ = "RtApiCore::closeStream(): error removing property listener!";
1478 error( RtAudioError::WARNING );
1481 if ( stream_.state == STREAM_RUNNING )
1482 AudioDeviceStop( handle->id[0], callbackHandler );
1483 #if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
1484 AudioDeviceDestroyIOProcID( handle->id[0], handle->procId[0] );
1486 // deprecated in favor of AudioDeviceDestroyIOProcID()
1487 AudioDeviceRemoveIOProc( handle->id[0], callbackHandler );
1491 if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
1493 AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
1494 kAudioObjectPropertyScopeGlobal,
1495 kAudioObjectPropertyElementMaster };
1497 property.mSelector = kAudioDeviceProcessorOverload;
1498 property.mScope = kAudioObjectPropertyScopeGlobal;
1499 if (AudioObjectRemovePropertyListener( handle->id[1], &property, xrunListener, (void *) handle ) != noErr) {
1500 errorText_ = "RtApiCore::closeStream(): error removing property listener!";
1501 error( RtAudioError::WARNING );
1504 if ( stream_.state == STREAM_RUNNING )
1505 AudioDeviceStop( handle->id[1], callbackHandler );
1506 #if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
1507 AudioDeviceDestroyIOProcID( handle->id[1], handle->procId[1] );
1509 // deprecated in favor of AudioDeviceDestroyIOProcID()
1510 AudioDeviceRemoveIOProc( handle->id[1], callbackHandler );
1514 for ( int i=0; i<2; i++ ) {
1515 if ( stream_.userBuffer[i] ) {
1516 free( stream_.userBuffer[i] );
1517 stream_.userBuffer[i] = 0;
1521 if ( stream_.deviceBuffer ) {
1522 free( stream_.deviceBuffer );
1523 stream_.deviceBuffer = 0;
1526 // Destroy pthread condition variable.
1527 pthread_cond_destroy( &handle->condition );
1529 stream_.apiHandle = 0;
1531 stream_.mode = UNINITIALIZED;
1532 stream_.state = STREAM_CLOSED;
1535 void RtApiCore :: startStream( void )
1538 if ( stream_.state == STREAM_RUNNING ) {
1539 errorText_ = "RtApiCore::startStream(): the stream is already running!";
1540 error( RtAudioError::WARNING );
1544 OSStatus result = noErr;
1545 CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
1546 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
1548 result = AudioDeviceStart( handle->id[0], callbackHandler );
1549 if ( result != noErr ) {
1550 errorStream_ << "RtApiCore::startStream: system error (" << getErrorCode( result ) << ") starting callback procedure on device (" << stream_.device[0] << ").";
1551 errorText_ = errorStream_.str();
1556 if ( stream_.mode == INPUT ||
1557 ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
1559 result = AudioDeviceStart( handle->id[1], callbackHandler );
1560 if ( result != noErr ) {
1561 errorStream_ << "RtApiCore::startStream: system error starting input callback procedure on device (" << stream_.device[1] << ").";
1562 errorText_ = errorStream_.str();
1567 handle->drainCounter = 0;
1568 handle->internalDrain = false;
1569 stream_.state = STREAM_RUNNING;
1572 if ( result == noErr ) return;
1573 error( RtAudioError::SYSTEM_ERROR );
1576 void RtApiCore :: stopStream( void )
1579 if ( stream_.state == STREAM_STOPPED ) {
1580 errorText_ = "RtApiCore::stopStream(): the stream is already stopped!";
1581 error( RtAudioError::WARNING );
1585 OSStatus result = noErr;
1586 CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
1587 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
1589 if ( handle->drainCounter == 0 ) {
1590 handle->drainCounter = 2;
1591 pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
1594 result = AudioDeviceStop( handle->id[0], callbackHandler );
1595 if ( result != noErr ) {
1596 errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping callback procedure on device (" << stream_.device[0] << ").";
1597 errorText_ = errorStream_.str();
1602 if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
1604 result = AudioDeviceStop( handle->id[1], callbackHandler );
1605 if ( result != noErr ) {
1606 errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping input callback procedure on device (" << stream_.device[1] << ").";
1607 errorText_ = errorStream_.str();
1612 stream_.state = STREAM_STOPPED;
1615 if ( result == noErr ) return;
1616 error( RtAudioError::SYSTEM_ERROR );
1619 void RtApiCore :: abortStream( void )
1622 if ( stream_.state == STREAM_STOPPED ) {
1623 errorText_ = "RtApiCore::abortStream(): the stream is already stopped!";
1624 error( RtAudioError::WARNING );
1628 CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
1629 handle->drainCounter = 2;
1634 // This function will be called by a spawned thread when the user
1635 // callback function signals that the stream should be stopped or
1636 // aborted. It is better to handle it this way because the
1637 // callbackEvent() function probably should return before the AudioDeviceStop()
1638 // function is called.
1639 static void *coreStopStream( void *ptr )
1641 CallbackInfo *info = (CallbackInfo *) ptr;
1642 RtApiCore *object = (RtApiCore *) info->object;
1644 object->stopStream();
1645 pthread_exit( NULL );
1648 bool RtApiCore :: callbackEvent( AudioDeviceID deviceId,
1649 const AudioBufferList *inBufferList,
1650 const AudioBufferList *outBufferList )
1652 if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;
1653 if ( stream_.state == STREAM_CLOSED ) {
1654 errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";
1655 error( RtAudioError::WARNING );
1659 CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
1660 CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
1662 // Check if we were draining the stream and signal is finished.
1663 if ( handle->drainCounter > 3 ) {
1664 ThreadHandle threadId;
1666 stream_.state = STREAM_STOPPING;
1667 if ( handle->internalDrain == true )
1668 pthread_create( &threadId, NULL, coreStopStream, info );
1669 else // external call to stopStream()
1670 pthread_cond_signal( &handle->condition );
1674 AudioDeviceID outputDevice = handle->id[0];
1676 // Invoke user callback to get fresh output data UNLESS we are
1677 // draining stream or duplex mode AND the input/output devices are
1678 // different AND this function is called for the input device.
1679 if ( handle->drainCounter == 0 && ( stream_.mode != DUPLEX || deviceId == outputDevice ) ) {
1680 RtAudioCallback callback = (RtAudioCallback) info->callback;
1681 double streamTime = getStreamTime();
1682 RtAudioStreamStatus status = 0;
1683 if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
1684 status |= RTAUDIO_OUTPUT_UNDERFLOW;
1685 handle->xrun[0] = false;
1687 if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
1688 status |= RTAUDIO_INPUT_OVERFLOW;
1689 handle->xrun[1] = false;
1692 int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
1693 stream_.bufferSize, streamTime, status, info->userData );
1694 if ( cbReturnValue == 2 ) {
1695 stream_.state = STREAM_STOPPING;
1696 handle->drainCounter = 2;
1700 else if ( cbReturnValue == 1 ) {
1701 handle->drainCounter = 1;
1702 handle->internalDrain = true;
1706 if ( stream_.mode == OUTPUT || ( stream_.mode == DUPLEX && deviceId == outputDevice ) ) {
1708 if ( handle->drainCounter > 1 ) { // write zeros to the output stream
1710 if ( handle->nStreams[0] == 1 ) {
1711 memset( outBufferList->mBuffers[handle->iStream[0]].mData,
1713 outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );
1715 else { // fill multiple streams with zeros
1716 for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) {
1717 memset( outBufferList->mBuffers[handle->iStream[0]+i].mData,
1719 outBufferList->mBuffers[handle->iStream[0]+i].mDataByteSize );
1723 else if ( handle->nStreams[0] == 1 ) {
1724 if ( stream_.doConvertBuffer[0] ) { // convert directly to CoreAudio stream buffer
1725 convertBuffer( (char *) outBufferList->mBuffers[handle->iStream[0]].mData,
1726 stream_.userBuffer[0], stream_.convertInfo[0] );
1728 else { // copy from user buffer
1729 memcpy( outBufferList->mBuffers[handle->iStream[0]].mData,
1730 stream_.userBuffer[0],
1731 outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );
1734 else { // fill multiple streams
1735 Float32 *inBuffer = (Float32 *) stream_.userBuffer[0];
1736 if ( stream_.doConvertBuffer[0] ) {
1737 convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
1738 inBuffer = (Float32 *) stream_.deviceBuffer;
1741 if ( stream_.deviceInterleaved[0] == false ) { // mono mode
1742 UInt32 bufferBytes = outBufferList->mBuffers[handle->iStream[0]].mDataByteSize;
1743 for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
1744 memcpy( outBufferList->mBuffers[handle->iStream[0]+i].mData,
1745 (void *)&inBuffer[i*stream_.bufferSize], bufferBytes );
1748 else { // fill multiple multi-channel streams with interleaved data
1749 UInt32 streamChannels, channelsLeft, inJump, outJump, inOffset;
1752 bool inInterleaved = ( stream_.userInterleaved ) ? true : false;
1753 UInt32 inChannels = stream_.nUserChannels[0];
1754 if ( stream_.doConvertBuffer[0] ) {
1755 inInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode
1756 inChannels = stream_.nDeviceChannels[0];
1759 if ( inInterleaved ) inOffset = 1;
1760 else inOffset = stream_.bufferSize;
1762 channelsLeft = inChannels;
1763 for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) {
1765 out = (Float32 *) outBufferList->mBuffers[handle->iStream[0]+i].mData;
1766 streamChannels = outBufferList->mBuffers[handle->iStream[0]+i].mNumberChannels;
1769 // Account for possible channel offset in first stream
1770 if ( i == 0 && stream_.channelOffset[0] > 0 ) {
1771 streamChannels -= stream_.channelOffset[0];
1772 outJump = stream_.channelOffset[0];
1776 // Account for possible unfilled channels at end of the last stream
1777 if ( streamChannels > channelsLeft ) {
1778 outJump = streamChannels - channelsLeft;
1779 streamChannels = channelsLeft;
1782 // Determine input buffer offsets and skips
1783 if ( inInterleaved ) {
1784 inJump = inChannels;
1785 in += inChannels - channelsLeft;
1789 in += (inChannels - channelsLeft) * inOffset;
1792 for ( unsigned int i=0; i<stream_.bufferSize; i++ ) {
1793 for ( unsigned int j=0; j<streamChannels; j++ ) {
1794 *out++ = in[j*inOffset];
1799 channelsLeft -= streamChannels;
1805 // Don't bother draining input
1806 if ( handle->drainCounter ) {
1807 handle->drainCounter++;
1811 AudioDeviceID inputDevice;
1812 inputDevice = handle->id[1];
1813 if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && deviceId == inputDevice ) ) {
1815 if ( handle->nStreams[1] == 1 ) {
1816 if ( stream_.doConvertBuffer[1] ) { // convert directly from CoreAudio stream buffer
1817 convertBuffer( stream_.userBuffer[1],
1818 (char *) inBufferList->mBuffers[handle->iStream[1]].mData,
1819 stream_.convertInfo[1] );
1821 else { // copy to user buffer
1822 memcpy( stream_.userBuffer[1],
1823 inBufferList->mBuffers[handle->iStream[1]].mData,
1824 inBufferList->mBuffers[handle->iStream[1]].mDataByteSize );
1827 else { // read from multiple streams
1828 Float32 *outBuffer = (Float32 *) stream_.userBuffer[1];
1829 if ( stream_.doConvertBuffer[1] ) outBuffer = (Float32 *) stream_.deviceBuffer;
1831 if ( stream_.deviceInterleaved[1] == false ) { // mono mode
1832 UInt32 bufferBytes = inBufferList->mBuffers[handle->iStream[1]].mDataByteSize;
1833 for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
1834 memcpy( (void *)&outBuffer[i*stream_.bufferSize],
1835 inBufferList->mBuffers[handle->iStream[1]+i].mData, bufferBytes );
1838 else { // read from multiple multi-channel streams
1839 UInt32 streamChannels, channelsLeft, inJump, outJump, outOffset;
1842 bool outInterleaved = ( stream_.userInterleaved ) ? true : false;
1843 UInt32 outChannels = stream_.nUserChannels[1];
1844 if ( stream_.doConvertBuffer[1] ) {
1845 outInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode
1846 outChannels = stream_.nDeviceChannels[1];
1849 if ( outInterleaved ) outOffset = 1;
1850 else outOffset = stream_.bufferSize;
1852 channelsLeft = outChannels;
1853 for ( unsigned int i=0; i<handle->nStreams[1]; i++ ) {
1855 in = (Float32 *) inBufferList->mBuffers[handle->iStream[1]+i].mData;
1856 streamChannels = inBufferList->mBuffers[handle->iStream[1]+i].mNumberChannels;
1859 // Account for possible channel offset in first stream
1860 if ( i == 0 && stream_.channelOffset[1] > 0 ) {
1861 streamChannels -= stream_.channelOffset[1];
1862 inJump = stream_.channelOffset[1];
1866 // Account for possible unread channels at end of the last stream
1867 if ( streamChannels > channelsLeft ) {
1868 inJump = streamChannels - channelsLeft;
1869 streamChannels = channelsLeft;
1872 // Determine output buffer offsets and skips
1873 if ( outInterleaved ) {
1874 outJump = outChannels;
1875 out += outChannels - channelsLeft;
1879 out += (outChannels - channelsLeft) * outOffset;
1882 for ( unsigned int i=0; i<stream_.bufferSize; i++ ) {
1883 for ( unsigned int j=0; j<streamChannels; j++ ) {
1884 out[j*outOffset] = *in++;
1889 channelsLeft -= streamChannels;
1893 if ( stream_.doConvertBuffer[1] ) { // convert from our internal "device" buffer
1894 convertBuffer( stream_.userBuffer[1],
1895 stream_.deviceBuffer,
1896 stream_.convertInfo[1] );
1902 //MUTEX_UNLOCK( &stream_.mutex );
1904 RtApi::tickStreamTime();
1908 const char* RtApiCore :: getErrorCode( OSStatus code )
1912 case kAudioHardwareNotRunningError:
1913 return "kAudioHardwareNotRunningError";
1915 case kAudioHardwareUnspecifiedError:
1916 return "kAudioHardwareUnspecifiedError";
1918 case kAudioHardwareUnknownPropertyError:
1919 return "kAudioHardwareUnknownPropertyError";
1921 case kAudioHardwareBadPropertySizeError:
1922 return "kAudioHardwareBadPropertySizeError";
1924 case kAudioHardwareIllegalOperationError:
1925 return "kAudioHardwareIllegalOperationError";
1927 case kAudioHardwareBadObjectError:
1928 return "kAudioHardwareBadObjectError";
1930 case kAudioHardwareBadDeviceError:
1931 return "kAudioHardwareBadDeviceError";
1933 case kAudioHardwareBadStreamError:
1934 return "kAudioHardwareBadStreamError";
1936 case kAudioHardwareUnsupportedOperationError:
1937 return "kAudioHardwareUnsupportedOperationError";
1939 case kAudioDeviceUnsupportedFormatError:
1940 return "kAudioDeviceUnsupportedFormatError";
1942 case kAudioDevicePermissionsError:
1943 return "kAudioDevicePermissionsError";
1946 return "CoreAudio unknown error";
1950 //******************** End of __MACOSX_CORE__ *********************//
1953 #if defined(__UNIX_JACK__)
1955 // JACK is a low-latency audio server, originally written for the
1956 // GNU/Linux operating system and now also ported to OS-X. It can
1957 // connect a number of different applications to an audio device, as
1958 // well as allowing them to share audio between themselves.
1960 // When using JACK with RtAudio, "devices" refer to JACK clients that
1961 // have ports connected to the server. The JACK server is typically
1962 // started in a terminal as follows:
1964 // .jackd -d alsa -d hw:0
1966 // or through an interface program such as qjackctl. Many of the
1967 // parameters normally set for a stream are fixed by the JACK server
1968 // and can be specified when the JACK server is started. In
1971 // .jackd -d alsa -d hw:0 -r 44100 -p 512 -n 4
1973 // specifies a sample rate of 44100 Hz, a buffer size of 512 sample
1974 // frames, and number of buffers = 4. Once the server is running, it
1975 // is not possible to override these values. If the values are not
1976 // specified in the command-line, the JACK server uses default values.
1978 // The JACK server does not have to be running when an instance of
1979 // RtApiJack is created, though the function getDeviceCount() will
1980 // report 0 devices found until JACK has been started. When no
1981 // devices are available (i.e., the JACK server is not running), a
1982 // stream cannot be opened.
1984 #include <jack/jack.h>
1988 // A structure to hold various information related to the Jack API
1991 jack_client_t *client;
1992 jack_port_t **ports[2];
1993 std::string deviceName[2];
1995 pthread_cond_t condition;
1996 int drainCounter; // Tracks callback counts when draining
1997 bool internalDrain; // Indicates if stop is initiated from callback or not.
2000 :client(0), drainCounter(0), internalDrain(false) { ports[0] = 0; ports[1] = 0; xrun[0] = false; xrun[1] = false; }
2003 #if !defined(__RTAUDIO_DEBUG__)
2004 static void jackSilentError( const char * ) {};
2007 RtApiJack :: RtApiJack()
2008 :shouldAutoconnect_(true) {
2009 // Nothing to do here.
2010 #if !defined(__RTAUDIO_DEBUG__)
2011 // Turn off Jack's internal error reporting.
2012 jack_set_error_function( &jackSilentError );
2016 RtApiJack :: ~RtApiJack()
2018 if ( stream_.state != STREAM_CLOSED ) closeStream();
2021 unsigned int RtApiJack :: getDeviceCount( void )
2023 // See if we can become a jack client.
2024 jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption;
2025 jack_status_t *status = NULL;
2026 jack_client_t *client = jack_client_open( "RtApiJackCount", options, status );
2027 if ( client == 0 ) return 0;
2030 std::string port, previousPort;
2031 unsigned int nChannels = 0, nDevices = 0;
2032 ports = jack_get_ports( client, NULL, JACK_DEFAULT_AUDIO_TYPE, 0 );
2034 // Parse the port names up to the first colon (:).
2037 port = (char *) ports[ nChannels ];
2038 iColon = port.find(":");
2039 if ( iColon != std::string::npos ) {
2040 port = port.substr( 0, iColon + 1 );
2041 if ( port != previousPort ) {
2043 previousPort = port;
2046 } while ( ports[++nChannels] );
2050 jack_client_close( client );
2054 RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device )
2056 RtAudio::DeviceInfo info;
2057 info.probed = false;
2059 jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption
2060 jack_status_t *status = NULL;
2061 jack_client_t *client = jack_client_open( "RtApiJackInfo", options, status );
2062 if ( client == 0 ) {
2063 errorText_ = "RtApiJack::getDeviceInfo: Jack server not found or connection error!";
2064 error( RtAudioError::WARNING );
2069 std::string port, previousPort;
2070 unsigned int nPorts = 0, nDevices = 0;
2071 ports = jack_get_ports( client, NULL, JACK_DEFAULT_AUDIO_TYPE, 0 );
2073 // Parse the port names up to the first colon (:).
2076 port = (char *) ports[ nPorts ];
2077 iColon = port.find(":");
2078 if ( iColon != std::string::npos ) {
2079 port = port.substr( 0, iColon );
2080 if ( port != previousPort ) {
2081 if ( nDevices == device ) info.name = port;
2083 previousPort = port;
2086 } while ( ports[++nPorts] );
2090 if ( device >= nDevices ) {
2091 jack_client_close( client );
2092 errorText_ = "RtApiJack::getDeviceInfo: device ID is invalid!";
2093 error( RtAudioError::INVALID_USE );
2097 // Get the current jack server sample rate.
2098 info.sampleRates.clear();
2100 info.preferredSampleRate = jack_get_sample_rate( client );
2101 info.sampleRates.push_back( info.preferredSampleRate );
2103 // Count the available ports containing the client name as device
2104 // channels. Jack "input ports" equal RtAudio output channels.
2105 unsigned int nChannels = 0;
2106 ports = jack_get_ports( client, info.name.c_str(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput );
2108 while ( ports[ nChannels ] ) nChannels++;
2110 info.outputChannels = nChannels;
2113 // Jack "output ports" equal RtAudio input channels.
2115 ports = jack_get_ports( client, info.name.c_str(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput );
2117 while ( ports[ nChannels ] ) nChannels++;
2119 info.inputChannels = nChannels;
2122 if ( info.outputChannels == 0 && info.inputChannels == 0 ) {
2123 jack_client_close(client);
2124 errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!";
2125 error( RtAudioError::WARNING );
2129 // If device opens for both playback and capture, we determine the channels.
2130 if ( info.outputChannels > 0 && info.inputChannels > 0 )
2131 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
2133 // Jack always uses 32-bit floats.
2134 info.nativeFormats = RTAUDIO_FLOAT32;
2136 // Jack doesn't provide default devices so we'll use the first available one.
2137 if ( device == 0 && info.outputChannels > 0 )
2138 info.isDefaultOutput = true;
2139 if ( device == 0 && info.inputChannels > 0 )
2140 info.isDefaultInput = true;
2142 jack_client_close(client);
2147 static int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer )
2149 CallbackInfo *info = (CallbackInfo *) infoPointer;
2151 RtApiJack *object = (RtApiJack *) info->object;
2152 if ( object->callbackEvent( (unsigned long) nframes ) == false ) return 1;
2157 // This function will be called by a spawned thread when the Jack
2158 // server signals that it is shutting down. It is necessary to handle
2159 // it this way because the jackShutdown() function must return before
2160 // the jack_deactivate() function (in closeStream()) will return.
2161 static void *jackCloseStream( void *ptr )
2163 CallbackInfo *info = (CallbackInfo *) ptr;
2164 RtApiJack *object = (RtApiJack *) info->object;
2166 object->closeStream();
2168 pthread_exit( NULL );
2170 static void jackShutdown( void *infoPointer )
2172 CallbackInfo *info = (CallbackInfo *) infoPointer;
2173 RtApiJack *object = (RtApiJack *) info->object;
2175 // Check current stream state. If stopped, then we'll assume this
2176 // was called as a result of a call to RtApiJack::stopStream (the
2177 // deactivation of a client handle causes this function to be called).
2178 // If not, we'll assume the Jack server is shutting down or some
2179 // other problem occurred and we should close the stream.
2180 if ( object->isStreamRunning() == false ) return;
2182 ThreadHandle threadId;
2183 pthread_create( &threadId, NULL, jackCloseStream, info );
2184 std::cerr << "\nRtApiJack: the Jack server is shutting down this client ... stream stopped and closed!!\n" << std::endl;
2187 static int jackXrun( void *infoPointer )
2189 JackHandle *handle = *((JackHandle **) infoPointer);
2191 if ( handle->ports[0] ) handle->xrun[0] = true;
2192 if ( handle->ports[1] ) handle->xrun[1] = true;
2197 bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
2198 unsigned int firstChannel, unsigned int sampleRate,
2199 RtAudioFormat format, unsigned int *bufferSize,
2200 RtAudio::StreamOptions *options )
2202 JackHandle *handle = (JackHandle *) stream_.apiHandle;
2204 // Look for jack server and try to become a client (only do once per stream).
2205 jack_client_t *client = 0;
2206 if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) {
2207 jack_options_t jackoptions = (jack_options_t) ( JackNoStartServer ); //JackNullOption;
2208 jack_status_t *status = NULL;
2209 if ( options && !options->streamName.empty() )
2210 client = jack_client_open( options->streamName.c_str(), jackoptions, status );
2212 client = jack_client_open( "RtApiJack", jackoptions, status );
2213 if ( client == 0 ) {
2214 errorText_ = "RtApiJack::probeDeviceOpen: Jack server not found or connection error!";
2215 error( RtAudioError::WARNING );
2220 // The handle must have been created on an earlier pass.
2221 client = handle->client;
2225 std::string port, previousPort, deviceName;
2226 unsigned int nPorts = 0, nDevices = 0;
2227 ports = jack_get_ports( client, NULL, JACK_DEFAULT_AUDIO_TYPE, 0 );
2229 // Parse the port names up to the first colon (:).
2232 port = (char *) ports[ nPorts ];
2233 iColon = port.find(":");
2234 if ( iColon != std::string::npos ) {
2235 port = port.substr( 0, iColon );
2236 if ( port != previousPort ) {
2237 if ( nDevices == device ) deviceName = port;
2239 previousPort = port;
2242 } while ( ports[++nPorts] );
2246 if ( device >= nDevices ) {
2247 errorText_ = "RtApiJack::probeDeviceOpen: device ID is invalid!";
2251 unsigned long flag = JackPortIsInput;
2252 if ( mode == INPUT ) flag = JackPortIsOutput;
2254 if ( ! (options && (options->flags & RTAUDIO_JACK_DONT_CONNECT)) ) {
2255 // Count the available ports containing the client name as device
2256 // channels. Jack "input ports" equal RtAudio output channels.
2257 unsigned int nChannels = 0;
2258 ports = jack_get_ports( client, deviceName.c_str(), JACK_DEFAULT_AUDIO_TYPE, flag );
2260 while ( ports[ nChannels ] ) nChannels++;
2263 // Compare the jack ports for specified client to the requested number of channels.
2264 if ( nChannels < (channels + firstChannel) ) {
2265 errorStream_ << "RtApiJack::probeDeviceOpen: requested number of channels (" << channels << ") + offset (" << firstChannel << ") not found for specified device (" << device << ":" << deviceName << ").";
2266 errorText_ = errorStream_.str();
2271 // Check the jack server sample rate.
2272 unsigned int jackRate = jack_get_sample_rate( client );
2273 if ( sampleRate != jackRate ) {
2274 jack_client_close( client );
2275 errorStream_ << "RtApiJack::probeDeviceOpen: the requested sample rate (" << sampleRate << ") is different than the JACK server rate (" << jackRate << ").";
2276 errorText_ = errorStream_.str();
2279 stream_.sampleRate = jackRate;
2281 // Get the latency of the JACK port.
2282 ports = jack_get_ports( client, deviceName.c_str(), JACK_DEFAULT_AUDIO_TYPE, flag );
2283 if ( ports[ firstChannel ] ) {
2285 jack_latency_callback_mode_t cbmode = (mode == INPUT ? JackCaptureLatency : JackPlaybackLatency);
2286 // the range (usually the min and max are equal)
2287 jack_latency_range_t latrange; latrange.min = latrange.max = 0;
2288 // get the latency range
2289 jack_port_get_latency_range( jack_port_by_name( client, ports[firstChannel] ), cbmode, &latrange );
2290 // be optimistic, use the min!
2291 stream_.latency[mode] = latrange.min;
2292 //stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) );
2296 // The jack server always uses 32-bit floating-point data.
2297 stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
2298 stream_.userFormat = format;
2300 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
2301 else stream_.userInterleaved = true;
2303 // Jack always uses non-interleaved buffers.
2304 stream_.deviceInterleaved[mode] = false;
2306 // Jack always provides host byte-ordered data.
2307 stream_.doByteSwap[mode] = false;
2309 // Get the buffer size. The buffer size and number of buffers
2310 // (periods) is set when the jack server is started.
2311 stream_.bufferSize = (int) jack_get_buffer_size( client );
2312 *bufferSize = stream_.bufferSize;
2314 stream_.nDeviceChannels[mode] = channels;
2315 stream_.nUserChannels[mode] = channels;
2317 // Set flags for buffer conversion.
2318 stream_.doConvertBuffer[mode] = false;
2319 if ( stream_.userFormat != stream_.deviceFormat[mode] )
2320 stream_.doConvertBuffer[mode] = true;
2321 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
2322 stream_.nUserChannels[mode] > 1 )
2323 stream_.doConvertBuffer[mode] = true;
2325 // Allocate our JackHandle structure for the stream.
2326 if ( handle == 0 ) {
2328 handle = new JackHandle;
2330 catch ( std::bad_alloc& ) {
2331 errorText_ = "RtApiJack::probeDeviceOpen: error allocating JackHandle memory.";
2335 if ( pthread_cond_init(&handle->condition, NULL) ) {
2336 errorText_ = "RtApiJack::probeDeviceOpen: error initializing pthread condition variable.";
2339 stream_.apiHandle = (void *) handle;
2340 handle->client = client;
2342 handle->deviceName[mode] = deviceName;
2344 // Allocate necessary internal buffers.
2345 unsigned long bufferBytes;
2346 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
2347 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
2348 if ( stream_.userBuffer[mode] == NULL ) {
2349 errorText_ = "RtApiJack::probeDeviceOpen: error allocating user buffer memory.";
2353 if ( stream_.doConvertBuffer[mode] ) {
2355 bool makeBuffer = true;
2356 if ( mode == OUTPUT )
2357 bufferBytes = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
2358 else { // mode == INPUT
2359 bufferBytes = stream_.nDeviceChannels[1] * formatBytes( stream_.deviceFormat[1] );
2360 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
2361 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]);
2362 if ( bufferBytes < bytesOut ) makeBuffer = false;
2367 bufferBytes *= *bufferSize;
2368 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
2369 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
2370 if ( stream_.deviceBuffer == NULL ) {
2371 errorText_ = "RtApiJack::probeDeviceOpen: error allocating device buffer memory.";
2377 // Allocate memory for the Jack ports (channels) identifiers.
2378 handle->ports[mode] = (jack_port_t **) malloc ( sizeof (jack_port_t *) * channels );
2379 if ( handle->ports[mode] == NULL ) {
2380 errorText_ = "RtApiJack::probeDeviceOpen: error allocating port memory.";
2384 stream_.device[mode] = device;
2385 stream_.channelOffset[mode] = firstChannel;
2386 stream_.state = STREAM_STOPPED;
2387 stream_.callbackInfo.object = (void *) this;
2389 if ( stream_.mode == OUTPUT && mode == INPUT )
2390 // We had already set up the stream for output.
2391 stream_.mode = DUPLEX;
2393 stream_.mode = mode;
2394 jack_set_process_callback( handle->client, jackCallbackHandler, (void *) &stream_.callbackInfo );
2395 jack_set_xrun_callback( handle->client, jackXrun, (void *) &stream_.apiHandle );
2396 jack_on_shutdown( handle->client, jackShutdown, (void *) &stream_.callbackInfo );
2399 // Register our ports.
2401 if ( mode == OUTPUT ) {
2402 for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
2403 snprintf( label, 64, "outport %d", i );
2404 handle->ports[0][i] = jack_port_register( handle->client, (const char *)label,
2405 JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0 );
2409 for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
2410 snprintf( label, 64, "inport %d", i );
2411 handle->ports[1][i] = jack_port_register( handle->client, (const char *)label,
2412 JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0 );
2416 // Setup the buffer conversion information structure. We don't use
2417 // buffers to do channel offsets, so we override that parameter
2419 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );
2421 if ( options && options->flags & RTAUDIO_JACK_DONT_CONNECT ) shouldAutoconnect_ = false;
2427 pthread_cond_destroy( &handle->condition );
2428 jack_client_close( handle->client );
2430 if ( handle->ports[0] ) free( handle->ports[0] );
2431 if ( handle->ports[1] ) free( handle->ports[1] );
2434 stream_.apiHandle = 0;
2437 for ( int i=0; i<2; i++ ) {
2438 if ( stream_.userBuffer[i] ) {
2439 free( stream_.userBuffer[i] );
2440 stream_.userBuffer[i] = 0;
2444 if ( stream_.deviceBuffer ) {
2445 free( stream_.deviceBuffer );
2446 stream_.deviceBuffer = 0;
2452 void RtApiJack :: closeStream( void )
2454 if ( stream_.state == STREAM_CLOSED ) {
2455 errorText_ = "RtApiJack::closeStream(): no open stream to close!";
2456 error( RtAudioError::WARNING );
2460 JackHandle *handle = (JackHandle *) stream_.apiHandle;
2463 if ( stream_.state == STREAM_RUNNING )
2464 jack_deactivate( handle->client );
2466 jack_client_close( handle->client );
2470 if ( handle->ports[0] ) free( handle->ports[0] );
2471 if ( handle->ports[1] ) free( handle->ports[1] );
2472 pthread_cond_destroy( &handle->condition );
2474 stream_.apiHandle = 0;
2477 for ( int i=0; i<2; i++ ) {
2478 if ( stream_.userBuffer[i] ) {
2479 free( stream_.userBuffer[i] );
2480 stream_.userBuffer[i] = 0;
2484 if ( stream_.deviceBuffer ) {
2485 free( stream_.deviceBuffer );
2486 stream_.deviceBuffer = 0;
2489 stream_.mode = UNINITIALIZED;
2490 stream_.state = STREAM_CLOSED;
2493 void RtApiJack :: startStream( void )
2496 if ( stream_.state == STREAM_RUNNING ) {
2497 errorText_ = "RtApiJack::startStream(): the stream is already running!";
2498 error( RtAudioError::WARNING );
2502 JackHandle *handle = (JackHandle *) stream_.apiHandle;
2503 int result = jack_activate( handle->client );
2505 errorText_ = "RtApiJack::startStream(): unable to activate JACK client!";
2511 // Get the list of available ports.
2512 if ( shouldAutoconnect_ && (stream_.mode == OUTPUT || stream_.mode == DUPLEX) ) {
2514 ports = jack_get_ports( handle->client, handle->deviceName[0].c_str(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput);
2515 if ( ports == NULL) {
2516 errorText_ = "RtApiJack::startStream(): error determining available JACK input ports!";
2520 // Now make the port connections. Since RtAudio wasn't designed to
2521 // allow the user to select particular channels of a device, we'll
2522 // just open the first "nChannels" ports with offset.
2523 for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
2525 if ( ports[ stream_.channelOffset[0] + i ] )
2526 result = jack_connect( handle->client, jack_port_name( handle->ports[0][i] ), ports[ stream_.channelOffset[0] + i ] );
2529 errorText_ = "RtApiJack::startStream(): error connecting output ports!";
2536 if ( shouldAutoconnect_ && (stream_.mode == INPUT || stream_.mode == DUPLEX) ) {
2538 ports = jack_get_ports( handle->client, handle->deviceName[1].c_str(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput );
2539 if ( ports == NULL) {
2540 errorText_ = "RtApiJack::startStream(): error determining available JACK output ports!";
2544 // Now make the port connections. See note above.
2545 for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
2547 if ( ports[ stream_.channelOffset[1] + i ] )
2548 result = jack_connect( handle->client, ports[ stream_.channelOffset[1] + i ], jack_port_name( handle->ports[1][i] ) );
2551 errorText_ = "RtApiJack::startStream(): error connecting input ports!";
2558 handle->drainCounter = 0;
2559 handle->internalDrain = false;
2560 stream_.state = STREAM_RUNNING;
2563 if ( result == 0 ) return;
2564 error( RtAudioError::SYSTEM_ERROR );
2567 void RtApiJack :: stopStream( void )
2570 if ( stream_.state == STREAM_STOPPED ) {
2571 errorText_ = "RtApiJack::stopStream(): the stream is already stopped!";
2572 error( RtAudioError::WARNING );
2576 JackHandle *handle = (JackHandle *) stream_.apiHandle;
2577 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
2579 if ( handle->drainCounter == 0 ) {
2580 handle->drainCounter = 2;
2581 pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
2585 jack_deactivate( handle->client );
2586 stream_.state = STREAM_STOPPED;
2589 void RtApiJack :: abortStream( void )
2592 if ( stream_.state == STREAM_STOPPED ) {
2593 errorText_ = "RtApiJack::abortStream(): the stream is already stopped!";
2594 error( RtAudioError::WARNING );
2598 JackHandle *handle = (JackHandle *) stream_.apiHandle;
2599 handle->drainCounter = 2;
2604 // This function will be called by a spawned thread when the user
2605 // callback function signals that the stream should be stopped or
2606 // aborted. It is necessary to handle it this way because the
2607 // callbackEvent() function must return before the jack_deactivate()
2608 // function will return.
2609 static void *jackStopStream( void *ptr )
2611 CallbackInfo *info = (CallbackInfo *) ptr;
2612 RtApiJack *object = (RtApiJack *) info->object;
2614 object->stopStream();
2615 pthread_exit( NULL );
2618 bool RtApiJack :: callbackEvent( unsigned long nframes )
2620 if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;
2621 if ( stream_.state == STREAM_CLOSED ) {
2622 errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";
2623 error( RtAudioError::WARNING );
2626 if ( stream_.bufferSize != nframes ) {
2627 errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!";
2628 error( RtAudioError::WARNING );
2632 CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
2633 JackHandle *handle = (JackHandle *) stream_.apiHandle;
2635 // Check if we were draining the stream and signal is finished.
2636 if ( handle->drainCounter > 3 ) {
2637 ThreadHandle threadId;
2639 stream_.state = STREAM_STOPPING;
2640 if ( handle->internalDrain == true )
2641 pthread_create( &threadId, NULL, jackStopStream, info );
2643 pthread_cond_signal( &handle->condition );
2647 // Invoke user callback first, to get fresh output data.
2648 if ( handle->drainCounter == 0 ) {
2649 RtAudioCallback callback = (RtAudioCallback) info->callback;
2650 double streamTime = getStreamTime();
2651 RtAudioStreamStatus status = 0;
2652 if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
2653 status |= RTAUDIO_OUTPUT_UNDERFLOW;
2654 handle->xrun[0] = false;
2656 if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
2657 status |= RTAUDIO_INPUT_OVERFLOW;
2658 handle->xrun[1] = false;
2660 int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
2661 stream_.bufferSize, streamTime, status, info->userData );
2662 if ( cbReturnValue == 2 ) {
2663 stream_.state = STREAM_STOPPING;
2664 handle->drainCounter = 2;
2666 pthread_create( &id, NULL, jackStopStream, info );
2669 else if ( cbReturnValue == 1 ) {
2670 handle->drainCounter = 1;
2671 handle->internalDrain = true;
2675 jack_default_audio_sample_t *jackbuffer;
2676 unsigned long bufferBytes = nframes * sizeof( jack_default_audio_sample_t );
2677 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
2679 if ( handle->drainCounter > 1 ) { // write zeros to the output stream
2681 for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
2682 jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
2683 memset( jackbuffer, 0, bufferBytes );
2687 else if ( stream_.doConvertBuffer[0] ) {
2689 convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
2691 for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
2692 jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
2693 memcpy( jackbuffer, &stream_.deviceBuffer[i*bufferBytes], bufferBytes );
2696 else { // no buffer conversion
2697 for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
2698 jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
2699 memcpy( jackbuffer, &stream_.userBuffer[0][i*bufferBytes], bufferBytes );
2704 // Don't bother draining input
2705 if ( handle->drainCounter ) {
2706 handle->drainCounter++;
2710 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
2712 if ( stream_.doConvertBuffer[1] ) {
2713 for ( unsigned int i=0; i<stream_.nDeviceChannels[1]; i++ ) {
2714 jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );
2715 memcpy( &stream_.deviceBuffer[i*bufferBytes], jackbuffer, bufferBytes );
2717 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
2719 else { // no buffer conversion
2720 for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
2721 jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );
2722 memcpy( &stream_.userBuffer[1][i*bufferBytes], jackbuffer, bufferBytes );
2728 RtApi::tickStreamTime();
2731 //******************** End of __UNIX_JACK__ *********************//
2734 #if defined(__WINDOWS_ASIO__) // ASIO API on Windows
2736 // The ASIO API is designed around a callback scheme, so this
2737 // implementation is similar to that used for OS-X CoreAudio and Linux
2738 // Jack. The primary constraint with ASIO is that it only allows
2739 // access to a single driver at a time. Thus, it is not possible to
2740 // have more than one simultaneous RtAudio stream.
2742 // This implementation also requires a number of external ASIO files
2743 // and a few global variables. The ASIO callback scheme does not
2744 // allow for the passing of user data, so we must create a global
2745 // pointer to our callbackInfo structure.
2747 // On unix systems, we make use of a pthread condition variable.
2748 // Since there is no equivalent in Windows, I hacked something based
2749 // on information found in
2750 // http://www.cs.wustl.edu/~schmidt/win32-cv-1.html.
2752 #include "asiosys.h"
2754 #include "iasiothiscallresolver.h"
2755 #include "asiodrivers.h"
2758 static AsioDrivers drivers;
2759 static ASIOCallbacks asioCallbacks;
2760 static ASIODriverInfo driverInfo;
2761 static CallbackInfo *asioCallbackInfo;
2762 static bool asioXRun;
2765 int drainCounter; // Tracks callback counts when draining
2766 bool internalDrain; // Indicates if stop is initiated from callback or not.
2767 ASIOBufferInfo *bufferInfos;
2771 :drainCounter(0), internalDrain(false), bufferInfos(0) {}
2774 // Function declarations (definitions at end of section)
2775 static const char* getAsioErrorString( ASIOError result );
2776 static void sampleRateChanged( ASIOSampleRate sRate );
2777 static long asioMessages( long selector, long value, void* message, double* opt );
2779 RtApiAsio :: RtApiAsio()
2781 // ASIO cannot run on a multi-threaded appartment. You can call
2782 // CoInitialize beforehand, but it must be for appartment threading
2783 // (in which case, CoInitilialize will return S_FALSE here).
2784 coInitialized_ = false;
2785 HRESULT hr = CoInitialize( NULL );
2787 errorText_ = "RtApiAsio::ASIO requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)";
2788 error( RtAudioError::WARNING );
2790 coInitialized_ = true;
2792 drivers.removeCurrentDriver();
2793 driverInfo.asioVersion = 2;
2795 // See note in DirectSound implementation about GetDesktopWindow().
2796 driverInfo.sysRef = GetForegroundWindow();
2799 RtApiAsio :: ~RtApiAsio()
2801 if ( stream_.state != STREAM_CLOSED ) closeStream();
2802 if ( coInitialized_ ) CoUninitialize();
2805 unsigned int RtApiAsio :: getDeviceCount( void )
2807 return (unsigned int) drivers.asioGetNumDev();
2810 RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device )
2812 RtAudio::DeviceInfo info;
2813 info.probed = false;
2816 unsigned int nDevices = getDeviceCount();
2817 if ( nDevices == 0 ) {
2818 errorText_ = "RtApiAsio::getDeviceInfo: no devices found!";
2819 error( RtAudioError::INVALID_USE );
2823 if ( device >= nDevices ) {
2824 errorText_ = "RtApiAsio::getDeviceInfo: device ID is invalid!";
2825 error( RtAudioError::INVALID_USE );
2829 // If a stream is already open, we cannot probe other devices. Thus, use the saved results.
2830 if ( stream_.state != STREAM_CLOSED ) {
2831 if ( device >= devices_.size() ) {
2832 errorText_ = "RtApiAsio::getDeviceInfo: device ID was not present before stream was opened.";
2833 error( RtAudioError::WARNING );
2836 return devices_[ device ];
2839 char driverName[32];
2840 ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 );
2841 if ( result != ASE_OK ) {
2842 errorStream_ << "RtApiAsio::getDeviceInfo: unable to get driver name (" << getAsioErrorString( result ) << ").";
2843 errorText_ = errorStream_.str();
2844 error( RtAudioError::WARNING );
2848 info.name = driverName;
2850 if ( !drivers.loadDriver( driverName ) ) {
2851 errorStream_ << "RtApiAsio::getDeviceInfo: unable to load driver (" << driverName << ").";
2852 errorText_ = errorStream_.str();
2853 error( RtAudioError::WARNING );
2857 result = ASIOInit( &driverInfo );
2858 if ( result != ASE_OK ) {
2859 errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
2860 errorText_ = errorStream_.str();
2861 error( RtAudioError::WARNING );
2865 // Determine the device channel information.
2866 long inputChannels, outputChannels;
2867 result = ASIOGetChannels( &inputChannels, &outputChannels );
2868 if ( result != ASE_OK ) {
2869 drivers.removeCurrentDriver();
2870 errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
2871 errorText_ = errorStream_.str();
2872 error( RtAudioError::WARNING );
2876 info.outputChannels = outputChannels;
2877 info.inputChannels = inputChannels;
2878 if ( info.outputChannels > 0 && info.inputChannels > 0 )
2879 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
2881 // Determine the supported sample rates.
2882 info.sampleRates.clear();
2883 for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {
2884 result = ASIOCanSampleRate( (ASIOSampleRate) SAMPLE_RATES[i] );
2885 if ( result == ASE_OK ) {
2886 info.sampleRates.push_back( SAMPLE_RATES[i] );
2888 if ( !info.preferredSampleRate || ( SAMPLE_RATES[i] <= 48000 && SAMPLE_RATES[i] > info.preferredSampleRate ) )
2889 info.preferredSampleRate = SAMPLE_RATES[i];
2893 // Determine supported data types ... just check first channel and assume rest are the same.
2894 ASIOChannelInfo channelInfo;
2895 channelInfo.channel = 0;
2896 channelInfo.isInput = true;
2897 if ( info.inputChannels <= 0 ) channelInfo.isInput = false;
2898 result = ASIOGetChannelInfo( &channelInfo );
2899 if ( result != ASE_OK ) {
2900 drivers.removeCurrentDriver();
2901 errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting driver channel info (" << driverName << ").";
2902 errorText_ = errorStream_.str();
2903 error( RtAudioError::WARNING );
2907 info.nativeFormats = 0;
2908 if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB )
2909 info.nativeFormats |= RTAUDIO_SINT16;
2910 else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB )
2911 info.nativeFormats |= RTAUDIO_SINT32;
2912 else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB )
2913 info.nativeFormats |= RTAUDIO_FLOAT32;
2914 else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB )
2915 info.nativeFormats |= RTAUDIO_FLOAT64;
2916 else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB )
2917 info.nativeFormats |= RTAUDIO_SINT24;
2919 if ( info.outputChannels > 0 )
2920 if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true;
2921 if ( info.inputChannels > 0 )
2922 if ( getDefaultInputDevice() == device ) info.isDefaultInput = true;
2925 drivers.removeCurrentDriver();
2929 static void bufferSwitch( long index, ASIOBool /*processNow*/ )
2931 RtApiAsio *object = (RtApiAsio *) asioCallbackInfo->object;
2932 object->callbackEvent( index );
2935 void RtApiAsio :: saveDeviceInfo( void )
2939 unsigned int nDevices = getDeviceCount();
2940 devices_.resize( nDevices );
2941 for ( unsigned int i=0; i<nDevices; i++ )
2942 devices_[i] = getDeviceInfo( i );
2945 bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
2946 unsigned int firstChannel, unsigned int sampleRate,
2947 RtAudioFormat format, unsigned int *bufferSize,
2948 RtAudio::StreamOptions *options )
2949 {////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
2951 bool isDuplexInput = mode == INPUT && stream_.mode == OUTPUT;
2953 // For ASIO, a duplex stream MUST use the same driver.
2954 if ( isDuplexInput && stream_.device[0] != device ) {
2955 errorText_ = "RtApiAsio::probeDeviceOpen: an ASIO duplex stream must use the same device for input and output!";
2959 char driverName[32];
2960 ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 );
2961 if ( result != ASE_OK ) {
2962 errorStream_ << "RtApiAsio::probeDeviceOpen: unable to get driver name (" << getAsioErrorString( result ) << ").";
2963 errorText_ = errorStream_.str();
2967 // Only load the driver once for duplex stream.
2968 if ( !isDuplexInput ) {
2969 // The getDeviceInfo() function will not work when a stream is open
2970 // because ASIO does not allow multiple devices to run at the same
2971 // time. Thus, we'll probe the system before opening a stream and
2972 // save the results for use by getDeviceInfo().
2973 this->saveDeviceInfo();
2975 if ( !drivers.loadDriver( driverName ) ) {
2976 errorStream_ << "RtApiAsio::probeDeviceOpen: unable to load driver (" << driverName << ").";
2977 errorText_ = errorStream_.str();
2981 result = ASIOInit( &driverInfo );
2982 if ( result != ASE_OK ) {
2983 errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
2984 errorText_ = errorStream_.str();
2989 // keep them before any "goto error", they are used for error cleanup + goto device boundary checks
2990 bool buffersAllocated = false;
2991 AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
2992 unsigned int nChannels;
2995 // Check the device channel count.
2996 long inputChannels, outputChannels;
2997 result = ASIOGetChannels( &inputChannels, &outputChannels );
2998 if ( result != ASE_OK ) {
2999 errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
3000 errorText_ = errorStream_.str();
3004 if ( ( mode == OUTPUT && (channels+firstChannel) > (unsigned int) outputChannels) ||
3005 ( mode == INPUT && (channels+firstChannel) > (unsigned int) inputChannels) ) {
3006 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested channel count (" << channels << ") + offset (" << firstChannel << ").";
3007 errorText_ = errorStream_.str();
3010 stream_.nDeviceChannels[mode] = channels;
3011 stream_.nUserChannels[mode] = channels;
3012 stream_.channelOffset[mode] = firstChannel;
3014 // Verify the sample rate is supported.
3015 result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate );
3016 if ( result != ASE_OK ) {
3017 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested sample rate (" << sampleRate << ").";
3018 errorText_ = errorStream_.str();
3022 // Get the current sample rate
3023 ASIOSampleRate currentRate;
3024 result = ASIOGetSampleRate( ¤tRate );
3025 if ( result != ASE_OK ) {
3026 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error getting sample rate.";
3027 errorText_ = errorStream_.str();
3031 // Set the sample rate only if necessary
3032 if ( currentRate != sampleRate ) {
3033 result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate );
3034 if ( result != ASE_OK ) {
3035 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ").";
3036 errorText_ = errorStream_.str();
3041 // Determine the driver data type.
3042 ASIOChannelInfo channelInfo;
3043 channelInfo.channel = 0;
3044 if ( mode == OUTPUT ) channelInfo.isInput = false;
3045 else channelInfo.isInput = true;
3046 result = ASIOGetChannelInfo( &channelInfo );
3047 if ( result != ASE_OK ) {
3048 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting data format.";
3049 errorText_ = errorStream_.str();
3053 // Assuming WINDOWS host is always little-endian.
3054 stream_.doByteSwap[mode] = false;
3055 stream_.userFormat = format;
3056 stream_.deviceFormat[mode] = 0;
3057 if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) {
3058 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
3059 if ( channelInfo.type == ASIOSTInt16MSB ) stream_.doByteSwap[mode] = true;
3061 else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) {
3062 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
3063 if ( channelInfo.type == ASIOSTInt32MSB ) stream_.doByteSwap[mode] = true;
3065 else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) {
3066 stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
3067 if ( channelInfo.type == ASIOSTFloat32MSB ) stream_.doByteSwap[mode] = true;
3069 else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) {
3070 stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
3071 if ( channelInfo.type == ASIOSTFloat64MSB ) stream_.doByteSwap[mode] = true;
3073 else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB ) {
3074 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
3075 if ( channelInfo.type == ASIOSTInt24MSB ) stream_.doByteSwap[mode] = true;
3078 if ( stream_.deviceFormat[mode] == 0 ) {
3079 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") data format not supported by RtAudio.";
3080 errorText_ = errorStream_.str();
3084 // Set the buffer size. For a duplex stream, this will end up
3085 // setting the buffer size based on the input constraints, which
3087 long minSize, maxSize, preferSize, granularity;
3088 result = ASIOGetBufferSize( &minSize, &maxSize, &preferSize, &granularity );
3089 if ( result != ASE_OK ) {
3090 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting buffer size.";
3091 errorText_ = errorStream_.str();
3095 if ( isDuplexInput ) {
3096 // When this is the duplex input (output was opened before), then we have to use the same
3097 // buffersize as the output, because it might use the preferred buffer size, which most
3098 // likely wasn't passed as input to this. The buffer sizes have to be identically anyway,
3099 // So instead of throwing an error, make them equal. The caller uses the reference
3100 // to the "bufferSize" param as usual to set up processing buffers.
3102 *bufferSize = stream_.bufferSize;
3105 if ( *bufferSize == 0 ) *bufferSize = preferSize;
3106 else if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;
3107 else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;
3108 else if ( granularity == -1 ) {
3109 // Make sure bufferSize is a power of two.
3110 int log2_of_min_size = 0;
3111 int log2_of_max_size = 0;
3113 for ( unsigned int i = 0; i < sizeof(long) * 8; i++ ) {
3114 if ( minSize & ((long)1 << i) ) log2_of_min_size = i;
3115 if ( maxSize & ((long)1 << i) ) log2_of_max_size = i;
3118 long min_delta = std::abs( (long)*bufferSize - ((long)1 << log2_of_min_size) );
3119 int min_delta_num = log2_of_min_size;
3121 for (int i = log2_of_min_size + 1; i <= log2_of_max_size; i++) {
3122 long current_delta = std::abs( (long)*bufferSize - ((long)1 << i) );
3123 if (current_delta < min_delta) {
3124 min_delta = current_delta;
3129 *bufferSize = ( (unsigned int)1 << min_delta_num );
3130 if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;
3131 else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;
3133 else if ( granularity != 0 ) {
3134 // Set to an even multiple of granularity, rounding up.
3135 *bufferSize = (*bufferSize + granularity-1) / granularity * granularity;
3140 // we don't use it anymore, see above!
3141 // Just left it here for the case...
3142 if ( isDuplexInput && stream_.bufferSize != *bufferSize ) {
3143 errorText_ = "RtApiAsio::probeDeviceOpen: input/output buffersize discrepancy!";
3148 stream_.bufferSize = *bufferSize;
3149 stream_.nBuffers = 2;
3151 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
3152 else stream_.userInterleaved = true;
3154 // ASIO always uses non-interleaved buffers.
3155 stream_.deviceInterleaved[mode] = false;
3157 // Allocate, if necessary, our AsioHandle structure for the stream.
3158 if ( handle == 0 ) {
3160 handle = new AsioHandle;
3162 catch ( std::bad_alloc& ) {
3163 errorText_ = "RtApiAsio::probeDeviceOpen: error allocating AsioHandle memory.";
3166 handle->bufferInfos = 0;
3168 // Create a manual-reset event.
3169 handle->condition = CreateEvent( NULL, // no security
3170 TRUE, // manual-reset
3171 FALSE, // non-signaled initially
3173 stream_.apiHandle = (void *) handle;
3176 // Create the ASIO internal buffers. Since RtAudio sets up input
3177 // and output separately, we'll have to dispose of previously
3178 // created output buffers for a duplex stream.
3179 if ( mode == INPUT && stream_.mode == OUTPUT ) {
3180 ASIODisposeBuffers();
3181 if ( handle->bufferInfos ) free( handle->bufferInfos );
3184 // Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure.
3186 nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
3187 handle->bufferInfos = (ASIOBufferInfo *) malloc( nChannels * sizeof(ASIOBufferInfo) );
3188 if ( handle->bufferInfos == NULL ) {
3189 errorStream_ << "RtApiAsio::probeDeviceOpen: error allocating bufferInfo memory for driver (" << driverName << ").";
3190 errorText_ = errorStream_.str();
3194 ASIOBufferInfo *infos;
3195 infos = handle->bufferInfos;
3196 for ( i=0; i<stream_.nDeviceChannels[0]; i++, infos++ ) {
3197 infos->isInput = ASIOFalse;
3198 infos->channelNum = i + stream_.channelOffset[0];
3199 infos->buffers[0] = infos->buffers[1] = 0;
3201 for ( i=0; i<stream_.nDeviceChannels[1]; i++, infos++ ) {
3202 infos->isInput = ASIOTrue;
3203 infos->channelNum = i + stream_.channelOffset[1];
3204 infos->buffers[0] = infos->buffers[1] = 0;
3207 // prepare for callbacks
3208 stream_.sampleRate = sampleRate;
3209 stream_.device[mode] = device;
3210 stream_.mode = isDuplexInput ? DUPLEX : mode;
3212 // store this class instance before registering callbacks, that are going to use it
3213 asioCallbackInfo = &stream_.callbackInfo;
3214 stream_.callbackInfo.object = (void *) this;
3216 // Set up the ASIO callback structure and create the ASIO data buffers.
3217 asioCallbacks.bufferSwitch = &bufferSwitch;
3218 asioCallbacks.sampleRateDidChange = &sampleRateChanged;
3219 asioCallbacks.asioMessage = &asioMessages;
3220 asioCallbacks.bufferSwitchTimeInfo = NULL;
3221 result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks );
3222 if ( result != ASE_OK ) {
3223 // Standard method failed. This can happen with strict/misbehaving drivers that return valid buffer size ranges
3224 // but only accept the preferred buffer size as parameter for ASIOCreateBuffers (e.g. Creative's ASIO driver).
3225 // In that case, let's be naïve and try that instead.
3226 *bufferSize = preferSize;
3227 stream_.bufferSize = *bufferSize;
3228 result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks );
3231 if ( result != ASE_OK ) {
3232 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") creating buffers.";
3233 errorText_ = errorStream_.str();
3236 buffersAllocated = true;
3237 stream_.state = STREAM_STOPPED;
3239 // Set flags for buffer conversion.
3240 stream_.doConvertBuffer[mode] = false;
3241 if ( stream_.userFormat != stream_.deviceFormat[mode] )
3242 stream_.doConvertBuffer[mode] = true;
3243 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
3244 stream_.nUserChannels[mode] > 1 )
3245 stream_.doConvertBuffer[mode] = true;
3247 // Allocate necessary internal buffers
3248 unsigned long bufferBytes;
3249 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
3250 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
3251 if ( stream_.userBuffer[mode] == NULL ) {
3252 errorText_ = "RtApiAsio::probeDeviceOpen: error allocating user buffer memory.";
3256 if ( stream_.doConvertBuffer[mode] ) {
3258 bool makeBuffer = true;
3259 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
3260 if ( isDuplexInput && stream_.deviceBuffer ) {
3261 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
3262 if ( bufferBytes <= bytesOut ) makeBuffer = false;
3266 bufferBytes *= *bufferSize;
3267 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
3268 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
3269 if ( stream_.deviceBuffer == NULL ) {
3270 errorText_ = "RtApiAsio::probeDeviceOpen: error allocating device buffer memory.";
3276 // Determine device latencies
3277 long inputLatency, outputLatency;
3278 result = ASIOGetLatencies( &inputLatency, &outputLatency );
3279 if ( result != ASE_OK ) {
3280 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting latency.";
3281 errorText_ = errorStream_.str();
3282 error( RtAudioError::WARNING); // warn but don't fail
3285 stream_.latency[0] = outputLatency;
3286 stream_.latency[1] = inputLatency;
3289 // Setup the buffer conversion information structure. We don't use
3290 // buffers to do channel offsets, so we override that parameter
3292 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );
3297 if ( !isDuplexInput ) {
3298 // the cleanup for error in the duplex input, is done by RtApi::openStream
3299 // So we clean up for single channel only
3301 if ( buffersAllocated )
3302 ASIODisposeBuffers();
3304 drivers.removeCurrentDriver();
3307 CloseHandle( handle->condition );
3308 if ( handle->bufferInfos )
3309 free( handle->bufferInfos );
3312 stream_.apiHandle = 0;
3316 if ( stream_.userBuffer[mode] ) {
3317 free( stream_.userBuffer[mode] );
3318 stream_.userBuffer[mode] = 0;
3321 if ( stream_.deviceBuffer ) {
3322 free( stream_.deviceBuffer );
3323 stream_.deviceBuffer = 0;
3328 }////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
3330 void RtApiAsio :: closeStream()
3332 if ( stream_.state == STREAM_CLOSED ) {
3333 errorText_ = "RtApiAsio::closeStream(): no open stream to close!";
3334 error( RtAudioError::WARNING );
3338 if ( stream_.state == STREAM_RUNNING ) {
3339 stream_.state = STREAM_STOPPED;
3342 ASIODisposeBuffers();
3343 drivers.removeCurrentDriver();
3345 AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
3347 CloseHandle( handle->condition );
3348 if ( handle->bufferInfos )
3349 free( handle->bufferInfos );
3351 stream_.apiHandle = 0;
3354 for ( int i=0; i<2; i++ ) {
3355 if ( stream_.userBuffer[i] ) {
3356 free( stream_.userBuffer[i] );
3357 stream_.userBuffer[i] = 0;
3361 if ( stream_.deviceBuffer ) {
3362 free( stream_.deviceBuffer );
3363 stream_.deviceBuffer = 0;
3366 stream_.mode = UNINITIALIZED;
3367 stream_.state = STREAM_CLOSED;
3370 bool stopThreadCalled = false;
3372 void RtApiAsio :: startStream()
3375 if ( stream_.state == STREAM_RUNNING ) {
3376 errorText_ = "RtApiAsio::startStream(): the stream is already running!";
3377 error( RtAudioError::WARNING );
3381 AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
3382 ASIOError result = ASIOStart();
3383 if ( result != ASE_OK ) {
3384 errorStream_ << "RtApiAsio::startStream: error (" << getAsioErrorString( result ) << ") starting device.";
3385 errorText_ = errorStream_.str();
3389 handle->drainCounter = 0;
3390 handle->internalDrain = false;
3391 ResetEvent( handle->condition );
3392 stream_.state = STREAM_RUNNING;
3396 stopThreadCalled = false;
3398 if ( result == ASE_OK ) return;
3399 error( RtAudioError::SYSTEM_ERROR );
3402 void RtApiAsio :: stopStream()
3405 if ( stream_.state == STREAM_STOPPED ) {
3406 errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!";
3407 error( RtAudioError::WARNING );
3411 AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
3412 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
3413 if ( handle->drainCounter == 0 ) {
3414 handle->drainCounter = 2;
3415 WaitForSingleObject( handle->condition, INFINITE ); // block until signaled
3419 stream_.state = STREAM_STOPPED;
3421 ASIOError result = ASIOStop();
3422 if ( result != ASE_OK ) {
3423 errorStream_ << "RtApiAsio::stopStream: error (" << getAsioErrorString( result ) << ") stopping device.";
3424 errorText_ = errorStream_.str();
3427 if ( result == ASE_OK ) return;
3428 error( RtAudioError::SYSTEM_ERROR );
3431 void RtApiAsio :: abortStream()
3434 if ( stream_.state == STREAM_STOPPED ) {
3435 errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!";
3436 error( RtAudioError::WARNING );
3440 // The following lines were commented-out because some behavior was
3441 // noted where the device buffers need to be zeroed to avoid
3442 // continuing sound, even when the device buffers are completely
3443 // disposed. So now, calling abort is the same as calling stop.
3444 // AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
3445 // handle->drainCounter = 2;
3449 // This function will be called by a spawned thread when the user
3450 // callback function signals that the stream should be stopped or
3451 // aborted. It is necessary to handle it this way because the
3452 // callbackEvent() function must return before the ASIOStop()
3453 // function will return.
3454 static unsigned __stdcall asioStopStream( void *ptr )
3456 CallbackInfo *info = (CallbackInfo *) ptr;
3457 RtApiAsio *object = (RtApiAsio *) info->object;
3459 object->stopStream();
3464 bool RtApiAsio :: callbackEvent( long bufferIndex )
3466 if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;
3467 if ( stream_.state == STREAM_CLOSED ) {
3468 errorText_ = "RtApiAsio::callbackEvent(): the stream is closed ... this shouldn't happen!";
3469 error( RtAudioError::WARNING );
3473 CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
3474 AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
3476 // Check if we were draining the stream and signal if finished.
3477 if ( handle->drainCounter > 3 ) {
3479 stream_.state = STREAM_STOPPING;
3480 if ( handle->internalDrain == false )
3481 SetEvent( handle->condition );
3482 else { // spawn a thread to stop the stream
3484 stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream,
3485 &stream_.callbackInfo, 0, &threadId );
3490 // Invoke user callback to get fresh output data UNLESS we are
3492 if ( handle->drainCounter == 0 ) {
3493 RtAudioCallback callback = (RtAudioCallback) info->callback;
3494 double streamTime = getStreamTime();
3495 RtAudioStreamStatus status = 0;
3496 if ( stream_.mode != INPUT && asioXRun == true ) {
3497 status |= RTAUDIO_OUTPUT_UNDERFLOW;
3500 if ( stream_.mode != OUTPUT && asioXRun == true ) {
3501 status |= RTAUDIO_INPUT_OVERFLOW;
3504 int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
3505 stream_.bufferSize, streamTime, status, info->userData );
3506 if ( cbReturnValue == 2 ) {
3507 stream_.state = STREAM_STOPPING;
3508 handle->drainCounter = 2;
3510 stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream,
3511 &stream_.callbackInfo, 0, &threadId );
3514 else if ( cbReturnValue == 1 ) {
3515 handle->drainCounter = 1;
3516 handle->internalDrain = true;
3520 unsigned int nChannels, bufferBytes, i, j;
3521 nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
3522 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
3524 bufferBytes = stream_.bufferSize * formatBytes( stream_.deviceFormat[0] );
3526 if ( handle->drainCounter > 1 ) { // write zeros to the output stream
3528 for ( i=0, j=0; i<nChannels; i++ ) {
3529 if ( handle->bufferInfos[i].isInput != ASIOTrue )
3530 memset( handle->bufferInfos[i].buffers[bufferIndex], 0, bufferBytes );
3534 else if ( stream_.doConvertBuffer[0] ) {
3536 convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
3537 if ( stream_.doByteSwap[0] )
3538 byteSwapBuffer( stream_.deviceBuffer,
3539 stream_.bufferSize * stream_.nDeviceChannels[0],
3540 stream_.deviceFormat[0] );
3542 for ( i=0, j=0; i<nChannels; i++ ) {
3543 if ( handle->bufferInfos[i].isInput != ASIOTrue )
3544 memcpy( handle->bufferInfos[i].buffers[bufferIndex],
3545 &stream_.deviceBuffer[j++*bufferBytes], bufferBytes );
3551 if ( stream_.doByteSwap[0] )
3552 byteSwapBuffer( stream_.userBuffer[0],
3553 stream_.bufferSize * stream_.nUserChannels[0],
3554 stream_.userFormat );
3556 for ( i=0, j=0; i<nChannels; i++ ) {
3557 if ( handle->bufferInfos[i].isInput != ASIOTrue )
3558 memcpy( handle->bufferInfos[i].buffers[bufferIndex],
3559 &stream_.userBuffer[0][bufferBytes*j++], bufferBytes );
3565 // Don't bother draining input
3566 if ( handle->drainCounter ) {
3567 handle->drainCounter++;
3571 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
3573 bufferBytes = stream_.bufferSize * formatBytes(stream_.deviceFormat[1]);
3575 if (stream_.doConvertBuffer[1]) {
3577 // Always interleave ASIO input data.
3578 for ( i=0, j=0; i<nChannels; i++ ) {
3579 if ( handle->bufferInfos[i].isInput == ASIOTrue )
3580 memcpy( &stream_.deviceBuffer[j++*bufferBytes],
3581 handle->bufferInfos[i].buffers[bufferIndex],
3585 if ( stream_.doByteSwap[1] )
3586 byteSwapBuffer( stream_.deviceBuffer,
3587 stream_.bufferSize * stream_.nDeviceChannels[1],
3588 stream_.deviceFormat[1] );
3589 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
3593 for ( i=0, j=0; i<nChannels; i++ ) {
3594 if ( handle->bufferInfos[i].isInput == ASIOTrue ) {
3595 memcpy( &stream_.userBuffer[1][bufferBytes*j++],
3596 handle->bufferInfos[i].buffers[bufferIndex],
3601 if ( stream_.doByteSwap[1] )
3602 byteSwapBuffer( stream_.userBuffer[1],
3603 stream_.bufferSize * stream_.nUserChannels[1],
3604 stream_.userFormat );
3609 // The following call was suggested by Malte Clasen. While the API
3610 // documentation indicates it should not be required, some device
3611 // drivers apparently do not function correctly without it.
3614 RtApi::tickStreamTime();
3618 static void sampleRateChanged( ASIOSampleRate sRate )
3620 // The ASIO documentation says that this usually only happens during
3621 // external sync. Audio processing is not stopped by the driver,
3622 // actual sample rate might not have even changed, maybe only the
3623 // sample rate status of an AES/EBU or S/PDIF digital input at the
3626 RtApi *object = (RtApi *) asioCallbackInfo->object;
3628 object->stopStream();
3630 catch ( RtAudioError &exception ) {
3631 std::cerr << "\nRtApiAsio: sampleRateChanged() error (" << exception.getMessage() << ")!\n" << std::endl;
3635 std::cerr << "\nRtApiAsio: driver reports sample rate changed to " << sRate << " ... stream stopped!!!\n" << std::endl;
3638 static long asioMessages( long selector, long value, void* /*message*/, double* /*opt*/ )
3642 switch( selector ) {
3643 case kAsioSelectorSupported:
3644 if ( value == kAsioResetRequest
3645 || value == kAsioEngineVersion
3646 || value == kAsioResyncRequest
3647 || value == kAsioLatenciesChanged
3648 // The following three were added for ASIO 2.0, you don't
3649 // necessarily have to support them.
3650 || value == kAsioSupportsTimeInfo
3651 || value == kAsioSupportsTimeCode
3652 || value == kAsioSupportsInputMonitor)
3655 case kAsioResetRequest:
3656 // Defer the task and perform the reset of the driver during the
3657 // next "safe" situation. You cannot reset the driver right now,
3658 // as this code is called from the driver. Reset the driver is
3659 // done by completely destruct is. I.e. ASIOStop(),
3660 // ASIODisposeBuffers(), Destruction Afterwards you initialize the
3662 std::cerr << "\nRtApiAsio: driver reset requested!!!" << std::endl;
3665 case kAsioResyncRequest:
3666 // This informs the application that the driver encountered some
3667 // non-fatal data loss. It is used for synchronization purposes
3668 // of different media. Added mainly to work around the Win16Mutex
3669 // problems in Windows 95/98 with the Windows Multimedia system,
3670 // which could lose data because the Mutex was held too long by
3671 // another thread. However a driver can issue it in other
3673 // std::cerr << "\nRtApiAsio: driver resync requested!!!" << std::endl;
3677 case kAsioLatenciesChanged:
3678 // This will inform the host application that the drivers were
3679 // latencies changed. Beware, it this does not mean that the
3680 // buffer sizes have changed! You might need to update internal
3682 std::cerr << "\nRtApiAsio: driver latency may have changed!!!" << std::endl;
3685 case kAsioEngineVersion:
3686 // Return the supported ASIO version of the host application. If
3687 // a host application does not implement this selector, ASIO 1.0
3688 // is assumed by the driver.
3691 case kAsioSupportsTimeInfo:
3692 // Informs the driver whether the
3693 // asioCallbacks.bufferSwitchTimeInfo() callback is supported.
3694 // For compatibility with ASIO 1.0 drivers the host application
3695 // should always support the "old" bufferSwitch method, too.
3698 case kAsioSupportsTimeCode:
3699 // Informs the driver whether application is interested in time
3700 // code info. If an application does not need to know about time
3701 // code, the driver has less work to do.
3708 static const char* getAsioErrorString( ASIOError result )
3716 static const Messages m[] =
3718 { ASE_NotPresent, "Hardware input or output is not present or available." },
3719 { ASE_HWMalfunction, "Hardware is malfunctioning." },
3720 { ASE_InvalidParameter, "Invalid input parameter." },
3721 { ASE_InvalidMode, "Invalid mode." },
3722 { ASE_SPNotAdvancing, "Sample position not advancing." },
3723 { ASE_NoClock, "Sample clock or rate cannot be determined or is not present." },
3724 { ASE_NoMemory, "Not enough memory to complete the request." }
3727 for ( unsigned int i = 0; i < sizeof(m)/sizeof(m[0]); ++i )
3728 if ( m[i].value == result ) return m[i].message;
3730 return "Unknown error.";
3733 //******************** End of __WINDOWS_ASIO__ *********************//
3737 #if defined(__WINDOWS_WASAPI__) // Windows WASAPI API
3739 // Authored by Marcus Tomlinson <themarcustomlinson@gmail.com>, April 2014
3740 // - Introduces support for the Windows WASAPI API
3741 // - Aims to deliver bit streams to and from hardware at the lowest possible latency, via the absolute minimum buffer sizes required
3742 // - Provides flexible stream configuration to an otherwise strict and inflexible WASAPI interface
3743 // - Includes automatic internal conversion of sample rate and buffer size between hardware and the user
3750 #include <mferror.h>
3752 #include <mftransform.h>
3753 #include <wmcodecdsp.h>
3755 #include <audioclient.h>
3757 #include <mmdeviceapi.h>
3758 #include <functiondiscoverykeys_devpkey.h>
3760 #ifndef MF_E_TRANSFORM_NEED_MORE_INPUT
3761 #define MF_E_TRANSFORM_NEED_MORE_INPUT _HRESULT_TYPEDEF_(0xc00d6d72)
3764 #ifndef MFSTARTUP_NOSOCKET
3765 #define MFSTARTUP_NOSOCKET 0x1
3769 #pragma comment( lib, "ksuser" )
3770 #pragma comment( lib, "mfplat.lib" )
3771 #pragma comment( lib, "mfuuid.lib" )
3772 #pragma comment( lib, "wmcodecdspuuid" )
3775 //=============================================================================
3777 #define SAFE_RELEASE( objectPtr )\
3780 objectPtr->Release();\
3784 typedef HANDLE ( __stdcall *TAvSetMmThreadCharacteristicsPtr )( LPCWSTR TaskName, LPDWORD TaskIndex );
3786 //-----------------------------------------------------------------------------
3788 // WASAPI dictates stream sample rate, format, channel count, and in some cases, buffer size.
3789 // Therefore we must perform all necessary conversions to user buffers in order to satisfy these
3790 // requirements. WasapiBuffer ring buffers are used between HwIn->UserIn and UserOut->HwOut to
3791 // provide intermediate storage for read / write synchronization.
3805 // sets the length of the internal ring buffer
3806 void setBufferSize( unsigned int bufferSize, unsigned int formatBytes ) {
3809 buffer_ = ( char* ) calloc( bufferSize, formatBytes );
3811 bufferSize_ = bufferSize;
3816 // attempt to push a buffer into the ring buffer at the current "in" index
3817 bool pushBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format )
3819 if ( !buffer || // incoming buffer is NULL
3820 bufferSize == 0 || // incoming buffer has no data
3821 bufferSize > bufferSize_ ) // incoming buffer too large
3826 unsigned int relOutIndex = outIndex_;
3827 unsigned int inIndexEnd = inIndex_ + bufferSize;
3828 if ( relOutIndex < inIndex_ && inIndexEnd >= bufferSize_ ) {
3829 relOutIndex += bufferSize_;
3832 // "in" index can end on the "out" index but cannot begin at it
3833 if ( inIndex_ <= relOutIndex && inIndexEnd > relOutIndex ) {
3834 return false; // not enough space between "in" index and "out" index
3837 // copy buffer from external to internal
3838 int fromZeroSize = inIndex_ + bufferSize - bufferSize_;
3839 fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize;
3840 int fromInSize = bufferSize - fromZeroSize;
3845 memcpy( &( ( char* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( char ) );
3846 memcpy( buffer_, &( ( char* ) buffer )[fromInSize], fromZeroSize * sizeof( char ) );
3848 case RTAUDIO_SINT16:
3849 memcpy( &( ( short* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( short ) );
3850 memcpy( buffer_, &( ( short* ) buffer )[fromInSize], fromZeroSize * sizeof( short ) );
3852 case RTAUDIO_SINT24:
3853 memcpy( &( ( S24* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( S24 ) );
3854 memcpy( buffer_, &( ( S24* ) buffer )[fromInSize], fromZeroSize * sizeof( S24 ) );
3856 case RTAUDIO_SINT32:
3857 memcpy( &( ( int* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( int ) );
3858 memcpy( buffer_, &( ( int* ) buffer )[fromInSize], fromZeroSize * sizeof( int ) );
3860 case RTAUDIO_FLOAT32:
3861 memcpy( &( ( float* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( float ) );
3862 memcpy( buffer_, &( ( float* ) buffer )[fromInSize], fromZeroSize * sizeof( float ) );
3864 case RTAUDIO_FLOAT64:
3865 memcpy( &( ( double* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( double ) );
3866 memcpy( buffer_, &( ( double* ) buffer )[fromInSize], fromZeroSize * sizeof( double ) );
3870 // update "in" index
3871 inIndex_ += bufferSize;
3872 inIndex_ %= bufferSize_;
3877 // attempt to pull a buffer from the ring buffer from the current "out" index
3878 bool pullBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format )
3880 if ( !buffer || // incoming buffer is NULL
3881 bufferSize == 0 || // incoming buffer has no data
3882 bufferSize > bufferSize_ ) // incoming buffer too large
3887 unsigned int relInIndex = inIndex_;
3888 unsigned int outIndexEnd = outIndex_ + bufferSize;
3889 if ( relInIndex < outIndex_ && outIndexEnd >= bufferSize_ ) {
3890 relInIndex += bufferSize_;
3893 // "out" index can begin at and end on the "in" index
3894 if ( outIndex_ < relInIndex && outIndexEnd > relInIndex ) {
3895 return false; // not enough space between "out" index and "in" index
3898 // copy buffer from internal to external
3899 int fromZeroSize = outIndex_ + bufferSize - bufferSize_;
3900 fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize;
3901 int fromOutSize = bufferSize - fromZeroSize;
3906 memcpy( buffer, &( ( char* ) buffer_ )[outIndex_], fromOutSize * sizeof( char ) );
3907 memcpy( &( ( char* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( char ) );
3909 case RTAUDIO_SINT16:
3910 memcpy( buffer, &( ( short* ) buffer_ )[outIndex_], fromOutSize * sizeof( short ) );
3911 memcpy( &( ( short* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( short ) );
3913 case RTAUDIO_SINT24:
3914 memcpy( buffer, &( ( S24* ) buffer_ )[outIndex_], fromOutSize * sizeof( S24 ) );
3915 memcpy( &( ( S24* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( S24 ) );
3917 case RTAUDIO_SINT32:
3918 memcpy( buffer, &( ( int* ) buffer_ )[outIndex_], fromOutSize * sizeof( int ) );
3919 memcpy( &( ( int* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( int ) );
3921 case RTAUDIO_FLOAT32:
3922 memcpy( buffer, &( ( float* ) buffer_ )[outIndex_], fromOutSize * sizeof( float ) );
3923 memcpy( &( ( float* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( float ) );
3925 case RTAUDIO_FLOAT64:
3926 memcpy( buffer, &( ( double* ) buffer_ )[outIndex_], fromOutSize * sizeof( double ) );
3927 memcpy( &( ( double* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( double ) );
3931 // update "out" index
3932 outIndex_ += bufferSize;
3933 outIndex_ %= bufferSize_;
3940 unsigned int bufferSize_;
3941 unsigned int inIndex_;
3942 unsigned int outIndex_;
3945 //-----------------------------------------------------------------------------
3947 // In order to satisfy WASAPI's buffer requirements, we need a means of converting sample rate
3948 // between HW and the user. The WasapiResampler class is used to perform this conversion between
3949 // HwIn->UserIn and UserOut->HwOut during the stream callback loop.
3950 class WasapiResampler
3953 WasapiResampler( bool isFloat, unsigned int bitsPerSample, unsigned int channelCount,
3954 unsigned int inSampleRate, unsigned int outSampleRate )
3955 : _bytesPerSample( bitsPerSample / 8 )
3956 , _channelCount( channelCount )
3957 , _sampleRatio( ( float ) outSampleRate / inSampleRate )
3958 , _transformUnk( NULL )
3959 , _transform( NULL )
3960 , _mediaType( NULL )
3961 , _inputMediaType( NULL )
3962 , _outputMediaType( NULL )
3964 #ifdef __IWMResamplerProps_FWD_DEFINED__
3965 , _resamplerProps( NULL )
3968 // 1. Initialization
3970 MFStartup( MF_VERSION, MFSTARTUP_NOSOCKET );
3972 // 2. Create Resampler Transform Object
3974 CoCreateInstance( CLSID_CResamplerMediaObject, NULL, CLSCTX_INPROC_SERVER,
3975 IID_IUnknown, ( void** ) &_transformUnk );
3977 _transformUnk->QueryInterface( IID_PPV_ARGS( &_transform ) );
3979 #ifdef __IWMResamplerProps_FWD_DEFINED__
3980 _transformUnk->QueryInterface( IID_PPV_ARGS( &_resamplerProps ) );
3981 _resamplerProps->SetHalfFilterLength( 60 ); // best conversion quality
3984 // 3. Specify input / output format
3986 MFCreateMediaType( &_mediaType );
3987 _mediaType->SetGUID( MF_MT_MAJOR_TYPE, MFMediaType_Audio );
3988 _mediaType->SetGUID( MF_MT_SUBTYPE, isFloat ? MFAudioFormat_Float : MFAudioFormat_PCM );
3989 _mediaType->SetUINT32( MF_MT_AUDIO_NUM_CHANNELS, channelCount );
3990 _mediaType->SetUINT32( MF_MT_AUDIO_SAMPLES_PER_SECOND, inSampleRate );
3991 _mediaType->SetUINT32( MF_MT_AUDIO_BLOCK_ALIGNMENT, _bytesPerSample * channelCount );
3992 _mediaType->SetUINT32( MF_MT_AUDIO_AVG_BYTES_PER_SECOND, _bytesPerSample * channelCount * inSampleRate );
3993 _mediaType->SetUINT32( MF_MT_AUDIO_BITS_PER_SAMPLE, bitsPerSample );
3994 _mediaType->SetUINT32( MF_MT_ALL_SAMPLES_INDEPENDENT, TRUE );
3996 MFCreateMediaType( &_inputMediaType );
3997 _mediaType->CopyAllItems( _inputMediaType );
3999 _transform->SetInputType( 0, _inputMediaType, 0 );
4001 MFCreateMediaType( &_outputMediaType );
4002 _mediaType->CopyAllItems( _outputMediaType );
4004 _outputMediaType->SetUINT32( MF_MT_AUDIO_SAMPLES_PER_SECOND, outSampleRate );
4005 _outputMediaType->SetUINT32( MF_MT_AUDIO_AVG_BYTES_PER_SECOND, _bytesPerSample * channelCount * outSampleRate );
4007 _transform->SetOutputType( 0, _outputMediaType, 0 );
4009 // 4. Send stream start messages to Resampler
4011 _transform->ProcessMessage( MFT_MESSAGE_COMMAND_FLUSH, 0 );
4012 _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_BEGIN_STREAMING, 0 );
4013 _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_START_OF_STREAM, 0 );
4018 // 8. Send stream stop messages to Resampler
4020 _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_END_OF_STREAM, 0 );
4021 _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_END_STREAMING, 0 );
4027 SAFE_RELEASE( _transformUnk );
4028 SAFE_RELEASE( _transform );
4029 SAFE_RELEASE( _mediaType );
4030 SAFE_RELEASE( _inputMediaType );
4031 SAFE_RELEASE( _outputMediaType );
4033 #ifdef __IWMResamplerProps_FWD_DEFINED__
4034 SAFE_RELEASE( _resamplerProps );
4038 void Convert( char* outBuffer, const char* inBuffer, unsigned int inSampleCount, unsigned int& outSampleCount )
4040 unsigned int inputBufferSize = _bytesPerSample * _channelCount * inSampleCount;
4041 if ( _sampleRatio == 1 )
4043 // no sample rate conversion required
4044 memcpy( outBuffer, inBuffer, inputBufferSize );
4045 outSampleCount = inSampleCount;
4049 unsigned int outputBufferSize = ( unsigned int ) ceilf( inputBufferSize * _sampleRatio ) + ( _bytesPerSample * _channelCount );
4051 IMFMediaBuffer* rInBuffer;
4052 IMFSample* rInSample;
4053 BYTE* rInByteBuffer = NULL;
4055 // 5. Create Sample object from input data
4057 MFCreateMemoryBuffer( inputBufferSize, &rInBuffer );
4059 rInBuffer->Lock( &rInByteBuffer, NULL, NULL );
4060 memcpy( rInByteBuffer, inBuffer, inputBufferSize );
4061 rInBuffer->Unlock();
4062 rInByteBuffer = NULL;
4064 rInBuffer->SetCurrentLength( inputBufferSize );
4066 MFCreateSample( &rInSample );
4067 rInSample->AddBuffer( rInBuffer );
4069 // 6. Pass input data to Resampler
4071 _transform->ProcessInput( 0, rInSample, 0 );
4073 SAFE_RELEASE( rInBuffer );
4074 SAFE_RELEASE( rInSample );
4076 // 7. Perform sample rate conversion
4078 IMFMediaBuffer* rOutBuffer = NULL;
4079 BYTE* rOutByteBuffer = NULL;
4081 MFT_OUTPUT_DATA_BUFFER rOutDataBuffer;
4083 DWORD rBytes = outputBufferSize; // maximum bytes accepted per ProcessOutput
4085 // 7.1 Create Sample object for output data
4087 memset( &rOutDataBuffer, 0, sizeof rOutDataBuffer );
4088 MFCreateSample( &( rOutDataBuffer.pSample ) );
4089 MFCreateMemoryBuffer( rBytes, &rOutBuffer );
4090 rOutDataBuffer.pSample->AddBuffer( rOutBuffer );
4091 rOutDataBuffer.dwStreamID = 0;
4092 rOutDataBuffer.dwStatus = 0;
4093 rOutDataBuffer.pEvents = NULL;
4095 // 7.2 Get output data from Resampler
4097 if ( _transform->ProcessOutput( 0, 1, &rOutDataBuffer, &rStatus ) == MF_E_TRANSFORM_NEED_MORE_INPUT )
4100 SAFE_RELEASE( rOutBuffer );
4101 SAFE_RELEASE( rOutDataBuffer.pSample );
4105 // 7.3 Write output data to outBuffer
4107 SAFE_RELEASE( rOutBuffer );
4108 rOutDataBuffer.pSample->ConvertToContiguousBuffer( &rOutBuffer );
4109 rOutBuffer->GetCurrentLength( &rBytes );
4111 rOutBuffer->Lock( &rOutByteBuffer, NULL, NULL );
4112 memcpy( outBuffer, rOutByteBuffer, rBytes );
4113 rOutBuffer->Unlock();
4114 rOutByteBuffer = NULL;
4116 outSampleCount = rBytes / _bytesPerSample / _channelCount;
4117 SAFE_RELEASE( rOutBuffer );
4118 SAFE_RELEASE( rOutDataBuffer.pSample );
4122 unsigned int _bytesPerSample;
4123 unsigned int _channelCount;
4126 IUnknown* _transformUnk;
4127 IMFTransform* _transform;
4128 IMFMediaType* _mediaType;
4129 IMFMediaType* _inputMediaType;
4130 IMFMediaType* _outputMediaType;
4132 #ifdef __IWMResamplerProps_FWD_DEFINED__
4133 IWMResamplerProps* _resamplerProps;
4137 //-----------------------------------------------------------------------------
4139 // A structure to hold various information related to the WASAPI implementation.
4142 IAudioClient* captureAudioClient;
4143 IAudioClient* renderAudioClient;
4144 IAudioCaptureClient* captureClient;
4145 IAudioRenderClient* renderClient;
4146 HANDLE captureEvent;
4150 : captureAudioClient( NULL ),
4151 renderAudioClient( NULL ),
4152 captureClient( NULL ),
4153 renderClient( NULL ),
4154 captureEvent( NULL ),
4155 renderEvent( NULL ) {}
4158 //=============================================================================
4160 RtApiWasapi::RtApiWasapi()
4161 : coInitialized_( false ), deviceEnumerator_( NULL )
4163 // WASAPI can run either apartment or multi-threaded
4164 HRESULT hr = CoInitialize( NULL );
4165 if ( !FAILED( hr ) )
4166 coInitialized_ = true;
4168 // Instantiate device enumerator
4169 hr = CoCreateInstance( __uuidof( MMDeviceEnumerator ), NULL,
4170 CLSCTX_ALL, __uuidof( IMMDeviceEnumerator ),
4171 ( void** ) &deviceEnumerator_ );
4173 // If this runs on an old Windows, it will fail. Ignore and proceed.
4175 deviceEnumerator_ = NULL;
4178 //-----------------------------------------------------------------------------
4180 RtApiWasapi::~RtApiWasapi()
4182 if ( stream_.state != STREAM_CLOSED )
4185 SAFE_RELEASE( deviceEnumerator_ );
4187 // If this object previously called CoInitialize()
4188 if ( coInitialized_ )
4192 //=============================================================================
4194 unsigned int RtApiWasapi::getDeviceCount( void )
4196 unsigned int captureDeviceCount = 0;
4197 unsigned int renderDeviceCount = 0;
4199 IMMDeviceCollection* captureDevices = NULL;
4200 IMMDeviceCollection* renderDevices = NULL;
4202 if ( !deviceEnumerator_ )
4205 // Count capture devices
4207 HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
4208 if ( FAILED( hr ) ) {
4209 errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device collection.";
4213 hr = captureDevices->GetCount( &captureDeviceCount );
4214 if ( FAILED( hr ) ) {
4215 errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device count.";
4219 // Count render devices
4220 hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
4221 if ( FAILED( hr ) ) {
4222 errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device collection.";
4226 hr = renderDevices->GetCount( &renderDeviceCount );
4227 if ( FAILED( hr ) ) {
4228 errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device count.";
4233 // release all references
4234 SAFE_RELEASE( captureDevices );
4235 SAFE_RELEASE( renderDevices );
4237 if ( errorText_.empty() )
4238 return captureDeviceCount + renderDeviceCount;
4240 error( RtAudioError::DRIVER_ERROR );
4244 //-----------------------------------------------------------------------------
4246 RtAudio::DeviceInfo RtApiWasapi::getDeviceInfo( unsigned int device )
4248 RtAudio::DeviceInfo info;
4249 unsigned int captureDeviceCount = 0;
4250 unsigned int renderDeviceCount = 0;
4251 std::string defaultDeviceName;
4252 bool isCaptureDevice = false;
4254 PROPVARIANT deviceNameProp;
4255 PROPVARIANT defaultDeviceNameProp;
4257 IMMDeviceCollection* captureDevices = NULL;
4258 IMMDeviceCollection* renderDevices = NULL;
4259 IMMDevice* devicePtr = NULL;
4260 IMMDevice* defaultDevicePtr = NULL;
4261 IAudioClient* audioClient = NULL;
4262 IPropertyStore* devicePropStore = NULL;
4263 IPropertyStore* defaultDevicePropStore = NULL;
4265 WAVEFORMATEX* deviceFormat = NULL;
4266 WAVEFORMATEX* closestMatchFormat = NULL;
4269 info.probed = false;
4271 // Count capture devices
4273 RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
4274 HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
4275 if ( FAILED( hr ) ) {
4276 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device collection.";
4280 hr = captureDevices->GetCount( &captureDeviceCount );
4281 if ( FAILED( hr ) ) {
4282 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device count.";
4286 // Count render devices
4287 hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
4288 if ( FAILED( hr ) ) {
4289 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device collection.";
4293 hr = renderDevices->GetCount( &renderDeviceCount );
4294 if ( FAILED( hr ) ) {
4295 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device count.";
4299 // validate device index
4300 if ( device >= captureDeviceCount + renderDeviceCount ) {
4301 errorText_ = "RtApiWasapi::getDeviceInfo: Invalid device index.";
4302 errorType = RtAudioError::INVALID_USE;
4306 // determine whether index falls within capture or render devices
4307 if ( device >= renderDeviceCount ) {
4308 hr = captureDevices->Item( device - renderDeviceCount, &devicePtr );
4309 if ( FAILED( hr ) ) {
4310 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device handle.";
4313 isCaptureDevice = true;
4316 hr = renderDevices->Item( device, &devicePtr );
4317 if ( FAILED( hr ) ) {
4318 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device handle.";
4321 isCaptureDevice = false;
4324 // get default device name
4325 if ( isCaptureDevice ) {
4326 hr = deviceEnumerator_->GetDefaultAudioEndpoint( eCapture, eConsole, &defaultDevicePtr );
4327 if ( FAILED( hr ) ) {
4328 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default capture device handle.";
4333 hr = deviceEnumerator_->GetDefaultAudioEndpoint( eRender, eConsole, &defaultDevicePtr );
4334 if ( FAILED( hr ) ) {
4335 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default render device handle.";
4340 hr = defaultDevicePtr->OpenPropertyStore( STGM_READ, &defaultDevicePropStore );
4341 if ( FAILED( hr ) ) {
4342 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open default device property store.";
4345 PropVariantInit( &defaultDeviceNameProp );
4347 hr = defaultDevicePropStore->GetValue( PKEY_Device_FriendlyName, &defaultDeviceNameProp );
4348 if ( FAILED( hr ) ) {
4349 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default device property: PKEY_Device_FriendlyName.";
4353 defaultDeviceName = convertCharPointerToStdString(defaultDeviceNameProp.pwszVal);
4356 hr = devicePtr->OpenPropertyStore( STGM_READ, &devicePropStore );
4357 if ( FAILED( hr ) ) {
4358 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open device property store.";
4362 PropVariantInit( &deviceNameProp );
4364 hr = devicePropStore->GetValue( PKEY_Device_FriendlyName, &deviceNameProp );
4365 if ( FAILED( hr ) ) {
4366 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device property: PKEY_Device_FriendlyName.";
4370 info.name =convertCharPointerToStdString(deviceNameProp.pwszVal);
4373 if ( isCaptureDevice ) {
4374 info.isDefaultInput = info.name == defaultDeviceName;
4375 info.isDefaultOutput = false;
4378 info.isDefaultInput = false;
4379 info.isDefaultOutput = info.name == defaultDeviceName;
4383 hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, NULL, ( void** ) &audioClient );
4384 if ( FAILED( hr ) ) {
4385 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device audio client.";
4389 hr = audioClient->GetMixFormat( &deviceFormat );
4390 if ( FAILED( hr ) ) {
4391 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device mix format.";
4395 if ( isCaptureDevice ) {
4396 info.inputChannels = deviceFormat->nChannels;
4397 info.outputChannels = 0;
4398 info.duplexChannels = 0;
4401 info.inputChannels = 0;
4402 info.outputChannels = deviceFormat->nChannels;
4403 info.duplexChannels = 0;
4407 info.sampleRates.clear();
4409 // allow support for all sample rates as we have a built-in sample rate converter
4410 for ( unsigned int i = 0; i < MAX_SAMPLE_RATES; i++ ) {
4411 info.sampleRates.push_back( SAMPLE_RATES[i] );
4413 info.preferredSampleRate = deviceFormat->nSamplesPerSec;
4416 info.nativeFormats = 0;
4418 if ( deviceFormat->wFormatTag == WAVE_FORMAT_IEEE_FLOAT ||
4419 ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
4420 ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_IEEE_FLOAT ) )
4422 if ( deviceFormat->wBitsPerSample == 32 ) {
4423 info.nativeFormats |= RTAUDIO_FLOAT32;
4425 else if ( deviceFormat->wBitsPerSample == 64 ) {
4426 info.nativeFormats |= RTAUDIO_FLOAT64;
4429 else if ( deviceFormat->wFormatTag == WAVE_FORMAT_PCM ||
4430 ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
4431 ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_PCM ) )
4433 if ( deviceFormat->wBitsPerSample == 8 ) {
4434 info.nativeFormats |= RTAUDIO_SINT8;
4436 else if ( deviceFormat->wBitsPerSample == 16 ) {
4437 info.nativeFormats |= RTAUDIO_SINT16;
4439 else if ( deviceFormat->wBitsPerSample == 24 ) {
4440 info.nativeFormats |= RTAUDIO_SINT24;
4442 else if ( deviceFormat->wBitsPerSample == 32 ) {
4443 info.nativeFormats |= RTAUDIO_SINT32;
4451 // release all references
4452 PropVariantClear( &deviceNameProp );
4453 PropVariantClear( &defaultDeviceNameProp );
4455 SAFE_RELEASE( captureDevices );
4456 SAFE_RELEASE( renderDevices );
4457 SAFE_RELEASE( devicePtr );
4458 SAFE_RELEASE( defaultDevicePtr );
4459 SAFE_RELEASE( audioClient );
4460 SAFE_RELEASE( devicePropStore );
4461 SAFE_RELEASE( defaultDevicePropStore );
4463 CoTaskMemFree( deviceFormat );
4464 CoTaskMemFree( closestMatchFormat );
4466 if ( !errorText_.empty() )
4471 //-----------------------------------------------------------------------------
4473 unsigned int RtApiWasapi::getDefaultOutputDevice( void )
4475 for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {
4476 if ( getDeviceInfo( i ).isDefaultOutput ) {
4484 //-----------------------------------------------------------------------------
4486 unsigned int RtApiWasapi::getDefaultInputDevice( void )
4488 for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {
4489 if ( getDeviceInfo( i ).isDefaultInput ) {
4497 //-----------------------------------------------------------------------------
4499 void RtApiWasapi::closeStream( void )
4501 if ( stream_.state == STREAM_CLOSED ) {
4502 errorText_ = "RtApiWasapi::closeStream: No open stream to close.";
4503 error( RtAudioError::WARNING );
4507 if ( stream_.state != STREAM_STOPPED )
4510 // clean up stream memory
4511 SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient )
4512 SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient )
4514 SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureClient )
4515 SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderClient )
4517 if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent )
4518 CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent );
4520 if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent )
4521 CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent );
4523 delete ( WasapiHandle* ) stream_.apiHandle;
4524 stream_.apiHandle = NULL;
4526 for ( int i = 0; i < 2; i++ ) {
4527 if ( stream_.userBuffer[i] ) {
4528 free( stream_.userBuffer[i] );
4529 stream_.userBuffer[i] = 0;
4533 if ( stream_.deviceBuffer ) {
4534 free( stream_.deviceBuffer );
4535 stream_.deviceBuffer = 0;
4538 // update stream state
4539 stream_.state = STREAM_CLOSED;
4542 //-----------------------------------------------------------------------------
4544 void RtApiWasapi::startStream( void )
4548 if ( stream_.state == STREAM_RUNNING ) {
4549 errorText_ = "RtApiWasapi::startStream: The stream is already running.";
4550 error( RtAudioError::WARNING );
4554 // update stream state
4555 stream_.state = STREAM_RUNNING;
4557 // create WASAPI stream thread
4558 stream_.callbackInfo.thread = ( ThreadHandle ) CreateThread( NULL, 0, runWasapiThread, this, CREATE_SUSPENDED, NULL );
4560 if ( !stream_.callbackInfo.thread ) {
4561 errorText_ = "RtApiWasapi::startStream: Unable to instantiate callback thread.";
4562 error( RtAudioError::THREAD_ERROR );
4565 SetThreadPriority( ( void* ) stream_.callbackInfo.thread, stream_.callbackInfo.priority );
4566 ResumeThread( ( void* ) stream_.callbackInfo.thread );
4570 //-----------------------------------------------------------------------------
4572 void RtApiWasapi::stopStream( void )
4576 if ( stream_.state == STREAM_STOPPED ) {
4577 errorText_ = "RtApiWasapi::stopStream: The stream is already stopped.";
4578 error( RtAudioError::WARNING );
4582 // inform stream thread by setting stream state to STREAM_STOPPING
4583 stream_.state = STREAM_STOPPING;
4585 // wait until stream thread is stopped
4586 while( stream_.state != STREAM_STOPPED ) {
4590 // Wait for the last buffer to play before stopping.
4591 Sleep( 1000 * stream_.bufferSize / stream_.sampleRate );
4593 // stop capture client if applicable
4594 if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) {
4595 HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop();
4596 if ( FAILED( hr ) ) {
4597 errorText_ = "RtApiWasapi::stopStream: Unable to stop capture stream.";
4598 error( RtAudioError::DRIVER_ERROR );
4603 // stop render client if applicable
4604 if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) {
4605 HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop();
4606 if ( FAILED( hr ) ) {
4607 errorText_ = "RtApiWasapi::stopStream: Unable to stop render stream.";
4608 error( RtAudioError::DRIVER_ERROR );
4613 // close thread handle
4614 if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
4615 errorText_ = "RtApiWasapi::stopStream: Unable to close callback thread.";
4616 error( RtAudioError::THREAD_ERROR );
4620 stream_.callbackInfo.thread = (ThreadHandle) NULL;
4623 //-----------------------------------------------------------------------------
4625 void RtApiWasapi::abortStream( void )
4629 if ( stream_.state == STREAM_STOPPED ) {
4630 errorText_ = "RtApiWasapi::abortStream: The stream is already stopped.";
4631 error( RtAudioError::WARNING );
4635 // inform stream thread by setting stream state to STREAM_STOPPING
4636 stream_.state = STREAM_STOPPING;
4638 // wait until stream thread is stopped
4639 while ( stream_.state != STREAM_STOPPED ) {
4643 // stop capture client if applicable
4644 if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) {
4645 HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop();
4646 if ( FAILED( hr ) ) {
4647 errorText_ = "RtApiWasapi::abortStream: Unable to stop capture stream.";
4648 error( RtAudioError::DRIVER_ERROR );
4653 // stop render client if applicable
4654 if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) {
4655 HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop();
4656 if ( FAILED( hr ) ) {
4657 errorText_ = "RtApiWasapi::abortStream: Unable to stop render stream.";
4658 error( RtAudioError::DRIVER_ERROR );
4663 // close thread handle
4664 if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
4665 errorText_ = "RtApiWasapi::abortStream: Unable to close callback thread.";
4666 error( RtAudioError::THREAD_ERROR );
4670 stream_.callbackInfo.thread = (ThreadHandle) NULL;
4673 //-----------------------------------------------------------------------------
4675 bool RtApiWasapi::probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
4676 unsigned int firstChannel, unsigned int sampleRate,
4677 RtAudioFormat format, unsigned int* bufferSize,
4678 RtAudio::StreamOptions* options )
4680 bool methodResult = FAILURE;
4681 unsigned int captureDeviceCount = 0;
4682 unsigned int renderDeviceCount = 0;
4684 IMMDeviceCollection* captureDevices = NULL;
4685 IMMDeviceCollection* renderDevices = NULL;
4686 IMMDevice* devicePtr = NULL;
4687 WAVEFORMATEX* deviceFormat = NULL;
4688 unsigned int bufferBytes;
4689 stream_.state = STREAM_STOPPED;
4691 // create API Handle if not already created
4692 if ( !stream_.apiHandle )
4693 stream_.apiHandle = ( void* ) new WasapiHandle();
4695 // Count capture devices
4697 RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
4698 HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
4699 if ( FAILED( hr ) ) {
4700 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device collection.";
4704 hr = captureDevices->GetCount( &captureDeviceCount );
4705 if ( FAILED( hr ) ) {
4706 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device count.";
4710 // Count render devices
4711 hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
4712 if ( FAILED( hr ) ) {
4713 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device collection.";
4717 hr = renderDevices->GetCount( &renderDeviceCount );
4718 if ( FAILED( hr ) ) {
4719 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device count.";
4723 // validate device index
4724 if ( device >= captureDeviceCount + renderDeviceCount ) {
4725 errorType = RtAudioError::INVALID_USE;
4726 errorText_ = "RtApiWasapi::probeDeviceOpen: Invalid device index.";
4730 // if device index falls within capture devices
4731 if ( device >= renderDeviceCount ) {
4732 if ( mode != INPUT ) {
4733 errorType = RtAudioError::INVALID_USE;
4734 errorText_ = "RtApiWasapi::probeDeviceOpen: Capture device selected as output device.";
4738 // retrieve captureAudioClient from devicePtr
4739 IAudioClient*& captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
4741 hr = captureDevices->Item( device - renderDeviceCount, &devicePtr );
4742 if ( FAILED( hr ) ) {
4743 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device handle.";
4747 hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
4748 NULL, ( void** ) &captureAudioClient );
4749 if ( FAILED( hr ) ) {
4750 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device audio client.";
4754 hr = captureAudioClient->GetMixFormat( &deviceFormat );
4755 if ( FAILED( hr ) ) {
4756 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device mix format.";
4760 stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
4761 captureAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
4764 // if device index falls within render devices and is configured for loopback
4765 if ( device < renderDeviceCount && mode == INPUT )
4767 // if renderAudioClient is not initialised, initialise it now
4768 IAudioClient*& renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
4769 if ( !renderAudioClient )
4771 probeDeviceOpen( device, OUTPUT, channels, firstChannel, sampleRate, format, bufferSize, options );
4774 // retrieve captureAudioClient from devicePtr
4775 IAudioClient*& captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
4777 hr = renderDevices->Item( device, &devicePtr );
4778 if ( FAILED( hr ) ) {
4779 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device handle.";
4783 hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
4784 NULL, ( void** ) &captureAudioClient );
4785 if ( FAILED( hr ) ) {
4786 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device audio client.";
4790 hr = captureAudioClient->GetMixFormat( &deviceFormat );
4791 if ( FAILED( hr ) ) {
4792 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device mix format.";
4796 stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
4797 captureAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
4800 // if device index falls within render devices and is configured for output
4801 if ( device < renderDeviceCount && mode == OUTPUT )
4803 // if renderAudioClient is already initialised, don't initialise it again
4804 IAudioClient*& renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
4805 if ( renderAudioClient )
4807 methodResult = SUCCESS;
4811 hr = renderDevices->Item( device, &devicePtr );
4812 if ( FAILED( hr ) ) {
4813 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device handle.";
4817 hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
4818 NULL, ( void** ) &renderAudioClient );
4819 if ( FAILED( hr ) ) {
4820 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device audio client.";
4824 hr = renderAudioClient->GetMixFormat( &deviceFormat );
4825 if ( FAILED( hr ) ) {
4826 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device mix format.";
4830 stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
4831 renderAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
4835 if ( ( stream_.mode == OUTPUT && mode == INPUT ) ||
4836 ( stream_.mode == INPUT && mode == OUTPUT ) ) {
4837 stream_.mode = DUPLEX;
4840 stream_.mode = mode;
4843 stream_.device[mode] = device;
4844 stream_.doByteSwap[mode] = false;
4845 stream_.sampleRate = sampleRate;
4846 stream_.bufferSize = *bufferSize;
4847 stream_.nBuffers = 1;
4848 stream_.nUserChannels[mode] = channels;
4849 stream_.channelOffset[mode] = firstChannel;
4850 stream_.userFormat = format;
4851 stream_.deviceFormat[mode] = getDeviceInfo( device ).nativeFormats;
4853 if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
4854 stream_.userInterleaved = false;
4856 stream_.userInterleaved = true;
4857 stream_.deviceInterleaved[mode] = true;
4859 // Set flags for buffer conversion.
4860 stream_.doConvertBuffer[mode] = false;
4861 if ( stream_.userFormat != stream_.deviceFormat[mode] ||
4862 stream_.nUserChannels[0] != stream_.nDeviceChannels[0] ||
4863 stream_.nUserChannels[1] != stream_.nDeviceChannels[1] )
4864 stream_.doConvertBuffer[mode] = true;
4865 else if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
4866 stream_.nUserChannels[mode] > 1 )
4867 stream_.doConvertBuffer[mode] = true;
4869 if ( stream_.doConvertBuffer[mode] )
4870 setConvertInfo( mode, 0 );
4872 // Allocate necessary internal buffers
4873 bufferBytes = stream_.nUserChannels[mode] * stream_.bufferSize * formatBytes( stream_.userFormat );
4875 stream_.userBuffer[mode] = ( char* ) calloc( bufferBytes, 1 );
4876 if ( !stream_.userBuffer[mode] ) {
4877 errorType = RtAudioError::MEMORY_ERROR;
4878 errorText_ = "RtApiWasapi::probeDeviceOpen: Error allocating user buffer memory.";
4882 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME )
4883 stream_.callbackInfo.priority = 15;
4885 stream_.callbackInfo.priority = 0;
4887 ///! TODO: RTAUDIO_MINIMIZE_LATENCY // Provide stream buffers directly to callback
4888 ///! TODO: RTAUDIO_HOG_DEVICE // Exclusive mode
4890 methodResult = SUCCESS;
4894 SAFE_RELEASE( captureDevices );
4895 SAFE_RELEASE( renderDevices );
4896 SAFE_RELEASE( devicePtr );
4897 CoTaskMemFree( deviceFormat );
4899 // if method failed, close the stream
4900 if ( methodResult == FAILURE )
4903 if ( !errorText_.empty() )
4905 return methodResult;
4908 //=============================================================================
4910 DWORD WINAPI RtApiWasapi::runWasapiThread( void* wasapiPtr )
4913 ( ( RtApiWasapi* ) wasapiPtr )->wasapiThread();
4918 DWORD WINAPI RtApiWasapi::stopWasapiThread( void* wasapiPtr )
4921 ( ( RtApiWasapi* ) wasapiPtr )->stopStream();
4926 DWORD WINAPI RtApiWasapi::abortWasapiThread( void* wasapiPtr )
4929 ( ( RtApiWasapi* ) wasapiPtr )->abortStream();
4934 //-----------------------------------------------------------------------------
4936 void RtApiWasapi::wasapiThread()
4938 // as this is a new thread, we must CoInitialize it
4939 CoInitialize( NULL );
4943 IAudioClient* captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
4944 IAudioClient* renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
4945 IAudioCaptureClient* captureClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureClient;
4946 IAudioRenderClient* renderClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderClient;
4947 HANDLE captureEvent = ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent;
4948 HANDLE renderEvent = ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent;
4950 WAVEFORMATEX* captureFormat = NULL;
4951 WAVEFORMATEX* renderFormat = NULL;
4952 float captureSrRatio = 0.0f;
4953 float renderSrRatio = 0.0f;
4954 WasapiBuffer captureBuffer;
4955 WasapiBuffer renderBuffer;
4956 WasapiResampler* captureResampler = NULL;
4957 WasapiResampler* renderResampler = NULL;
4959 // declare local stream variables
4960 RtAudioCallback callback = ( RtAudioCallback ) stream_.callbackInfo.callback;
4961 BYTE* streamBuffer = NULL;
4962 unsigned long captureFlags = 0;
4963 unsigned int bufferFrameCount = 0;
4964 unsigned int numFramesPadding = 0;
4965 unsigned int convBufferSize = 0;
4966 bool loopbackEnabled = stream_.device[INPUT] == stream_.device[OUTPUT];
4967 bool callbackPushed = true;
4968 bool callbackPulled = false;
4969 bool callbackStopped = false;
4970 int callbackResult = 0;
4972 // convBuffer is used to store converted buffers between WASAPI and the user
4973 char* convBuffer = NULL;
4974 unsigned int convBuffSize = 0;
4975 unsigned int deviceBuffSize = 0;
4977 std::string errorText;
4978 RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
4980 // Attempt to assign "Pro Audio" characteristic to thread
4981 HMODULE AvrtDll = LoadLibrary( (LPCTSTR) "AVRT.dll" );
4983 DWORD taskIndex = 0;
4984 TAvSetMmThreadCharacteristicsPtr AvSetMmThreadCharacteristicsPtr = ( TAvSetMmThreadCharacteristicsPtr ) GetProcAddress( AvrtDll, "AvSetMmThreadCharacteristicsW" );
4985 AvSetMmThreadCharacteristicsPtr( L"Pro Audio", &taskIndex );
4986 FreeLibrary( AvrtDll );
4989 // start capture stream if applicable
4990 if ( captureAudioClient ) {
4991 hr = captureAudioClient->GetMixFormat( &captureFormat );
4992 if ( FAILED( hr ) ) {
4993 errorText = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
4997 // init captureResampler
4998 captureResampler = new WasapiResampler( stream_.deviceFormat[INPUT] == RTAUDIO_FLOAT32 || stream_.deviceFormat[INPUT] == RTAUDIO_FLOAT64,
4999 formatBytes( stream_.deviceFormat[INPUT] ) * 8, stream_.nDeviceChannels[INPUT],
5000 captureFormat->nSamplesPerSec, stream_.sampleRate );
5002 captureSrRatio = ( ( float ) captureFormat->nSamplesPerSec / stream_.sampleRate );
5004 if ( !captureClient ) {
5005 hr = captureAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
5006 loopbackEnabled ? AUDCLNT_STREAMFLAGS_LOOPBACK : AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
5011 if ( FAILED( hr ) ) {
5012 errorText = "RtApiWasapi::wasapiThread: Unable to initialize capture audio client.";
5016 hr = captureAudioClient->GetService( __uuidof( IAudioCaptureClient ),
5017 ( void** ) &captureClient );
5018 if ( FAILED( hr ) ) {
5019 errorText = "RtApiWasapi::wasapiThread: Unable to retrieve capture client handle.";
5023 // don't configure captureEvent if in loopback mode
5024 if ( !loopbackEnabled )
5026 // configure captureEvent to trigger on every available capture buffer
5027 captureEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
5028 if ( !captureEvent ) {
5029 errorType = RtAudioError::SYSTEM_ERROR;
5030 errorText = "RtApiWasapi::wasapiThread: Unable to create capture event.";
5034 hr = captureAudioClient->SetEventHandle( captureEvent );
5035 if ( FAILED( hr ) ) {
5036 errorText = "RtApiWasapi::wasapiThread: Unable to set capture event handle.";
5040 ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent = captureEvent;
5043 ( ( WasapiHandle* ) stream_.apiHandle )->captureClient = captureClient;
5046 unsigned int inBufferSize = 0;
5047 hr = captureAudioClient->GetBufferSize( &inBufferSize );
5048 if ( FAILED( hr ) ) {
5049 errorText = "RtApiWasapi::wasapiThread: Unable to get capture buffer size.";
5053 // scale outBufferSize according to stream->user sample rate ratio
5054 unsigned int outBufferSize = ( unsigned int ) ceilf( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT];
5055 inBufferSize *= stream_.nDeviceChannels[INPUT];
5057 // set captureBuffer size
5058 captureBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[INPUT] ) );
5060 // reset the capture stream
5061 hr = captureAudioClient->Reset();
5062 if ( FAILED( hr ) ) {
5063 errorText = "RtApiWasapi::wasapiThread: Unable to reset capture stream.";
5067 // start the capture stream
5068 hr = captureAudioClient->Start();
5069 if ( FAILED( hr ) ) {
5070 errorText = "RtApiWasapi::wasapiThread: Unable to start capture stream.";
5075 // start render stream if applicable
5076 if ( renderAudioClient ) {
5077 hr = renderAudioClient->GetMixFormat( &renderFormat );
5078 if ( FAILED( hr ) ) {
5079 errorText = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
5083 // init renderResampler
5084 renderResampler = new WasapiResampler( stream_.deviceFormat[OUTPUT] == RTAUDIO_FLOAT32 || stream_.deviceFormat[OUTPUT] == RTAUDIO_FLOAT64,
5085 formatBytes( stream_.deviceFormat[OUTPUT] ) * 8, stream_.nDeviceChannels[OUTPUT],
5086 stream_.sampleRate, renderFormat->nSamplesPerSec );
5088 renderSrRatio = ( ( float ) renderFormat->nSamplesPerSec / stream_.sampleRate );
5090 if ( !renderClient ) {
5091 hr = renderAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
5092 AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
5097 if ( FAILED( hr ) ) {
5098 errorText = "RtApiWasapi::wasapiThread: Unable to initialize render audio client.";
5102 hr = renderAudioClient->GetService( __uuidof( IAudioRenderClient ),
5103 ( void** ) &renderClient );
5104 if ( FAILED( hr ) ) {
5105 errorText = "RtApiWasapi::wasapiThread: Unable to retrieve render client handle.";
5109 // configure renderEvent to trigger on every available render buffer
5110 renderEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
5111 if ( !renderEvent ) {
5112 errorType = RtAudioError::SYSTEM_ERROR;
5113 errorText = "RtApiWasapi::wasapiThread: Unable to create render event.";
5117 hr = renderAudioClient->SetEventHandle( renderEvent );
5118 if ( FAILED( hr ) ) {
5119 errorText = "RtApiWasapi::wasapiThread: Unable to set render event handle.";
5123 ( ( WasapiHandle* ) stream_.apiHandle )->renderClient = renderClient;
5124 ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent = renderEvent;
5127 unsigned int outBufferSize = 0;
5128 hr = renderAudioClient->GetBufferSize( &outBufferSize );
5129 if ( FAILED( hr ) ) {
5130 errorText = "RtApiWasapi::wasapiThread: Unable to get render buffer size.";
5134 // scale inBufferSize according to user->stream sample rate ratio
5135 unsigned int inBufferSize = ( unsigned int ) ceilf( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT];
5136 outBufferSize *= stream_.nDeviceChannels[OUTPUT];
5138 // set renderBuffer size
5139 renderBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[OUTPUT] ) );
5141 // reset the render stream
5142 hr = renderAudioClient->Reset();
5143 if ( FAILED( hr ) ) {
5144 errorText = "RtApiWasapi::wasapiThread: Unable to reset render stream.";
5148 // start the render stream
5149 hr = renderAudioClient->Start();
5150 if ( FAILED( hr ) ) {
5151 errorText = "RtApiWasapi::wasapiThread: Unable to start render stream.";
5156 // malloc buffer memory
5157 if ( stream_.mode == INPUT )
5159 using namespace std; // for ceilf
5160 convBuffSize = ( size_t ) ( ceilf( stream_.bufferSize * captureSrRatio ) ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
5161 deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
5163 else if ( stream_.mode == OUTPUT )
5165 convBuffSize = ( size_t ) ( ceilf( stream_.bufferSize * renderSrRatio ) ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
5166 deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
5168 else if ( stream_.mode == DUPLEX )
5170 convBuffSize = std::max( ( size_t ) ( ceilf( stream_.bufferSize * captureSrRatio ) ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
5171 ( size_t ) ( ceilf( stream_.bufferSize * renderSrRatio ) ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
5172 deviceBuffSize = std::max( stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
5173 stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
5176 convBuffSize *= 2; // allow overflow for *SrRatio remainders
5177 convBuffer = ( char* ) malloc( convBuffSize );
5178 stream_.deviceBuffer = ( char* ) malloc( deviceBuffSize );
5179 if ( !convBuffer || !stream_.deviceBuffer ) {
5180 errorType = RtAudioError::MEMORY_ERROR;
5181 errorText = "RtApiWasapi::wasapiThread: Error allocating device buffer memory.";
5185 // stream process loop
5186 while ( stream_.state != STREAM_STOPPING ) {
5187 if ( !callbackPulled ) {
5190 // 1. Pull callback buffer from inputBuffer
5191 // 2. If 1. was successful: Convert callback buffer to user sample rate and channel count
5192 // Convert callback buffer to user format
5194 if ( captureAudioClient )
5196 int samplesToPull = ( unsigned int ) floorf( stream_.bufferSize * captureSrRatio );
5197 if ( captureSrRatio != 1 )
5199 // account for remainders
5204 while ( convBufferSize < stream_.bufferSize )
5206 // Pull callback buffer from inputBuffer
5207 callbackPulled = captureBuffer.pullBuffer( convBuffer,
5208 samplesToPull * stream_.nDeviceChannels[INPUT],
5209 stream_.deviceFormat[INPUT] );
5211 if ( !callbackPulled )
5216 // Convert callback buffer to user sample rate
5217 unsigned int deviceBufferOffset = convBufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
5218 unsigned int convSamples = 0;
5220 captureResampler->Convert( stream_.deviceBuffer + deviceBufferOffset,
5225 convBufferSize += convSamples;
5226 samplesToPull = 1; // now pull one sample at a time until we have stream_.bufferSize samples
5229 if ( callbackPulled )
5231 if ( stream_.doConvertBuffer[INPUT] ) {
5232 // Convert callback buffer to user format
5233 convertBuffer( stream_.userBuffer[INPUT],
5234 stream_.deviceBuffer,
5235 stream_.convertInfo[INPUT] );
5238 // no further conversion, simple copy deviceBuffer to userBuffer
5239 memcpy( stream_.userBuffer[INPUT],
5240 stream_.deviceBuffer,
5241 stream_.bufferSize * stream_.nUserChannels[INPUT] * formatBytes( stream_.userFormat ) );
5246 // if there is no capture stream, set callbackPulled flag
5247 callbackPulled = true;
5252 // 1. Execute user callback method
5253 // 2. Handle return value from callback
5255 // if callback has not requested the stream to stop
5256 if ( callbackPulled && !callbackStopped ) {
5257 // Execute user callback method
5258 callbackResult = callback( stream_.userBuffer[OUTPUT],
5259 stream_.userBuffer[INPUT],
5262 captureFlags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY ? RTAUDIO_INPUT_OVERFLOW : 0,
5263 stream_.callbackInfo.userData );
5265 // Handle return value from callback
5266 if ( callbackResult == 1 ) {
5267 // instantiate a thread to stop this thread
5268 HANDLE threadHandle = CreateThread( NULL, 0, stopWasapiThread, this, 0, NULL );
5269 if ( !threadHandle ) {
5270 errorType = RtAudioError::THREAD_ERROR;
5271 errorText = "RtApiWasapi::wasapiThread: Unable to instantiate stream stop thread.";
5274 else if ( !CloseHandle( threadHandle ) ) {
5275 errorType = RtAudioError::THREAD_ERROR;
5276 errorText = "RtApiWasapi::wasapiThread: Unable to close stream stop thread handle.";
5280 callbackStopped = true;
5282 else if ( callbackResult == 2 ) {
5283 // instantiate a thread to stop this thread
5284 HANDLE threadHandle = CreateThread( NULL, 0, abortWasapiThread, this, 0, NULL );
5285 if ( !threadHandle ) {
5286 errorType = RtAudioError::THREAD_ERROR;
5287 errorText = "RtApiWasapi::wasapiThread: Unable to instantiate stream abort thread.";
5290 else if ( !CloseHandle( threadHandle ) ) {
5291 errorType = RtAudioError::THREAD_ERROR;
5292 errorText = "RtApiWasapi::wasapiThread: Unable to close stream abort thread handle.";
5296 callbackStopped = true;
5303 // 1. Convert callback buffer to stream format
5304 // 2. Convert callback buffer to stream sample rate and channel count
5305 // 3. Push callback buffer into outputBuffer
5307 if ( renderAudioClient && callbackPulled )
5309 // if the last call to renderBuffer.PushBuffer() was successful
5310 if ( callbackPushed || convBufferSize == 0 )
5312 if ( stream_.doConvertBuffer[OUTPUT] )
5314 // Convert callback buffer to stream format
5315 convertBuffer( stream_.deviceBuffer,
5316 stream_.userBuffer[OUTPUT],
5317 stream_.convertInfo[OUTPUT] );
5321 // Convert callback buffer to stream sample rate
5322 renderResampler->Convert( convBuffer,
5323 stream_.deviceBuffer,
5328 // Push callback buffer into outputBuffer
5329 callbackPushed = renderBuffer.pushBuffer( convBuffer,
5330 convBufferSize * stream_.nDeviceChannels[OUTPUT],
5331 stream_.deviceFormat[OUTPUT] );
5334 // if there is no render stream, set callbackPushed flag
5335 callbackPushed = true;
5340 // 1. Get capture buffer from stream
5341 // 2. Push capture buffer into inputBuffer
5342 // 3. If 2. was successful: Release capture buffer
5344 if ( captureAudioClient ) {
5345 // if the callback input buffer was not pulled from captureBuffer, wait for next capture event
5346 if ( !callbackPulled ) {
5347 WaitForSingleObject( loopbackEnabled ? renderEvent : captureEvent, INFINITE );
5350 // Get capture buffer from stream
5351 hr = captureClient->GetBuffer( &streamBuffer,
5353 &captureFlags, NULL, NULL );
5354 if ( FAILED( hr ) ) {
5355 errorText = "RtApiWasapi::wasapiThread: Unable to retrieve capture buffer.";
5359 if ( bufferFrameCount != 0 ) {
5360 // Push capture buffer into inputBuffer
5361 if ( captureBuffer.pushBuffer( ( char* ) streamBuffer,
5362 bufferFrameCount * stream_.nDeviceChannels[INPUT],
5363 stream_.deviceFormat[INPUT] ) )
5365 // Release capture buffer
5366 hr = captureClient->ReleaseBuffer( bufferFrameCount );
5367 if ( FAILED( hr ) ) {
5368 errorText = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
5374 // Inform WASAPI that capture was unsuccessful
5375 hr = captureClient->ReleaseBuffer( 0 );
5376 if ( FAILED( hr ) ) {
5377 errorText = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
5384 // Inform WASAPI that capture was unsuccessful
5385 hr = captureClient->ReleaseBuffer( 0 );
5386 if ( FAILED( hr ) ) {
5387 errorText = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
5395 // 1. Get render buffer from stream
5396 // 2. Pull next buffer from outputBuffer
5397 // 3. If 2. was successful: Fill render buffer with next buffer
5398 // Release render buffer
5400 if ( renderAudioClient ) {
5401 // if the callback output buffer was not pushed to renderBuffer, wait for next render event
5402 if ( callbackPulled && !callbackPushed ) {
5403 WaitForSingleObject( renderEvent, INFINITE );
5406 // Get render buffer from stream
5407 hr = renderAudioClient->GetBufferSize( &bufferFrameCount );
5408 if ( FAILED( hr ) ) {
5409 errorText = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer size.";
5413 hr = renderAudioClient->GetCurrentPadding( &numFramesPadding );
5414 if ( FAILED( hr ) ) {
5415 errorText = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer padding.";
5419 bufferFrameCount -= numFramesPadding;
5421 if ( bufferFrameCount != 0 ) {
5422 hr = renderClient->GetBuffer( bufferFrameCount, &streamBuffer );
5423 if ( FAILED( hr ) ) {
5424 errorText = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer.";
5428 // Pull next buffer from outputBuffer
5429 // Fill render buffer with next buffer
5430 if ( renderBuffer.pullBuffer( ( char* ) streamBuffer,
5431 bufferFrameCount * stream_.nDeviceChannels[OUTPUT],
5432 stream_.deviceFormat[OUTPUT] ) )
5434 // Release render buffer
5435 hr = renderClient->ReleaseBuffer( bufferFrameCount, 0 );
5436 if ( FAILED( hr ) ) {
5437 errorText = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
5443 // Inform WASAPI that render was unsuccessful
5444 hr = renderClient->ReleaseBuffer( 0, 0 );
5445 if ( FAILED( hr ) ) {
5446 errorText = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
5453 // Inform WASAPI that render was unsuccessful
5454 hr = renderClient->ReleaseBuffer( 0, 0 );
5455 if ( FAILED( hr ) ) {
5456 errorText = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
5462 // if the callback buffer was pushed renderBuffer reset callbackPulled flag
5463 if ( callbackPushed ) {
5464 // unsetting the callbackPulled flag lets the stream know that
5465 // the audio device is ready for another callback output buffer.
5466 callbackPulled = false;
5469 RtApi::tickStreamTime();
5476 CoTaskMemFree( captureFormat );
5477 CoTaskMemFree( renderFormat );
5479 free ( convBuffer );
5480 delete renderResampler;
5481 delete captureResampler;
5485 // update stream state
5486 stream_.state = STREAM_STOPPED;
5488 if ( !errorText.empty() )
5490 errorText_ = errorText;
5495 //******************** End of __WINDOWS_WASAPI__ *********************//
5499 #if defined(__WINDOWS_DS__) // Windows DirectSound API
5501 // Modified by Robin Davies, October 2005
5502 // - Improvements to DirectX pointer chasing.
5503 // - Bug fix for non-power-of-two Asio granularity used by Edirol PCR-A30.
5504 // - Auto-call CoInitialize for DSOUND and ASIO platforms.
5505 // Various revisions for RtAudio 4.0 by Gary Scavone, April 2007
5506 // Changed device query structure for RtAudio 4.0.7, January 2010
5508 #include <windows.h>
5509 #include <process.h>
5510 #include <mmsystem.h>
5514 #include <algorithm>
5516 #if defined(__MINGW32__)
5517 // missing from latest mingw winapi
5518 #define WAVE_FORMAT_96M08 0x00010000 /* 96 kHz, Mono, 8-bit */
5519 #define WAVE_FORMAT_96S08 0x00020000 /* 96 kHz, Stereo, 8-bit */
5520 #define WAVE_FORMAT_96M16 0x00040000 /* 96 kHz, Mono, 16-bit */
5521 #define WAVE_FORMAT_96S16 0x00080000 /* 96 kHz, Stereo, 16-bit */
5524 #define MINIMUM_DEVICE_BUFFER_SIZE 32768
5526 #ifdef _MSC_VER // if Microsoft Visual C++
5527 #pragma comment( lib, "winmm.lib" ) // then, auto-link winmm.lib. Otherwise, it has to be added manually.
5530 static inline DWORD dsPointerBetween( DWORD pointer, DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize )
5532 if ( pointer > bufferSize ) pointer -= bufferSize;
5533 if ( laterPointer < earlierPointer ) laterPointer += bufferSize;
5534 if ( pointer < earlierPointer ) pointer += bufferSize;
5535 return pointer >= earlierPointer && pointer < laterPointer;
5538 // A structure to hold various information related to the DirectSound
5539 // API implementation.
5541 unsigned int drainCounter; // Tracks callback counts when draining
5542 bool internalDrain; // Indicates if stop is initiated from callback or not.
5546 UINT bufferPointer[2];
5547 DWORD dsBufferSize[2];
5548 DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by.
5552 :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; }
5555 // Declarations for utility functions, callbacks, and structures
5556 // specific to the DirectSound implementation.
5557 static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
5558 LPCTSTR description,
5562 static const char* getErrorString( int code );
5564 static unsigned __stdcall callbackHandler( void *ptr );
5573 : found(false) { validId[0] = false; validId[1] = false; }
5576 struct DsProbeData {
5578 std::vector<struct DsDevice>* dsDevices;
5581 RtApiDs :: RtApiDs()
5583 // Dsound will run both-threaded. If CoInitialize fails, then just
5584 // accept whatever the mainline chose for a threading model.
5585 coInitialized_ = false;
5586 HRESULT hr = CoInitialize( NULL );
5587 if ( !FAILED( hr ) ) coInitialized_ = true;
5590 RtApiDs :: ~RtApiDs()
5592 if ( stream_.state != STREAM_CLOSED ) closeStream();
5593 if ( coInitialized_ ) CoUninitialize(); // balanced call.
5596 // The DirectSound default output is always the first device.
5597 unsigned int RtApiDs :: getDefaultOutputDevice( void )
5602 // The DirectSound default input is always the first input device,
5603 // which is the first capture device enumerated.
5604 unsigned int RtApiDs :: getDefaultInputDevice( void )
5609 unsigned int RtApiDs :: getDeviceCount( void )
5611 // Set query flag for previously found devices to false, so that we
5612 // can check for any devices that have disappeared.
5613 for ( unsigned int i=0; i<dsDevices.size(); i++ )
5614 dsDevices[i].found = false;
5616 // Query DirectSound devices.
5617 struct DsProbeData probeInfo;
5618 probeInfo.isInput = false;
5619 probeInfo.dsDevices = &dsDevices;
5620 HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );
5621 if ( FAILED( result ) ) {
5622 errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!";
5623 errorText_ = errorStream_.str();
5624 error( RtAudioError::WARNING );
5627 // Query DirectSoundCapture devices.
5628 probeInfo.isInput = true;
5629 result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );
5630 if ( FAILED( result ) ) {
5631 errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!";
5632 errorText_ = errorStream_.str();
5633 error( RtAudioError::WARNING );
5636 // Clean out any devices that may have disappeared (code update submitted by Eli Zehngut).
5637 for ( unsigned int i=0; i<dsDevices.size(); ) {
5638 if ( dsDevices[i].found == false ) dsDevices.erase( dsDevices.begin() + i );
5642 return static_cast<unsigned int>(dsDevices.size());
5645 RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device )
5647 RtAudio::DeviceInfo info;
5648 info.probed = false;
5650 if ( dsDevices.size() == 0 ) {
5651 // Force a query of all devices
5653 if ( dsDevices.size() == 0 ) {
5654 errorText_ = "RtApiDs::getDeviceInfo: no devices found!";
5655 error( RtAudioError::INVALID_USE );
5660 if ( device >= dsDevices.size() ) {
5661 errorText_ = "RtApiDs::getDeviceInfo: device ID is invalid!";
5662 error( RtAudioError::INVALID_USE );
5667 if ( dsDevices[ device ].validId[0] == false ) goto probeInput;
5669 LPDIRECTSOUND output;
5671 result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );
5672 if ( FAILED( result ) ) {
5673 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!";
5674 errorText_ = errorStream_.str();
5675 error( RtAudioError::WARNING );
5679 outCaps.dwSize = sizeof( outCaps );
5680 result = output->GetCaps( &outCaps );
5681 if ( FAILED( result ) ) {
5683 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting capabilities!";
5684 errorText_ = errorStream_.str();
5685 error( RtAudioError::WARNING );
5689 // Get output channel information.
5690 info.outputChannels = ( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1;
5692 // Get sample rate information.
5693 info.sampleRates.clear();
5694 for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
5695 if ( SAMPLE_RATES[k] >= (unsigned int) outCaps.dwMinSecondarySampleRate &&
5696 SAMPLE_RATES[k] <= (unsigned int) outCaps.dwMaxSecondarySampleRate ) {
5697 info.sampleRates.push_back( SAMPLE_RATES[k] );
5699 if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
5700 info.preferredSampleRate = SAMPLE_RATES[k];
5704 // Get format information.
5705 if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT ) info.nativeFormats |= RTAUDIO_SINT16;
5706 if ( outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) info.nativeFormats |= RTAUDIO_SINT8;
5710 if ( getDefaultOutputDevice() == device )
5711 info.isDefaultOutput = true;
5713 if ( dsDevices[ device ].validId[1] == false ) {
5714 info.name = dsDevices[ device ].name;
5721 LPDIRECTSOUNDCAPTURE input;
5722 result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );
5723 if ( FAILED( result ) ) {
5724 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";
5725 errorText_ = errorStream_.str();
5726 error( RtAudioError::WARNING );
5731 inCaps.dwSize = sizeof( inCaps );
5732 result = input->GetCaps( &inCaps );
5733 if ( FAILED( result ) ) {
5735 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsDevices[ device ].name << ")!";
5736 errorText_ = errorStream_.str();
5737 error( RtAudioError::WARNING );
5741 // Get input channel information.
5742 info.inputChannels = inCaps.dwChannels;
5744 // Get sample rate and format information.
5745 std::vector<unsigned int> rates;
5746 if ( inCaps.dwChannels >= 2 ) {
5747 if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.nativeFormats |= RTAUDIO_SINT16;
5748 if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.nativeFormats |= RTAUDIO_SINT16;
5749 if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.nativeFormats |= RTAUDIO_SINT16;
5750 if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.nativeFormats |= RTAUDIO_SINT16;
5751 if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.nativeFormats |= RTAUDIO_SINT8;
5752 if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.nativeFormats |= RTAUDIO_SINT8;
5753 if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.nativeFormats |= RTAUDIO_SINT8;
5754 if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.nativeFormats |= RTAUDIO_SINT8;
5756 if ( info.nativeFormats & RTAUDIO_SINT16 ) {
5757 if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) rates.push_back( 11025 );
5758 if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) rates.push_back( 22050 );
5759 if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) rates.push_back( 44100 );
5760 if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) rates.push_back( 96000 );
5762 else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
5763 if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) rates.push_back( 11025 );
5764 if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) rates.push_back( 22050 );
5765 if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) rates.push_back( 44100 );
5766 if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) rates.push_back( 96000 );
5769 else if ( inCaps.dwChannels == 1 ) {
5770 if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.nativeFormats |= RTAUDIO_SINT16;
5771 if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.nativeFormats |= RTAUDIO_SINT16;
5772 if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.nativeFormats |= RTAUDIO_SINT16;
5773 if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.nativeFormats |= RTAUDIO_SINT16;
5774 if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.nativeFormats |= RTAUDIO_SINT8;
5775 if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.nativeFormats |= RTAUDIO_SINT8;
5776 if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.nativeFormats |= RTAUDIO_SINT8;
5777 if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.nativeFormats |= RTAUDIO_SINT8;
5779 if ( info.nativeFormats & RTAUDIO_SINT16 ) {
5780 if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) rates.push_back( 11025 );
5781 if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) rates.push_back( 22050 );
5782 if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) rates.push_back( 44100 );
5783 if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) rates.push_back( 96000 );
5785 else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
5786 if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) rates.push_back( 11025 );
5787 if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) rates.push_back( 22050 );
5788 if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) rates.push_back( 44100 );
5789 if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) rates.push_back( 96000 );
5792 else info.inputChannels = 0; // technically, this would be an error
5796 if ( info.inputChannels == 0 ) return info;
5798 // Copy the supported rates to the info structure but avoid duplication.
5800 for ( unsigned int i=0; i<rates.size(); i++ ) {
5802 for ( unsigned int j=0; j<info.sampleRates.size(); j++ ) {
5803 if ( rates[i] == info.sampleRates[j] ) {
5808 if ( found == false ) info.sampleRates.push_back( rates[i] );
5810 std::sort( info.sampleRates.begin(), info.sampleRates.end() );
5812 // If device opens for both playback and capture, we determine the channels.
5813 if ( info.outputChannels > 0 && info.inputChannels > 0 )
5814 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
5816 if ( device == 0 ) info.isDefaultInput = true;
5818 // Copy name and return.
5819 info.name = dsDevices[ device ].name;
5824 bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
5825 unsigned int firstChannel, unsigned int sampleRate,
5826 RtAudioFormat format, unsigned int *bufferSize,
5827 RtAudio::StreamOptions *options )
5829 if ( channels + firstChannel > 2 ) {
5830 errorText_ = "RtApiDs::probeDeviceOpen: DirectSound does not support more than 2 channels per device.";
5834 size_t nDevices = dsDevices.size();
5835 if ( nDevices == 0 ) {
5836 // This should not happen because a check is made before this function is called.
5837 errorText_ = "RtApiDs::probeDeviceOpen: no devices found!";
5841 if ( device >= nDevices ) {
5842 // This should not happen because a check is made before this function is called.
5843 errorText_ = "RtApiDs::probeDeviceOpen: device ID is invalid!";
5847 if ( mode == OUTPUT ) {
5848 if ( dsDevices[ device ].validId[0] == false ) {
5849 errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support output!";
5850 errorText_ = errorStream_.str();
5854 else { // mode == INPUT
5855 if ( dsDevices[ device ].validId[1] == false ) {
5856 errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support input!";
5857 errorText_ = errorStream_.str();
5862 // According to a note in PortAudio, using GetDesktopWindow()
5863 // instead of GetForegroundWindow() is supposed to avoid problems
5864 // that occur when the application's window is not the foreground
5865 // window. Also, if the application window closes before the
5866 // DirectSound buffer, DirectSound can crash. In the past, I had
5867 // problems when using GetDesktopWindow() but it seems fine now
5868 // (January 2010). I'll leave it commented here.
5869 // HWND hWnd = GetForegroundWindow();
5870 HWND hWnd = GetDesktopWindow();
5872 // Check the numberOfBuffers parameter and limit the lowest value to
5873 // two. This is a judgement call and a value of two is probably too
5874 // low for capture, but it should work for playback.
5876 if ( options ) nBuffers = options->numberOfBuffers;
5877 if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) nBuffers = 2;
5878 if ( nBuffers < 2 ) nBuffers = 3;
5880 // Check the lower range of the user-specified buffer size and set
5881 // (arbitrarily) to a lower bound of 32.
5882 if ( *bufferSize < 32 ) *bufferSize = 32;
5884 // Create the wave format structure. The data format setting will
5885 // be determined later.
5886 WAVEFORMATEX waveFormat;
5887 ZeroMemory( &waveFormat, sizeof(WAVEFORMATEX) );
5888 waveFormat.wFormatTag = WAVE_FORMAT_PCM;
5889 waveFormat.nChannels = channels + firstChannel;
5890 waveFormat.nSamplesPerSec = (unsigned long) sampleRate;
5892 // Determine the device buffer size. By default, we'll use the value
5893 // defined above (32K), but we will grow it to make allowances for
5894 // very large software buffer sizes.
5895 DWORD dsBufferSize = MINIMUM_DEVICE_BUFFER_SIZE;
5896 DWORD dsPointerLeadTime = 0;
5898 void *ohandle = 0, *bhandle = 0;
5900 if ( mode == OUTPUT ) {
5902 LPDIRECTSOUND output;
5903 result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );
5904 if ( FAILED( result ) ) {
5905 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!";
5906 errorText_ = errorStream_.str();
5911 outCaps.dwSize = sizeof( outCaps );
5912 result = output->GetCaps( &outCaps );
5913 if ( FAILED( result ) ) {
5915 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting capabilities (" << dsDevices[ device ].name << ")!";
5916 errorText_ = errorStream_.str();
5920 // Check channel information.
5921 if ( channels + firstChannel == 2 && !( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ) {
5922 errorStream_ << "RtApiDs::getDeviceInfo: the output device (" << dsDevices[ device ].name << ") does not support stereo playback.";
5923 errorText_ = errorStream_.str();
5927 // Check format information. Use 16-bit format unless not
5928 // supported or user requests 8-bit.
5929 if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT &&
5930 !( format == RTAUDIO_SINT8 && outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) ) {
5931 waveFormat.wBitsPerSample = 16;
5932 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
5935 waveFormat.wBitsPerSample = 8;
5936 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
5938 stream_.userFormat = format;
5940 // Update wave format structure and buffer information.
5941 waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
5942 waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
5943 dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
5945 // If the user wants an even bigger buffer, increase the device buffer size accordingly.
5946 while ( dsPointerLeadTime * 2U > dsBufferSize )
5949 // Set cooperative level to DSSCL_EXCLUSIVE ... sound stops when window focus changes.
5950 // result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE );
5951 // Set cooperative level to DSSCL_PRIORITY ... sound remains when window focus changes.
5952 result = output->SetCooperativeLevel( hWnd, DSSCL_PRIORITY );
5953 if ( FAILED( result ) ) {
5955 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting cooperative level (" << dsDevices[ device ].name << ")!";
5956 errorText_ = errorStream_.str();
5960 // Even though we will write to the secondary buffer, we need to
5961 // access the primary buffer to set the correct output format
5962 // (since the default is 8-bit, 22 kHz!). Setup the DS primary
5963 // buffer description.
5964 DSBUFFERDESC bufferDescription;
5965 ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
5966 bufferDescription.dwSize = sizeof( DSBUFFERDESC );
5967 bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER;
5969 // Obtain the primary buffer
5970 LPDIRECTSOUNDBUFFER buffer;
5971 result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
5972 if ( FAILED( result ) ) {
5974 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") accessing primary buffer (" << dsDevices[ device ].name << ")!";
5975 errorText_ = errorStream_.str();
5979 // Set the primary DS buffer sound format.
5980 result = buffer->SetFormat( &waveFormat );
5981 if ( FAILED( result ) ) {
5983 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting primary buffer format (" << dsDevices[ device ].name << ")!";
5984 errorText_ = errorStream_.str();
5988 // Setup the secondary DS buffer description.
5989 ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
5990 bufferDescription.dwSize = sizeof( DSBUFFERDESC );
5991 bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
5992 DSBCAPS_GLOBALFOCUS |
5993 DSBCAPS_GETCURRENTPOSITION2 |
5994 DSBCAPS_LOCHARDWARE ); // Force hardware mixing
5995 bufferDescription.dwBufferBytes = dsBufferSize;
5996 bufferDescription.lpwfxFormat = &waveFormat;
5998 // Try to create the secondary DS buffer. If that doesn't work,
5999 // try to use software mixing. Otherwise, there's a problem.
6000 result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
6001 if ( FAILED( result ) ) {
6002 bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
6003 DSBCAPS_GLOBALFOCUS |
6004 DSBCAPS_GETCURRENTPOSITION2 |
6005 DSBCAPS_LOCSOFTWARE ); // Force software mixing
6006 result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
6007 if ( FAILED( result ) ) {
6009 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating secondary buffer (" << dsDevices[ device ].name << ")!";
6010 errorText_ = errorStream_.str();
6015 // Get the buffer size ... might be different from what we specified.
6017 dsbcaps.dwSize = sizeof( DSBCAPS );
6018 result = buffer->GetCaps( &dsbcaps );
6019 if ( FAILED( result ) ) {
6022 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";
6023 errorText_ = errorStream_.str();
6027 dsBufferSize = dsbcaps.dwBufferBytes;
6029 // Lock the DS buffer
6032 result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );
6033 if ( FAILED( result ) ) {
6036 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking buffer (" << dsDevices[ device ].name << ")!";
6037 errorText_ = errorStream_.str();
6041 // Zero the DS buffer
6042 ZeroMemory( audioPtr, dataLen );
6044 // Unlock the DS buffer
6045 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
6046 if ( FAILED( result ) ) {
6049 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking buffer (" << dsDevices[ device ].name << ")!";
6050 errorText_ = errorStream_.str();
6054 ohandle = (void *) output;
6055 bhandle = (void *) buffer;
6058 if ( mode == INPUT ) {
6060 LPDIRECTSOUNDCAPTURE input;
6061 result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );
6062 if ( FAILED( result ) ) {
6063 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";
6064 errorText_ = errorStream_.str();
6069 inCaps.dwSize = sizeof( inCaps );
6070 result = input->GetCaps( &inCaps );
6071 if ( FAILED( result ) ) {
6073 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsDevices[ device ].name << ")!";
6074 errorText_ = errorStream_.str();
6078 // Check channel information.
6079 if ( inCaps.dwChannels < channels + firstChannel ) {
6080 errorText_ = "RtApiDs::getDeviceInfo: the input device does not support requested input channels.";
6084 // Check format information. Use 16-bit format unless user
6086 DWORD deviceFormats;
6087 if ( channels + firstChannel == 2 ) {
6088 deviceFormats = WAVE_FORMAT_1S08 | WAVE_FORMAT_2S08 | WAVE_FORMAT_4S08 | WAVE_FORMAT_96S08;
6089 if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
6090 waveFormat.wBitsPerSample = 8;
6091 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
6093 else { // assume 16-bit is supported
6094 waveFormat.wBitsPerSample = 16;
6095 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
6098 else { // channel == 1
6099 deviceFormats = WAVE_FORMAT_1M08 | WAVE_FORMAT_2M08 | WAVE_FORMAT_4M08 | WAVE_FORMAT_96M08;
6100 if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
6101 waveFormat.wBitsPerSample = 8;
6102 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
6104 else { // assume 16-bit is supported
6105 waveFormat.wBitsPerSample = 16;
6106 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
6109 stream_.userFormat = format;
6111 // Update wave format structure and buffer information.
6112 waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
6113 waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
6114 dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
6116 // If the user wants an even bigger buffer, increase the device buffer size accordingly.
6117 while ( dsPointerLeadTime * 2U > dsBufferSize )
6120 // Setup the secondary DS buffer description.
6121 DSCBUFFERDESC bufferDescription;
6122 ZeroMemory( &bufferDescription, sizeof( DSCBUFFERDESC ) );
6123 bufferDescription.dwSize = sizeof( DSCBUFFERDESC );
6124 bufferDescription.dwFlags = 0;
6125 bufferDescription.dwReserved = 0;
6126 bufferDescription.dwBufferBytes = dsBufferSize;
6127 bufferDescription.lpwfxFormat = &waveFormat;
6129 // Create the capture buffer.
6130 LPDIRECTSOUNDCAPTUREBUFFER buffer;
6131 result = input->CreateCaptureBuffer( &bufferDescription, &buffer, NULL );
6132 if ( FAILED( result ) ) {
6134 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating input buffer (" << dsDevices[ device ].name << ")!";
6135 errorText_ = errorStream_.str();
6139 // Get the buffer size ... might be different from what we specified.
6141 dscbcaps.dwSize = sizeof( DSCBCAPS );
6142 result = buffer->GetCaps( &dscbcaps );
6143 if ( FAILED( result ) ) {
6146 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";
6147 errorText_ = errorStream_.str();
6151 dsBufferSize = dscbcaps.dwBufferBytes;
6153 // NOTE: We could have a problem here if this is a duplex stream
6154 // and the play and capture hardware buffer sizes are different
6155 // (I'm actually not sure if that is a problem or not).
6156 // Currently, we are not verifying that.
6158 // Lock the capture buffer
6161 result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );
6162 if ( FAILED( result ) ) {
6165 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking input buffer (" << dsDevices[ device ].name << ")!";
6166 errorText_ = errorStream_.str();
6171 ZeroMemory( audioPtr, dataLen );
6173 // Unlock the buffer
6174 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
6175 if ( FAILED( result ) ) {
6178 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking input buffer (" << dsDevices[ device ].name << ")!";
6179 errorText_ = errorStream_.str();
6183 ohandle = (void *) input;
6184 bhandle = (void *) buffer;
6187 // Set various stream parameters
6188 DsHandle *handle = 0;
6189 stream_.nDeviceChannels[mode] = channels + firstChannel;
6190 stream_.nUserChannels[mode] = channels;
6191 stream_.bufferSize = *bufferSize;
6192 stream_.channelOffset[mode] = firstChannel;
6193 stream_.deviceInterleaved[mode] = true;
6194 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
6195 else stream_.userInterleaved = true;
6197 // Set flag for buffer conversion
6198 stream_.doConvertBuffer[mode] = false;
6199 if (stream_.nUserChannels[mode] != stream_.nDeviceChannels[mode])
6200 stream_.doConvertBuffer[mode] = true;
6201 if (stream_.userFormat != stream_.deviceFormat[mode])
6202 stream_.doConvertBuffer[mode] = true;
6203 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
6204 stream_.nUserChannels[mode] > 1 )
6205 stream_.doConvertBuffer[mode] = true;
6207 // Allocate necessary internal buffers
6208 long bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
6209 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
6210 if ( stream_.userBuffer[mode] == NULL ) {
6211 errorText_ = "RtApiDs::probeDeviceOpen: error allocating user buffer memory.";
6215 if ( stream_.doConvertBuffer[mode] ) {
6217 bool makeBuffer = true;
6218 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
6219 if ( mode == INPUT ) {
6220 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
6221 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
6222 if ( bufferBytes <= (long) bytesOut ) makeBuffer = false;
6227 bufferBytes *= *bufferSize;
6228 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
6229 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
6230 if ( stream_.deviceBuffer == NULL ) {
6231 errorText_ = "RtApiDs::probeDeviceOpen: error allocating device buffer memory.";
6237 // Allocate our DsHandle structures for the stream.
6238 if ( stream_.apiHandle == 0 ) {
6240 handle = new DsHandle;
6242 catch ( std::bad_alloc& ) {
6243 errorText_ = "RtApiDs::probeDeviceOpen: error allocating AsioHandle memory.";
6247 // Create a manual-reset event.
6248 handle->condition = CreateEvent( NULL, // no security
6249 TRUE, // manual-reset
6250 FALSE, // non-signaled initially
6252 stream_.apiHandle = (void *) handle;
6255 handle = (DsHandle *) stream_.apiHandle;
6256 handle->id[mode] = ohandle;
6257 handle->buffer[mode] = bhandle;
6258 handle->dsBufferSize[mode] = dsBufferSize;
6259 handle->dsPointerLeadTime[mode] = dsPointerLeadTime;
6261 stream_.device[mode] = device;
6262 stream_.state = STREAM_STOPPED;
6263 if ( stream_.mode == OUTPUT && mode == INPUT )
6264 // We had already set up an output stream.
6265 stream_.mode = DUPLEX;
6267 stream_.mode = mode;
6268 stream_.nBuffers = nBuffers;
6269 stream_.sampleRate = sampleRate;
6271 // Setup the buffer conversion information structure.
6272 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
6274 // Setup the callback thread.
6275 if ( stream_.callbackInfo.isRunning == false ) {
6277 stream_.callbackInfo.isRunning = true;
6278 stream_.callbackInfo.object = (void *) this;
6279 stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &callbackHandler,
6280 &stream_.callbackInfo, 0, &threadId );
6281 if ( stream_.callbackInfo.thread == 0 ) {
6282 errorText_ = "RtApiDs::probeDeviceOpen: error creating callback thread!";
6286 // Boost DS thread priority
6287 SetThreadPriority( (HANDLE) stream_.callbackInfo.thread, THREAD_PRIORITY_HIGHEST );
6293 if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
6294 LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
6295 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6296 if ( buffer ) buffer->Release();
6299 if ( handle->buffer[1] ) {
6300 LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
6301 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6302 if ( buffer ) buffer->Release();
6305 CloseHandle( handle->condition );
6307 stream_.apiHandle = 0;
6310 for ( int i=0; i<2; i++ ) {
6311 if ( stream_.userBuffer[i] ) {
6312 free( stream_.userBuffer[i] );
6313 stream_.userBuffer[i] = 0;
6317 if ( stream_.deviceBuffer ) {
6318 free( stream_.deviceBuffer );
6319 stream_.deviceBuffer = 0;
6322 stream_.state = STREAM_CLOSED;
6326 void RtApiDs :: closeStream()
6328 if ( stream_.state == STREAM_CLOSED ) {
6329 errorText_ = "RtApiDs::closeStream(): no open stream to close!";
6330 error( RtAudioError::WARNING );
6334 // Stop the callback thread.
6335 stream_.callbackInfo.isRunning = false;
6336 WaitForSingleObject( (HANDLE) stream_.callbackInfo.thread, INFINITE );
6337 CloseHandle( (HANDLE) stream_.callbackInfo.thread );
6339 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6341 if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
6342 LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
6343 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6350 if ( handle->buffer[1] ) {
6351 LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
6352 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6359 CloseHandle( handle->condition );
6361 stream_.apiHandle = 0;
6364 for ( int i=0; i<2; i++ ) {
6365 if ( stream_.userBuffer[i] ) {
6366 free( stream_.userBuffer[i] );
6367 stream_.userBuffer[i] = 0;
6371 if ( stream_.deviceBuffer ) {
6372 free( stream_.deviceBuffer );
6373 stream_.deviceBuffer = 0;
6376 stream_.mode = UNINITIALIZED;
6377 stream_.state = STREAM_CLOSED;
6380 void RtApiDs :: startStream()
6383 if ( stream_.state == STREAM_RUNNING ) {
6384 errorText_ = "RtApiDs::startStream(): the stream is already running!";
6385 error( RtAudioError::WARNING );
6389 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6391 // Increase scheduler frequency on lesser windows (a side-effect of
6392 // increasing timer accuracy). On greater windows (Win2K or later),
6393 // this is already in effect.
6394 timeBeginPeriod( 1 );
6396 buffersRolling = false;
6397 duplexPrerollBytes = 0;
6399 if ( stream_.mode == DUPLEX ) {
6400 // 0.5 seconds of silence in DUPLEX mode while the devices spin up and synchronize.
6401 duplexPrerollBytes = (int) ( 0.5 * stream_.sampleRate * formatBytes( stream_.deviceFormat[1] ) * stream_.nDeviceChannels[1] );
6405 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
6407 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6408 result = buffer->Play( 0, 0, DSBPLAY_LOOPING );
6409 if ( FAILED( result ) ) {
6410 errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting output buffer!";
6411 errorText_ = errorStream_.str();
6416 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
6418 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6419 result = buffer->Start( DSCBSTART_LOOPING );
6420 if ( FAILED( result ) ) {
6421 errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting input buffer!";
6422 errorText_ = errorStream_.str();
6427 handle->drainCounter = 0;
6428 handle->internalDrain = false;
6429 ResetEvent( handle->condition );
6430 stream_.state = STREAM_RUNNING;
6433 if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );
6436 void RtApiDs :: stopStream()
6439 if ( stream_.state == STREAM_STOPPED ) {
6440 errorText_ = "RtApiDs::stopStream(): the stream is already stopped!";
6441 error( RtAudioError::WARNING );
6448 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6449 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
6450 if ( handle->drainCounter == 0 ) {
6451 handle->drainCounter = 2;
6452 WaitForSingleObject( handle->condition, INFINITE ); // block until signaled
6455 stream_.state = STREAM_STOPPED;
6457 MUTEX_LOCK( &stream_.mutex );
6459 // Stop the buffer and clear memory
6460 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6461 result = buffer->Stop();
6462 if ( FAILED( result ) ) {
6463 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping output buffer!";
6464 errorText_ = errorStream_.str();
6468 // Lock the buffer and clear it so that if we start to play again,
6469 // we won't have old data playing.
6470 result = buffer->Lock( 0, handle->dsBufferSize[0], &audioPtr, &dataLen, NULL, NULL, 0 );
6471 if ( FAILED( result ) ) {
6472 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking output buffer!";
6473 errorText_ = errorStream_.str();
6477 // Zero the DS buffer
6478 ZeroMemory( audioPtr, dataLen );
6480 // Unlock the DS buffer
6481 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
6482 if ( FAILED( result ) ) {
6483 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking output buffer!";
6484 errorText_ = errorStream_.str();
6488 // If we start playing again, we must begin at beginning of buffer.
6489 handle->bufferPointer[0] = 0;
6492 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
6493 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6497 stream_.state = STREAM_STOPPED;
6499 if ( stream_.mode != DUPLEX )
6500 MUTEX_LOCK( &stream_.mutex );
6502 result = buffer->Stop();
6503 if ( FAILED( result ) ) {
6504 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping input buffer!";
6505 errorText_ = errorStream_.str();
6509 // Lock the buffer and clear it so that if we start to play again,
6510 // we won't have old data playing.
6511 result = buffer->Lock( 0, handle->dsBufferSize[1], &audioPtr, &dataLen, NULL, NULL, 0 );
6512 if ( FAILED( result ) ) {
6513 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking input buffer!";
6514 errorText_ = errorStream_.str();
6518 // Zero the DS buffer
6519 ZeroMemory( audioPtr, dataLen );
6521 // Unlock the DS buffer
6522 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
6523 if ( FAILED( result ) ) {
6524 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking input buffer!";
6525 errorText_ = errorStream_.str();
6529 // If we start recording again, we must begin at beginning of buffer.
6530 handle->bufferPointer[1] = 0;
6534 timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows.
6535 MUTEX_UNLOCK( &stream_.mutex );
6537 if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );
6540 void RtApiDs :: abortStream()
6543 if ( stream_.state == STREAM_STOPPED ) {
6544 errorText_ = "RtApiDs::abortStream(): the stream is already stopped!";
6545 error( RtAudioError::WARNING );
6549 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6550 handle->drainCounter = 2;
6555 void RtApiDs :: callbackEvent()
6557 if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) {
6558 Sleep( 50 ); // sleep 50 milliseconds
6562 if ( stream_.state == STREAM_CLOSED ) {
6563 errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!";
6564 error( RtAudioError::WARNING );
6568 CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
6569 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6571 // Check if we were draining the stream and signal is finished.
6572 if ( handle->drainCounter > stream_.nBuffers + 2 ) {
6574 stream_.state = STREAM_STOPPING;
6575 if ( handle->internalDrain == false )
6576 SetEvent( handle->condition );
6582 // Invoke user callback to get fresh output data UNLESS we are
6584 if ( handle->drainCounter == 0 ) {
6585 RtAudioCallback callback = (RtAudioCallback) info->callback;
6586 double streamTime = getStreamTime();
6587 RtAudioStreamStatus status = 0;
6588 if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
6589 status |= RTAUDIO_OUTPUT_UNDERFLOW;
6590 handle->xrun[0] = false;
6592 if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
6593 status |= RTAUDIO_INPUT_OVERFLOW;
6594 handle->xrun[1] = false;
6596 int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
6597 stream_.bufferSize, streamTime, status, info->userData );
6598 if ( cbReturnValue == 2 ) {
6599 stream_.state = STREAM_STOPPING;
6600 handle->drainCounter = 2;
6604 else if ( cbReturnValue == 1 ) {
6605 handle->drainCounter = 1;
6606 handle->internalDrain = true;
6611 DWORD currentWritePointer, safeWritePointer;
6612 DWORD currentReadPointer, safeReadPointer;
6613 UINT nextWritePointer;
6615 LPVOID buffer1 = NULL;
6616 LPVOID buffer2 = NULL;
6617 DWORD bufferSize1 = 0;
6618 DWORD bufferSize2 = 0;
6623 MUTEX_LOCK( &stream_.mutex );
6624 if ( stream_.state == STREAM_STOPPED ) {
6625 MUTEX_UNLOCK( &stream_.mutex );
6629 if ( buffersRolling == false ) {
6630 if ( stream_.mode == DUPLEX ) {
6631 //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
6633 // It takes a while for the devices to get rolling. As a result,
6634 // there's no guarantee that the capture and write device pointers
6635 // will move in lockstep. Wait here for both devices to start
6636 // rolling, and then set our buffer pointers accordingly.
6637 // e.g. Crystal Drivers: the capture buffer starts up 5700 to 9600
6638 // bytes later than the write buffer.
6640 // Stub: a serious risk of having a pre-emptive scheduling round
6641 // take place between the two GetCurrentPosition calls... but I'm
6642 // really not sure how to solve the problem. Temporarily boost to
6643 // Realtime priority, maybe; but I'm not sure what priority the
6644 // DirectSound service threads run at. We *should* be roughly
6645 // within a ms or so of correct.
6647 LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6648 LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6650 DWORD startSafeWritePointer, startSafeReadPointer;
6652 result = dsWriteBuffer->GetCurrentPosition( NULL, &startSafeWritePointer );
6653 if ( FAILED( result ) ) {
6654 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
6655 errorText_ = errorStream_.str();
6656 MUTEX_UNLOCK( &stream_.mutex );
6657 error( RtAudioError::SYSTEM_ERROR );
6660 result = dsCaptureBuffer->GetCurrentPosition( NULL, &startSafeReadPointer );
6661 if ( FAILED( result ) ) {
6662 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
6663 errorText_ = errorStream_.str();
6664 MUTEX_UNLOCK( &stream_.mutex );
6665 error( RtAudioError::SYSTEM_ERROR );
6669 result = dsWriteBuffer->GetCurrentPosition( NULL, &safeWritePointer );
6670 if ( FAILED( result ) ) {
6671 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
6672 errorText_ = errorStream_.str();
6673 MUTEX_UNLOCK( &stream_.mutex );
6674 error( RtAudioError::SYSTEM_ERROR );
6677 result = dsCaptureBuffer->GetCurrentPosition( NULL, &safeReadPointer );
6678 if ( FAILED( result ) ) {
6679 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
6680 errorText_ = errorStream_.str();
6681 MUTEX_UNLOCK( &stream_.mutex );
6682 error( RtAudioError::SYSTEM_ERROR );
6685 if ( safeWritePointer != startSafeWritePointer && safeReadPointer != startSafeReadPointer ) break;
6689 //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
6691 handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];
6692 if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];
6693 handle->bufferPointer[1] = safeReadPointer;
6695 else if ( stream_.mode == OUTPUT ) {
6697 // Set the proper nextWritePosition after initial startup.
6698 LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6699 result = dsWriteBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer );
6700 if ( FAILED( result ) ) {
6701 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
6702 errorText_ = errorStream_.str();
6703 MUTEX_UNLOCK( &stream_.mutex );
6704 error( RtAudioError::SYSTEM_ERROR );
6707 handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];
6708 if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];
6711 buffersRolling = true;
6714 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
6716 LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6718 if ( handle->drainCounter > 1 ) { // write zeros to the output stream
6719 bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
6720 bufferBytes *= formatBytes( stream_.userFormat );
6721 memset( stream_.userBuffer[0], 0, bufferBytes );
6724 // Setup parameters and do buffer conversion if necessary.
6725 if ( stream_.doConvertBuffer[0] ) {
6726 buffer = stream_.deviceBuffer;
6727 convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
6728 bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[0];
6729 bufferBytes *= formatBytes( stream_.deviceFormat[0] );
6732 buffer = stream_.userBuffer[0];
6733 bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
6734 bufferBytes *= formatBytes( stream_.userFormat );
6737 // No byte swapping necessary in DirectSound implementation.
6739 // Ahhh ... windoze. 16-bit data is signed but 8-bit data is
6740 // unsigned. So, we need to convert our signed 8-bit data here to
6742 if ( stream_.deviceFormat[0] == RTAUDIO_SINT8 )
6743 for ( int i=0; i<bufferBytes; i++ ) buffer[i] = (unsigned char) ( buffer[i] + 128 );
6745 DWORD dsBufferSize = handle->dsBufferSize[0];
6746 nextWritePointer = handle->bufferPointer[0];
6748 DWORD endWrite, leadPointer;
6750 // Find out where the read and "safe write" pointers are.
6751 result = dsBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer );
6752 if ( FAILED( result ) ) {
6753 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
6754 errorText_ = errorStream_.str();
6755 MUTEX_UNLOCK( &stream_.mutex );
6756 error( RtAudioError::SYSTEM_ERROR );
6760 // We will copy our output buffer into the region between
6761 // safeWritePointer and leadPointer. If leadPointer is not
6762 // beyond the next endWrite position, wait until it is.
6763 leadPointer = safeWritePointer + handle->dsPointerLeadTime[0];
6764 //std::cout << "safeWritePointer = " << safeWritePointer << ", leadPointer = " << leadPointer << ", nextWritePointer = " << nextWritePointer << std::endl;
6765 if ( leadPointer > dsBufferSize ) leadPointer -= dsBufferSize;
6766 if ( leadPointer < nextWritePointer ) leadPointer += dsBufferSize; // unwrap offset
6767 endWrite = nextWritePointer + bufferBytes;
6769 // Check whether the entire write region is behind the play pointer.
6770 if ( leadPointer >= endWrite ) break;
6772 // If we are here, then we must wait until the leadPointer advances
6773 // beyond the end of our next write region. We use the
6774 // Sleep() function to suspend operation until that happens.
6775 double millis = ( endWrite - leadPointer ) * 1000.0;
6776 millis /= ( formatBytes( stream_.deviceFormat[0]) * stream_.nDeviceChannels[0] * stream_.sampleRate);
6777 if ( millis < 1.0 ) millis = 1.0;
6778 Sleep( (DWORD) millis );
6781 if ( dsPointerBetween( nextWritePointer, safeWritePointer, currentWritePointer, dsBufferSize )
6782 || dsPointerBetween( endWrite, safeWritePointer, currentWritePointer, dsBufferSize ) ) {
6783 // We've strayed into the forbidden zone ... resync the read pointer.
6784 handle->xrun[0] = true;
6785 nextWritePointer = safeWritePointer + handle->dsPointerLeadTime[0] - bufferBytes;
6786 if ( nextWritePointer >= dsBufferSize ) nextWritePointer -= dsBufferSize;
6787 handle->bufferPointer[0] = nextWritePointer;
6788 endWrite = nextWritePointer + bufferBytes;
6791 // Lock free space in the buffer
6792 result = dsBuffer->Lock( nextWritePointer, bufferBytes, &buffer1,
6793 &bufferSize1, &buffer2, &bufferSize2, 0 );
6794 if ( FAILED( result ) ) {
6795 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking buffer during playback!";
6796 errorText_ = errorStream_.str();
6797 MUTEX_UNLOCK( &stream_.mutex );
6798 error( RtAudioError::SYSTEM_ERROR );
6802 // Copy our buffer into the DS buffer
6803 CopyMemory( buffer1, buffer, bufferSize1 );
6804 if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 );
6806 // Update our buffer offset and unlock sound buffer
6807 dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
6808 if ( FAILED( result ) ) {
6809 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!";
6810 errorText_ = errorStream_.str();
6811 MUTEX_UNLOCK( &stream_.mutex );
6812 error( RtAudioError::SYSTEM_ERROR );
6815 nextWritePointer = ( nextWritePointer + bufferSize1 + bufferSize2 ) % dsBufferSize;
6816 handle->bufferPointer[0] = nextWritePointer;
6819 // Don't bother draining input
6820 if ( handle->drainCounter ) {
6821 handle->drainCounter++;
6825 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
6827 // Setup parameters.
6828 if ( stream_.doConvertBuffer[1] ) {
6829 buffer = stream_.deviceBuffer;
6830 bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[1];
6831 bufferBytes *= formatBytes( stream_.deviceFormat[1] );
6834 buffer = stream_.userBuffer[1];
6835 bufferBytes = stream_.bufferSize * stream_.nUserChannels[1];
6836 bufferBytes *= formatBytes( stream_.userFormat );
6839 LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6840 long nextReadPointer = handle->bufferPointer[1];
6841 DWORD dsBufferSize = handle->dsBufferSize[1];
6843 // Find out where the write and "safe read" pointers are.
6844 result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer );
6845 if ( FAILED( result ) ) {
6846 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
6847 errorText_ = errorStream_.str();
6848 MUTEX_UNLOCK( &stream_.mutex );
6849 error( RtAudioError::SYSTEM_ERROR );
6853 if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset
6854 DWORD endRead = nextReadPointer + bufferBytes;
6856 // Handling depends on whether we are INPUT or DUPLEX.
6857 // If we're in INPUT mode then waiting is a good thing. If we're in DUPLEX mode,
6858 // then a wait here will drag the write pointers into the forbidden zone.
6860 // In DUPLEX mode, rather than wait, we will back off the read pointer until
6861 // it's in a safe position. This causes dropouts, but it seems to be the only
6862 // practical way to sync up the read and write pointers reliably, given the
6863 // the very complex relationship between phase and increment of the read and write
6866 // In order to minimize audible dropouts in DUPLEX mode, we will
6867 // provide a pre-roll period of 0.5 seconds in which we return
6868 // zeros from the read buffer while the pointers sync up.
6870 if ( stream_.mode == DUPLEX ) {
6871 if ( safeReadPointer < endRead ) {
6872 if ( duplexPrerollBytes <= 0 ) {
6873 // Pre-roll time over. Be more agressive.
6874 int adjustment = endRead-safeReadPointer;
6876 handle->xrun[1] = true;
6878 // - large adjustments: we've probably run out of CPU cycles, so just resync exactly,
6879 // and perform fine adjustments later.
6880 // - small adjustments: back off by twice as much.
6881 if ( adjustment >= 2*bufferBytes )
6882 nextReadPointer = safeReadPointer-2*bufferBytes;
6884 nextReadPointer = safeReadPointer-bufferBytes-adjustment;
6886 if ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;
6890 // In pre=roll time. Just do it.
6891 nextReadPointer = safeReadPointer - bufferBytes;
6892 while ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;
6894 endRead = nextReadPointer + bufferBytes;
6897 else { // mode == INPUT
6898 while ( safeReadPointer < endRead && stream_.callbackInfo.isRunning ) {
6899 // See comments for playback.
6900 double millis = (endRead - safeReadPointer) * 1000.0;
6901 millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate);
6902 if ( millis < 1.0 ) millis = 1.0;
6903 Sleep( (DWORD) millis );
6905 // Wake up and find out where we are now.
6906 result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer );
6907 if ( FAILED( result ) ) {
6908 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
6909 errorText_ = errorStream_.str();
6910 MUTEX_UNLOCK( &stream_.mutex );
6911 error( RtAudioError::SYSTEM_ERROR );
6915 if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset
6919 // Lock free space in the buffer
6920 result = dsBuffer->Lock( nextReadPointer, bufferBytes, &buffer1,
6921 &bufferSize1, &buffer2, &bufferSize2, 0 );
6922 if ( FAILED( result ) ) {
6923 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking capture buffer!";
6924 errorText_ = errorStream_.str();
6925 MUTEX_UNLOCK( &stream_.mutex );
6926 error( RtAudioError::SYSTEM_ERROR );
6930 if ( duplexPrerollBytes <= 0 ) {
6931 // Copy our buffer into the DS buffer
6932 CopyMemory( buffer, buffer1, bufferSize1 );
6933 if ( buffer2 != NULL ) CopyMemory( buffer+bufferSize1, buffer2, bufferSize2 );
6936 memset( buffer, 0, bufferSize1 );
6937 if ( buffer2 != NULL ) memset( buffer + bufferSize1, 0, bufferSize2 );
6938 duplexPrerollBytes -= bufferSize1 + bufferSize2;
6941 // Update our buffer offset and unlock sound buffer
6942 nextReadPointer = ( nextReadPointer + bufferSize1 + bufferSize2 ) % dsBufferSize;
6943 dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
6944 if ( FAILED( result ) ) {
6945 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking capture buffer!";
6946 errorText_ = errorStream_.str();
6947 MUTEX_UNLOCK( &stream_.mutex );
6948 error( RtAudioError::SYSTEM_ERROR );
6951 handle->bufferPointer[1] = nextReadPointer;
6953 // No byte swapping necessary in DirectSound implementation.
6955 // If necessary, convert 8-bit data from unsigned to signed.
6956 if ( stream_.deviceFormat[1] == RTAUDIO_SINT8 )
6957 for ( int j=0; j<bufferBytes; j++ ) buffer[j] = (signed char) ( buffer[j] - 128 );
6959 // Do buffer conversion if necessary.
6960 if ( stream_.doConvertBuffer[1] )
6961 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
6965 MUTEX_UNLOCK( &stream_.mutex );
6966 RtApi::tickStreamTime();
6969 // Definitions for utility functions and callbacks
6970 // specific to the DirectSound implementation.
6972 static unsigned __stdcall callbackHandler( void *ptr )
6974 CallbackInfo *info = (CallbackInfo *) ptr;
6975 RtApiDs *object = (RtApiDs *) info->object;
6976 bool* isRunning = &info->isRunning;
6978 while ( *isRunning == true ) {
6979 object->callbackEvent();
6986 static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
6987 LPCTSTR description,
6991 struct DsProbeData& probeInfo = *(struct DsProbeData*) lpContext;
6992 std::vector<struct DsDevice>& dsDevices = *probeInfo.dsDevices;
6995 bool validDevice = false;
6996 if ( probeInfo.isInput == true ) {
6998 LPDIRECTSOUNDCAPTURE object;
7000 hr = DirectSoundCaptureCreate( lpguid, &object, NULL );
7001 if ( hr != DS_OK ) return TRUE;
7003 caps.dwSize = sizeof(caps);
7004 hr = object->GetCaps( &caps );
7005 if ( hr == DS_OK ) {
7006 if ( caps.dwChannels > 0 && caps.dwFormats > 0 )
7013 LPDIRECTSOUND object;
7014 hr = DirectSoundCreate( lpguid, &object, NULL );
7015 if ( hr != DS_OK ) return TRUE;
7017 caps.dwSize = sizeof(caps);
7018 hr = object->GetCaps( &caps );
7019 if ( hr == DS_OK ) {
7020 if ( caps.dwFlags & DSCAPS_PRIMARYMONO || caps.dwFlags & DSCAPS_PRIMARYSTEREO )
7026 // If good device, then save its name and guid.
7027 std::string name = convertCharPointerToStdString( description );
7028 //if ( name == "Primary Sound Driver" || name == "Primary Sound Capture Driver" )
7029 if ( lpguid == NULL )
7030 name = "Default Device";
7031 if ( validDevice ) {
7032 for ( unsigned int i=0; i<dsDevices.size(); i++ ) {
7033 if ( dsDevices[i].name == name ) {
7034 dsDevices[i].found = true;
7035 if ( probeInfo.isInput ) {
7036 dsDevices[i].id[1] = lpguid;
7037 dsDevices[i].validId[1] = true;
7040 dsDevices[i].id[0] = lpguid;
7041 dsDevices[i].validId[0] = true;
7049 device.found = true;
7050 if ( probeInfo.isInput ) {
7051 device.id[1] = lpguid;
7052 device.validId[1] = true;
7055 device.id[0] = lpguid;
7056 device.validId[0] = true;
7058 dsDevices.push_back( device );
7064 static const char* getErrorString( int code )
7068 case DSERR_ALLOCATED:
7069 return "Already allocated";
7071 case DSERR_CONTROLUNAVAIL:
7072 return "Control unavailable";
7074 case DSERR_INVALIDPARAM:
7075 return "Invalid parameter";
7077 case DSERR_INVALIDCALL:
7078 return "Invalid call";
7081 return "Generic error";
7083 case DSERR_PRIOLEVELNEEDED:
7084 return "Priority level needed";
7086 case DSERR_OUTOFMEMORY:
7087 return "Out of memory";
7089 case DSERR_BADFORMAT:
7090 return "The sample rate or the channel format is not supported";
7092 case DSERR_UNSUPPORTED:
7093 return "Not supported";
7095 case DSERR_NODRIVER:
7098 case DSERR_ALREADYINITIALIZED:
7099 return "Already initialized";
7101 case DSERR_NOAGGREGATION:
7102 return "No aggregation";
7104 case DSERR_BUFFERLOST:
7105 return "Buffer lost";
7107 case DSERR_OTHERAPPHASPRIO:
7108 return "Another application already has priority";
7110 case DSERR_UNINITIALIZED:
7111 return "Uninitialized";
7114 return "DirectSound unknown error";
7117 //******************** End of __WINDOWS_DS__ *********************//
7121 #if defined(__LINUX_ALSA__)
7123 #include <alsa/asoundlib.h>
7126 // A structure to hold various information related to the ALSA API
7129 snd_pcm_t *handles[2];
7132 pthread_cond_t runnable_cv;
7136 :synchronized(false), runnable(false) { xrun[0] = false; xrun[1] = false; }
7139 static void *alsaCallbackHandler( void * ptr );
7141 RtApiAlsa :: RtApiAlsa()
7143 // Nothing to do here.
7146 RtApiAlsa :: ~RtApiAlsa()
7148 if ( stream_.state != STREAM_CLOSED ) closeStream();
7151 unsigned int RtApiAlsa :: getDeviceCount( void )
7153 unsigned nDevices = 0;
7154 int result, subdevice, card;
7158 // Count cards and devices
7160 snd_card_next( &card );
7161 while ( card >= 0 ) {
7162 sprintf( name, "hw:%d", card );
7163 result = snd_ctl_open( &handle, name, 0 );
7165 errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << ".";
7166 errorText_ = errorStream_.str();
7167 error( RtAudioError::WARNING );
7172 result = snd_ctl_pcm_next_device( handle, &subdevice );
7174 errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
7175 errorText_ = errorStream_.str();
7176 error( RtAudioError::WARNING );
7179 if ( subdevice < 0 )
7184 snd_ctl_close( handle );
7185 snd_card_next( &card );
7188 result = snd_ctl_open( &handle, "default", 0 );
7191 snd_ctl_close( handle );
7197 RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device )
7199 RtAudio::DeviceInfo info;
7200 info.probed = false;
7202 unsigned nDevices = 0;
7203 int result, subdevice, card;
7207 // Count cards and devices
7210 snd_card_next( &card );
7211 while ( card >= 0 ) {
7212 sprintf( name, "hw:%d", card );
7213 result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
7215 errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << ".";
7216 errorText_ = errorStream_.str();
7217 error( RtAudioError::WARNING );
7222 result = snd_ctl_pcm_next_device( chandle, &subdevice );
7224 errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
7225 errorText_ = errorStream_.str();
7226 error( RtAudioError::WARNING );
7229 if ( subdevice < 0 ) break;
7230 if ( nDevices == device ) {
7231 sprintf( name, "hw:%d,%d", card, subdevice );
7237 snd_ctl_close( chandle );
7238 snd_card_next( &card );
7241 result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
7242 if ( result == 0 ) {
7243 if ( nDevices == device ) {
7244 strcpy( name, "default" );
7250 if ( nDevices == 0 ) {
7251 errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!";
7252 error( RtAudioError::INVALID_USE );
7256 if ( device >= nDevices ) {
7257 errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!";
7258 error( RtAudioError::INVALID_USE );
7264 // If a stream is already open, we cannot probe the stream devices.
7265 // Thus, use the saved results.
7266 if ( stream_.state != STREAM_CLOSED &&
7267 ( stream_.device[0] == device || stream_.device[1] == device ) ) {
7268 snd_ctl_close( chandle );
7269 if ( device >= devices_.size() ) {
7270 errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened.";
7271 error( RtAudioError::WARNING );
7274 return devices_[ device ];
7277 int openMode = SND_PCM_ASYNC;
7278 snd_pcm_stream_t stream;
7279 snd_pcm_info_t *pcminfo;
7280 snd_pcm_info_alloca( &pcminfo );
7282 snd_pcm_hw_params_t *params;
7283 snd_pcm_hw_params_alloca( ¶ms );
7285 // First try for playback unless default device (which has subdev -1)
7286 stream = SND_PCM_STREAM_PLAYBACK;
7287 snd_pcm_info_set_stream( pcminfo, stream );
7288 if ( subdevice != -1 ) {
7289 snd_pcm_info_set_device( pcminfo, subdevice );
7290 snd_pcm_info_set_subdevice( pcminfo, 0 );
7292 result = snd_ctl_pcm_info( chandle, pcminfo );
7294 // Device probably doesn't support playback.
7299 result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK );
7301 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
7302 errorText_ = errorStream_.str();
7303 error( RtAudioError::WARNING );
7307 // The device is open ... fill the parameter structure.
7308 result = snd_pcm_hw_params_any( phandle, params );
7310 snd_pcm_close( phandle );
7311 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
7312 errorText_ = errorStream_.str();
7313 error( RtAudioError::WARNING );
7317 // Get output channel information.
7319 result = snd_pcm_hw_params_get_channels_max( params, &value );
7321 snd_pcm_close( phandle );
7322 errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << ".";
7323 errorText_ = errorStream_.str();
7324 error( RtAudioError::WARNING );
7327 info.outputChannels = value;
7328 snd_pcm_close( phandle );
7331 stream = SND_PCM_STREAM_CAPTURE;
7332 snd_pcm_info_set_stream( pcminfo, stream );
7334 // Now try for capture unless default device (with subdev = -1)
7335 if ( subdevice != -1 ) {
7336 result = snd_ctl_pcm_info( chandle, pcminfo );
7337 snd_ctl_close( chandle );
7339 // Device probably doesn't support capture.
7340 if ( info.outputChannels == 0 ) return info;
7341 goto probeParameters;
7345 snd_ctl_close( chandle );
7347 result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
7349 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
7350 errorText_ = errorStream_.str();
7351 error( RtAudioError::WARNING );
7352 if ( info.outputChannels == 0 ) return info;
7353 goto probeParameters;
7356 // The device is open ... fill the parameter structure.
7357 result = snd_pcm_hw_params_any( phandle, params );
7359 snd_pcm_close( phandle );
7360 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
7361 errorText_ = errorStream_.str();
7362 error( RtAudioError::WARNING );
7363 if ( info.outputChannels == 0 ) return info;
7364 goto probeParameters;
7367 result = snd_pcm_hw_params_get_channels_max( params, &value );
7369 snd_pcm_close( phandle );
7370 errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << ".";
7371 errorText_ = errorStream_.str();
7372 error( RtAudioError::WARNING );
7373 if ( info.outputChannels == 0 ) return info;
7374 goto probeParameters;
7376 info.inputChannels = value;
7377 snd_pcm_close( phandle );
7379 // If device opens for both playback and capture, we determine the channels.
7380 if ( info.outputChannels > 0 && info.inputChannels > 0 )
7381 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
7383 // ALSA doesn't provide default devices so we'll use the first available one.
7384 if ( device == 0 && info.outputChannels > 0 )
7385 info.isDefaultOutput = true;
7386 if ( device == 0 && info.inputChannels > 0 )
7387 info.isDefaultInput = true;
7390 // At this point, we just need to figure out the supported data
7391 // formats and sample rates. We'll proceed by opening the device in
7392 // the direction with the maximum number of channels, or playback if
7393 // they are equal. This might limit our sample rate options, but so
7396 if ( info.outputChannels >= info.inputChannels )
7397 stream = SND_PCM_STREAM_PLAYBACK;
7399 stream = SND_PCM_STREAM_CAPTURE;
7400 snd_pcm_info_set_stream( pcminfo, stream );
7402 result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
7404 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
7405 errorText_ = errorStream_.str();
7406 error( RtAudioError::WARNING );
7410 // The device is open ... fill the parameter structure.
7411 result = snd_pcm_hw_params_any( phandle, params );
7413 snd_pcm_close( phandle );
7414 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
7415 errorText_ = errorStream_.str();
7416 error( RtAudioError::WARNING );
7420 // Test our discrete set of sample rate values.
7421 info.sampleRates.clear();
7422 for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {
7423 if ( snd_pcm_hw_params_test_rate( phandle, params, SAMPLE_RATES[i], 0 ) == 0 ) {
7424 info.sampleRates.push_back( SAMPLE_RATES[i] );
7426 if ( !info.preferredSampleRate || ( SAMPLE_RATES[i] <= 48000 && SAMPLE_RATES[i] > info.preferredSampleRate ) )
7427 info.preferredSampleRate = SAMPLE_RATES[i];
7430 if ( info.sampleRates.size() == 0 ) {
7431 snd_pcm_close( phandle );
7432 errorStream_ << "RtApiAlsa::getDeviceInfo: no supported sample rates found for device (" << name << ").";
7433 errorText_ = errorStream_.str();
7434 error( RtAudioError::WARNING );
7438 // Probe the supported data formats ... we don't care about endian-ness just yet
7439 snd_pcm_format_t format;
7440 info.nativeFormats = 0;
7441 format = SND_PCM_FORMAT_S8;
7442 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7443 info.nativeFormats |= RTAUDIO_SINT8;
7444 format = SND_PCM_FORMAT_S16;
7445 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7446 info.nativeFormats |= RTAUDIO_SINT16;
7447 format = SND_PCM_FORMAT_S24;
7448 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7449 info.nativeFormats |= RTAUDIO_SINT24;
7450 format = SND_PCM_FORMAT_S32;
7451 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7452 info.nativeFormats |= RTAUDIO_SINT32;
7453 format = SND_PCM_FORMAT_FLOAT;
7454 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7455 info.nativeFormats |= RTAUDIO_FLOAT32;
7456 format = SND_PCM_FORMAT_FLOAT64;
7457 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7458 info.nativeFormats |= RTAUDIO_FLOAT64;
7460 // Check that we have at least one supported format
7461 if ( info.nativeFormats == 0 ) {
7462 snd_pcm_close( phandle );
7463 errorStream_ << "RtApiAlsa::getDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio.";
7464 errorText_ = errorStream_.str();
7465 error( RtAudioError::WARNING );
7469 // Get the device name
7471 result = snd_card_get_name( card, &cardname );
7472 if ( result >= 0 ) {
7473 sprintf( name, "hw:%s,%d", cardname, subdevice );
7478 // That's all ... close the device and return
7479 snd_pcm_close( phandle );
7484 void RtApiAlsa :: saveDeviceInfo( void )
7488 unsigned int nDevices = getDeviceCount();
7489 devices_.resize( nDevices );
7490 for ( unsigned int i=0; i<nDevices; i++ )
7491 devices_[i] = getDeviceInfo( i );
7494 bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
7495 unsigned int firstChannel, unsigned int sampleRate,
7496 RtAudioFormat format, unsigned int *bufferSize,
7497 RtAudio::StreamOptions *options )
7500 #if defined(__RTAUDIO_DEBUG__)
7502 snd_output_stdio_attach(&out, stderr, 0);
7505 // I'm not using the "plug" interface ... too much inconsistent behavior.
7507 unsigned nDevices = 0;
7508 int result, subdevice, card;
7512 if ( options && options->flags & RTAUDIO_ALSA_USE_DEFAULT )
7513 snprintf(name, sizeof(name), "%s", "default");
7515 // Count cards and devices
7517 snd_card_next( &card );
7518 while ( card >= 0 ) {
7519 sprintf( name, "hw:%d", card );
7520 result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
7522 errorStream_ << "RtApiAlsa::probeDeviceOpen: control open, card = " << card << ", " << snd_strerror( result ) << ".";
7523 errorText_ = errorStream_.str();
7528 result = snd_ctl_pcm_next_device( chandle, &subdevice );
7529 if ( result < 0 ) break;
7530 if ( subdevice < 0 ) break;
7531 if ( nDevices == device ) {
7532 sprintf( name, "hw:%d,%d", card, subdevice );
7533 snd_ctl_close( chandle );
7538 snd_ctl_close( chandle );
7539 snd_card_next( &card );
7542 result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
7543 if ( result == 0 ) {
7544 if ( nDevices == device ) {
7545 strcpy( name, "default" );
7546 snd_ctl_close( chandle );
7551 snd_ctl_close( chandle );
7553 if ( nDevices == 0 ) {
7554 // This should not happen because a check is made before this function is called.
7555 errorText_ = "RtApiAlsa::probeDeviceOpen: no devices found!";
7559 if ( device >= nDevices ) {
7560 // This should not happen because a check is made before this function is called.
7561 errorText_ = "RtApiAlsa::probeDeviceOpen: device ID is invalid!";
7568 // The getDeviceInfo() function will not work for a device that is
7569 // already open. Thus, we'll probe the system before opening a
7570 // stream and save the results for use by getDeviceInfo().
7571 if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) // only do once
7572 this->saveDeviceInfo();
7574 snd_pcm_stream_t stream;
7575 if ( mode == OUTPUT )
7576 stream = SND_PCM_STREAM_PLAYBACK;
7578 stream = SND_PCM_STREAM_CAPTURE;
7581 int openMode = SND_PCM_ASYNC;
7582 result = snd_pcm_open( &phandle, name, stream, openMode );
7584 if ( mode == OUTPUT )
7585 errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for output.";
7587 errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for input.";
7588 errorText_ = errorStream_.str();
7592 // Fill the parameter structure.
7593 snd_pcm_hw_params_t *hw_params;
7594 snd_pcm_hw_params_alloca( &hw_params );
7595 result = snd_pcm_hw_params_any( phandle, hw_params );
7597 snd_pcm_close( phandle );
7598 errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") parameters, " << snd_strerror( result ) << ".";
7599 errorText_ = errorStream_.str();
7603 #if defined(__RTAUDIO_DEBUG__)
7604 fprintf( stderr, "\nRtApiAlsa: dump hardware params just after device open:\n\n" );
7605 snd_pcm_hw_params_dump( hw_params, out );
7608 // Set access ... check user preference.
7609 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) {
7610 stream_.userInterleaved = false;
7611 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
7613 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
7614 stream_.deviceInterleaved[mode] = true;
7617 stream_.deviceInterleaved[mode] = false;
7620 stream_.userInterleaved = true;
7621 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
7623 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
7624 stream_.deviceInterleaved[mode] = false;
7627 stream_.deviceInterleaved[mode] = true;
7631 snd_pcm_close( phandle );
7632 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") access, " << snd_strerror( result ) << ".";
7633 errorText_ = errorStream_.str();
7637 // Determine how to set the device format.
7638 stream_.userFormat = format;
7639 snd_pcm_format_t deviceFormat = SND_PCM_FORMAT_UNKNOWN;
7641 if ( format == RTAUDIO_SINT8 )
7642 deviceFormat = SND_PCM_FORMAT_S8;
7643 else if ( format == RTAUDIO_SINT16 )
7644 deviceFormat = SND_PCM_FORMAT_S16;
7645 else if ( format == RTAUDIO_SINT24 )
7646 deviceFormat = SND_PCM_FORMAT_S24;
7647 else if ( format == RTAUDIO_SINT32 )
7648 deviceFormat = SND_PCM_FORMAT_S32;
7649 else if ( format == RTAUDIO_FLOAT32 )
7650 deviceFormat = SND_PCM_FORMAT_FLOAT;
7651 else if ( format == RTAUDIO_FLOAT64 )
7652 deviceFormat = SND_PCM_FORMAT_FLOAT64;
7654 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat) == 0) {
7655 stream_.deviceFormat[mode] = format;
7659 // The user requested format is not natively supported by the device.
7660 deviceFormat = SND_PCM_FORMAT_FLOAT64;
7661 if ( snd_pcm_hw_params_test_format( phandle, hw_params, deviceFormat ) == 0 ) {
7662 stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
7666 deviceFormat = SND_PCM_FORMAT_FLOAT;
7667 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7668 stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
7672 deviceFormat = SND_PCM_FORMAT_S32;
7673 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7674 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
7678 deviceFormat = SND_PCM_FORMAT_S24;
7679 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7680 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
7684 deviceFormat = SND_PCM_FORMAT_S16;
7685 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7686 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
7690 deviceFormat = SND_PCM_FORMAT_S8;
7691 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7692 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
7696 // If we get here, no supported format was found.
7697 snd_pcm_close( phandle );
7698 errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device " << device << " data format not supported by RtAudio.";
7699 errorText_ = errorStream_.str();
7703 result = snd_pcm_hw_params_set_format( phandle, hw_params, deviceFormat );
7705 snd_pcm_close( phandle );
7706 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") data format, " << snd_strerror( result ) << ".";
7707 errorText_ = errorStream_.str();
7711 // Determine whether byte-swaping is necessary.
7712 stream_.doByteSwap[mode] = false;
7713 if ( deviceFormat != SND_PCM_FORMAT_S8 ) {
7714 result = snd_pcm_format_cpu_endian( deviceFormat );
7716 stream_.doByteSwap[mode] = true;
7717 else if (result < 0) {
7718 snd_pcm_close( phandle );
7719 errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") endian-ness, " << snd_strerror( result ) << ".";
7720 errorText_ = errorStream_.str();
7725 // Set the sample rate.
7726 result = snd_pcm_hw_params_set_rate_near( phandle, hw_params, (unsigned int*) &sampleRate, 0 );
7728 snd_pcm_close( phandle );
7729 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting sample rate on device (" << name << "), " << snd_strerror( result ) << ".";
7730 errorText_ = errorStream_.str();
7734 // Determine the number of channels for this device. We support a possible
7735 // minimum device channel number > than the value requested by the user.
7736 stream_.nUserChannels[mode] = channels;
7738 result = snd_pcm_hw_params_get_channels_max( hw_params, &value );
7739 unsigned int deviceChannels = value;
7740 if ( result < 0 || deviceChannels < channels + firstChannel ) {
7741 snd_pcm_close( phandle );
7742 errorStream_ << "RtApiAlsa::probeDeviceOpen: requested channel parameters not supported by device (" << name << "), " << snd_strerror( result ) << ".";
7743 errorText_ = errorStream_.str();
7747 result = snd_pcm_hw_params_get_channels_min( hw_params, &value );
7749 snd_pcm_close( phandle );
7750 errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting minimum channels for device (" << name << "), " << snd_strerror( result ) << ".";
7751 errorText_ = errorStream_.str();
7754 deviceChannels = value;
7755 if ( deviceChannels < channels + firstChannel ) deviceChannels = channels + firstChannel;
7756 stream_.nDeviceChannels[mode] = deviceChannels;
7758 // Set the device channels.
7759 result = snd_pcm_hw_params_set_channels( phandle, hw_params, deviceChannels );
7761 snd_pcm_close( phandle );
7762 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting channels for device (" << name << "), " << snd_strerror( result ) << ".";
7763 errorText_ = errorStream_.str();
7767 // Set the buffer (or period) size.
7769 snd_pcm_uframes_t periodSize = *bufferSize;
7770 result = snd_pcm_hw_params_set_period_size_near( phandle, hw_params, &periodSize, &dir );
7772 snd_pcm_close( phandle );
7773 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting period size for device (" << name << "), " << snd_strerror( result ) << ".";
7774 errorText_ = errorStream_.str();
7777 *bufferSize = periodSize;
7779 // Set the buffer number, which in ALSA is referred to as the "period".
7780 unsigned int periods = 0;
7781 if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2;
7782 if ( options && options->numberOfBuffers > 0 ) periods = options->numberOfBuffers;
7783 if ( periods < 2 ) periods = 4; // a fairly safe default value
7784 result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir );
7786 snd_pcm_close( phandle );
7787 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << ".";
7788 errorText_ = errorStream_.str();
7792 // If attempting to setup a duplex stream, the bufferSize parameter
7793 // MUST be the same in both directions!
7794 if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {
7795 snd_pcm_close( phandle );
7796 errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ").";
7797 errorText_ = errorStream_.str();
7801 stream_.bufferSize = *bufferSize;
7803 // Install the hardware configuration
7804 result = snd_pcm_hw_params( phandle, hw_params );
7806 snd_pcm_close( phandle );
7807 errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing hardware configuration on device (" << name << "), " << snd_strerror( result ) << ".";
7808 errorText_ = errorStream_.str();
7812 #if defined(__RTAUDIO_DEBUG__)
7813 fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n");
7814 snd_pcm_hw_params_dump( hw_params, out );
7817 // Set the software configuration to fill buffers with zeros and prevent device stopping on xruns.
7818 snd_pcm_sw_params_t *sw_params = NULL;
7819 snd_pcm_sw_params_alloca( &sw_params );
7820 snd_pcm_sw_params_current( phandle, sw_params );
7821 snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize );
7822 snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, ULONG_MAX );
7823 snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 );
7825 // The following two settings were suggested by Theo Veenker
7826 //snd_pcm_sw_params_set_avail_min( phandle, sw_params, *bufferSize );
7827 //snd_pcm_sw_params_set_xfer_align( phandle, sw_params, 1 );
7829 // here are two options for a fix
7830 //snd_pcm_sw_params_set_silence_size( phandle, sw_params, ULONG_MAX );
7831 snd_pcm_uframes_t val;
7832 snd_pcm_sw_params_get_boundary( sw_params, &val );
7833 snd_pcm_sw_params_set_silence_size( phandle, sw_params, val );
7835 result = snd_pcm_sw_params( phandle, sw_params );
7837 snd_pcm_close( phandle );
7838 errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing software configuration on device (" << name << "), " << snd_strerror( result ) << ".";
7839 errorText_ = errorStream_.str();
7843 #if defined(__RTAUDIO_DEBUG__)
7844 fprintf(stderr, "\nRtApiAlsa: dump software params after installation:\n\n");
7845 snd_pcm_sw_params_dump( sw_params, out );
7848 // Set flags for buffer conversion
7849 stream_.doConvertBuffer[mode] = false;
7850 if ( stream_.userFormat != stream_.deviceFormat[mode] )
7851 stream_.doConvertBuffer[mode] = true;
7852 if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
7853 stream_.doConvertBuffer[mode] = true;
7854 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
7855 stream_.nUserChannels[mode] > 1 )
7856 stream_.doConvertBuffer[mode] = true;
7858 // Allocate the ApiHandle if necessary and then save.
7859 AlsaHandle *apiInfo = 0;
7860 if ( stream_.apiHandle == 0 ) {
7862 apiInfo = (AlsaHandle *) new AlsaHandle;
7864 catch ( std::bad_alloc& ) {
7865 errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating AlsaHandle memory.";
7869 if ( pthread_cond_init( &apiInfo->runnable_cv, NULL ) ) {
7870 errorText_ = "RtApiAlsa::probeDeviceOpen: error initializing pthread condition variable.";
7874 stream_.apiHandle = (void *) apiInfo;
7875 apiInfo->handles[0] = 0;
7876 apiInfo->handles[1] = 0;
7879 apiInfo = (AlsaHandle *) stream_.apiHandle;
7881 apiInfo->handles[mode] = phandle;
7884 // Allocate necessary internal buffers.
7885 unsigned long bufferBytes;
7886 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
7887 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
7888 if ( stream_.userBuffer[mode] == NULL ) {
7889 errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating user buffer memory.";
7893 if ( stream_.doConvertBuffer[mode] ) {
7895 bool makeBuffer = true;
7896 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
7897 if ( mode == INPUT ) {
7898 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
7899 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
7900 if ( bufferBytes <= bytesOut ) makeBuffer = false;
7905 bufferBytes *= *bufferSize;
7906 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
7907 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
7908 if ( stream_.deviceBuffer == NULL ) {
7909 errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating device buffer memory.";
7915 stream_.sampleRate = sampleRate;
7916 stream_.nBuffers = periods;
7917 stream_.device[mode] = device;
7918 stream_.state = STREAM_STOPPED;
7920 // Setup the buffer conversion information structure.
7921 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
7923 // Setup thread if necessary.
7924 if ( stream_.mode == OUTPUT && mode == INPUT ) {
7925 // We had already set up an output stream.
7926 stream_.mode = DUPLEX;
7927 // Link the streams if possible.
7928 apiInfo->synchronized = false;
7929 if ( snd_pcm_link( apiInfo->handles[0], apiInfo->handles[1] ) == 0 )
7930 apiInfo->synchronized = true;
7932 errorText_ = "RtApiAlsa::probeDeviceOpen: unable to synchronize input and output devices.";
7933 error( RtAudioError::WARNING );
7937 stream_.mode = mode;
7939 // Setup callback thread.
7940 stream_.callbackInfo.object = (void *) this;
7942 // Set the thread attributes for joinable and realtime scheduling
7943 // priority (optional). The higher priority will only take affect
7944 // if the program is run as root or suid. Note, under Linux
7945 // processes with CAP_SYS_NICE privilege, a user can change
7946 // scheduling policy and priority (thus need not be root). See
7947 // POSIX "capabilities".
7948 pthread_attr_t attr;
7949 pthread_attr_init( &attr );
7950 pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
7951 #ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
7952 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
7953 stream_.callbackInfo.doRealtime = true;
7954 struct sched_param param;
7955 int priority = options->priority;
7956 int min = sched_get_priority_min( SCHED_RR );
7957 int max = sched_get_priority_max( SCHED_RR );
7958 if ( priority < min ) priority = min;
7959 else if ( priority > max ) priority = max;
7960 param.sched_priority = priority;
7962 // Set the policy BEFORE the priority. Otherwise it fails.
7963 pthread_attr_setschedpolicy(&attr, SCHED_RR);
7964 pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM);
7965 // This is definitely required. Otherwise it fails.
7966 pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED);
7967 pthread_attr_setschedparam(&attr, ¶m);
7970 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
7972 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
7975 stream_.callbackInfo.isRunning = true;
7976 result = pthread_create( &stream_.callbackInfo.thread, &attr, alsaCallbackHandler, &stream_.callbackInfo );
7977 pthread_attr_destroy( &attr );
7979 // Failed. Try instead with default attributes.
7980 result = pthread_create( &stream_.callbackInfo.thread, NULL, alsaCallbackHandler, &stream_.callbackInfo );
7982 stream_.callbackInfo.isRunning = false;
7983 errorText_ = "RtApiAlsa::error creating callback thread!";
7993 pthread_cond_destroy( &apiInfo->runnable_cv );
7994 if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
7995 if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
7997 stream_.apiHandle = 0;
8000 if ( phandle) snd_pcm_close( phandle );
8002 for ( int i=0; i<2; i++ ) {
8003 if ( stream_.userBuffer[i] ) {
8004 free( stream_.userBuffer[i] );
8005 stream_.userBuffer[i] = 0;
8009 if ( stream_.deviceBuffer ) {
8010 free( stream_.deviceBuffer );
8011 stream_.deviceBuffer = 0;
8014 stream_.state = STREAM_CLOSED;
8018 void RtApiAlsa :: closeStream()
8020 if ( stream_.state == STREAM_CLOSED ) {
8021 errorText_ = "RtApiAlsa::closeStream(): no open stream to close!";
8022 error( RtAudioError::WARNING );
8026 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
8027 stream_.callbackInfo.isRunning = false;
8028 MUTEX_LOCK( &stream_.mutex );
8029 if ( stream_.state == STREAM_STOPPED ) {
8030 apiInfo->runnable = true;
8031 pthread_cond_signal( &apiInfo->runnable_cv );
8033 MUTEX_UNLOCK( &stream_.mutex );
8034 pthread_join( stream_.callbackInfo.thread, NULL );
8036 if ( stream_.state == STREAM_RUNNING ) {
8037 stream_.state = STREAM_STOPPED;
8038 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
8039 snd_pcm_drop( apiInfo->handles[0] );
8040 if ( stream_.mode == INPUT || stream_.mode == DUPLEX )
8041 snd_pcm_drop( apiInfo->handles[1] );
8045 pthread_cond_destroy( &apiInfo->runnable_cv );
8046 if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
8047 if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
8049 stream_.apiHandle = 0;
8052 for ( int i=0; i<2; i++ ) {
8053 if ( stream_.userBuffer[i] ) {
8054 free( stream_.userBuffer[i] );
8055 stream_.userBuffer[i] = 0;
8059 if ( stream_.deviceBuffer ) {
8060 free( stream_.deviceBuffer );
8061 stream_.deviceBuffer = 0;
8064 stream_.mode = UNINITIALIZED;
8065 stream_.state = STREAM_CLOSED;
8068 void RtApiAlsa :: startStream()
8070 // This method calls snd_pcm_prepare if the device isn't already in that state.
8073 if ( stream_.state == STREAM_RUNNING ) {
8074 errorText_ = "RtApiAlsa::startStream(): the stream is already running!";
8075 error( RtAudioError::WARNING );
8079 MUTEX_LOCK( &stream_.mutex );
8082 snd_pcm_state_t state;
8083 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
8084 snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
8085 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
8086 state = snd_pcm_state( handle[0] );
8087 if ( state != SND_PCM_STATE_PREPARED ) {
8088 result = snd_pcm_prepare( handle[0] );
8090 errorStream_ << "RtApiAlsa::startStream: error preparing output pcm device, " << snd_strerror( result ) << ".";
8091 errorText_ = errorStream_.str();
8097 if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
8098 result = snd_pcm_drop(handle[1]); // fix to remove stale data received since device has been open
8099 state = snd_pcm_state( handle[1] );
8100 if ( state != SND_PCM_STATE_PREPARED ) {
8101 result = snd_pcm_prepare( handle[1] );
8103 errorStream_ << "RtApiAlsa::startStream: error preparing input pcm device, " << snd_strerror( result ) << ".";
8104 errorText_ = errorStream_.str();
8110 stream_.state = STREAM_RUNNING;
8113 apiInfo->runnable = true;
8114 pthread_cond_signal( &apiInfo->runnable_cv );
8115 MUTEX_UNLOCK( &stream_.mutex );
8117 if ( result >= 0 ) return;
8118 error( RtAudioError::SYSTEM_ERROR );
8121 void RtApiAlsa :: stopStream()
8124 if ( stream_.state == STREAM_STOPPED ) {
8125 errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!";
8126 error( RtAudioError::WARNING );
8130 stream_.state = STREAM_STOPPED;
8131 MUTEX_LOCK( &stream_.mutex );
8134 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
8135 snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
8136 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
8137 if ( apiInfo->synchronized )
8138 result = snd_pcm_drop( handle[0] );
8140 result = snd_pcm_drain( handle[0] );
8142 errorStream_ << "RtApiAlsa::stopStream: error draining output pcm device, " << snd_strerror( result ) << ".";
8143 errorText_ = errorStream_.str();
8148 if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
8149 result = snd_pcm_drop( handle[1] );
8151 errorStream_ << "RtApiAlsa::stopStream: error stopping input pcm device, " << snd_strerror( result ) << ".";
8152 errorText_ = errorStream_.str();
8158 apiInfo->runnable = false; // fixes high CPU usage when stopped
8159 MUTEX_UNLOCK( &stream_.mutex );
8161 if ( result >= 0 ) return;
8162 error( RtAudioError::SYSTEM_ERROR );
8165 void RtApiAlsa :: abortStream()
8168 if ( stream_.state == STREAM_STOPPED ) {
8169 errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!";
8170 error( RtAudioError::WARNING );
8174 stream_.state = STREAM_STOPPED;
8175 MUTEX_LOCK( &stream_.mutex );
8178 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
8179 snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
8180 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
8181 result = snd_pcm_drop( handle[0] );
8183 errorStream_ << "RtApiAlsa::abortStream: error aborting output pcm device, " << snd_strerror( result ) << ".";
8184 errorText_ = errorStream_.str();
8189 if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
8190 result = snd_pcm_drop( handle[1] );
8192 errorStream_ << "RtApiAlsa::abortStream: error aborting input pcm device, " << snd_strerror( result ) << ".";
8193 errorText_ = errorStream_.str();
8199 apiInfo->runnable = false; // fixes high CPU usage when stopped
8200 MUTEX_UNLOCK( &stream_.mutex );
8202 if ( result >= 0 ) return;
8203 error( RtAudioError::SYSTEM_ERROR );
8206 void RtApiAlsa :: callbackEvent()
8208 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
8209 if ( stream_.state == STREAM_STOPPED ) {
8210 MUTEX_LOCK( &stream_.mutex );
8211 while ( !apiInfo->runnable )
8212 pthread_cond_wait( &apiInfo->runnable_cv, &stream_.mutex );
8214 if ( stream_.state != STREAM_RUNNING ) {
8215 MUTEX_UNLOCK( &stream_.mutex );
8218 MUTEX_UNLOCK( &stream_.mutex );
8221 if ( stream_.state == STREAM_CLOSED ) {
8222 errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!";
8223 error( RtAudioError::WARNING );
8227 int doStopStream = 0;
8228 RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
8229 double streamTime = getStreamTime();
8230 RtAudioStreamStatus status = 0;
8231 if ( stream_.mode != INPUT && apiInfo->xrun[0] == true ) {
8232 status |= RTAUDIO_OUTPUT_UNDERFLOW;
8233 apiInfo->xrun[0] = false;
8235 if ( stream_.mode != OUTPUT && apiInfo->xrun[1] == true ) {
8236 status |= RTAUDIO_INPUT_OVERFLOW;
8237 apiInfo->xrun[1] = false;
8239 doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
8240 stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
8242 if ( doStopStream == 2 ) {
8247 MUTEX_LOCK( &stream_.mutex );
8249 // The state might change while waiting on a mutex.
8250 if ( stream_.state == STREAM_STOPPED ) goto unlock;
8256 snd_pcm_sframes_t frames;
8257 RtAudioFormat format;
8258 handle = (snd_pcm_t **) apiInfo->handles;
8260 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
8262 // Setup parameters.
8263 if ( stream_.doConvertBuffer[1] ) {
8264 buffer = stream_.deviceBuffer;
8265 channels = stream_.nDeviceChannels[1];
8266 format = stream_.deviceFormat[1];
8269 buffer = stream_.userBuffer[1];
8270 channels = stream_.nUserChannels[1];
8271 format = stream_.userFormat;
8274 // Read samples from device in interleaved/non-interleaved format.
8275 if ( stream_.deviceInterleaved[1] )
8276 result = snd_pcm_readi( handle[1], buffer, stream_.bufferSize );
8278 void *bufs[channels];
8279 size_t offset = stream_.bufferSize * formatBytes( format );
8280 for ( int i=0; i<channels; i++ )
8281 bufs[i] = (void *) (buffer + (i * offset));
8282 result = snd_pcm_readn( handle[1], bufs, stream_.bufferSize );
8285 if ( result < (int) stream_.bufferSize ) {
8286 // Either an error or overrun occured.
8287 if ( result == -EPIPE ) {
8288 snd_pcm_state_t state = snd_pcm_state( handle[1] );
8289 if ( state == SND_PCM_STATE_XRUN ) {
8290 apiInfo->xrun[1] = true;
8291 result = snd_pcm_prepare( handle[1] );
8293 errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after overrun, " << snd_strerror( result ) << ".";
8294 errorText_ = errorStream_.str();
8298 errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
8299 errorText_ = errorStream_.str();
8303 errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << ".";
8304 errorText_ = errorStream_.str();
8306 error( RtAudioError::WARNING );
8310 // Do byte swapping if necessary.
8311 if ( stream_.doByteSwap[1] )
8312 byteSwapBuffer( buffer, stream_.bufferSize * channels, format );
8314 // Do buffer conversion if necessary.
8315 if ( stream_.doConvertBuffer[1] )
8316 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
8318 // Check stream latency
8319 result = snd_pcm_delay( handle[1], &frames );
8320 if ( result == 0 && frames > 0 ) stream_.latency[1] = frames;
8325 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
8327 // Setup parameters and do buffer conversion if necessary.
8328 if ( stream_.doConvertBuffer[0] ) {
8329 buffer = stream_.deviceBuffer;
8330 convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
8331 channels = stream_.nDeviceChannels[0];
8332 format = stream_.deviceFormat[0];
8335 buffer = stream_.userBuffer[0];
8336 channels = stream_.nUserChannels[0];
8337 format = stream_.userFormat;
8340 // Do byte swapping if necessary.
8341 if ( stream_.doByteSwap[0] )
8342 byteSwapBuffer(buffer, stream_.bufferSize * channels, format);
8344 // Write samples to device in interleaved/non-interleaved format.
8345 if ( stream_.deviceInterleaved[0] )
8346 result = snd_pcm_writei( handle[0], buffer, stream_.bufferSize );
8348 void *bufs[channels];
8349 size_t offset = stream_.bufferSize * formatBytes( format );
8350 for ( int i=0; i<channels; i++ )
8351 bufs[i] = (void *) (buffer + (i * offset));
8352 result = snd_pcm_writen( handle[0], bufs, stream_.bufferSize );
8355 if ( result < (int) stream_.bufferSize ) {
8356 // Either an error or underrun occured.
8357 if ( result == -EPIPE ) {
8358 snd_pcm_state_t state = snd_pcm_state( handle[0] );
8359 if ( state == SND_PCM_STATE_XRUN ) {
8360 apiInfo->xrun[0] = true;
8361 result = snd_pcm_prepare( handle[0] );
8363 errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after underrun, " << snd_strerror( result ) << ".";
8364 errorText_ = errorStream_.str();
8367 errorText_ = "RtApiAlsa::callbackEvent: audio write error, underrun.";
8370 errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
8371 errorText_ = errorStream_.str();
8375 errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << ".";
8376 errorText_ = errorStream_.str();
8378 error( RtAudioError::WARNING );
8382 // Check stream latency
8383 result = snd_pcm_delay( handle[0], &frames );
8384 if ( result == 0 && frames > 0 ) stream_.latency[0] = frames;
8388 MUTEX_UNLOCK( &stream_.mutex );
8390 RtApi::tickStreamTime();
8391 if ( doStopStream == 1 ) this->stopStream();
8394 static void *alsaCallbackHandler( void *ptr )
8396 CallbackInfo *info = (CallbackInfo *) ptr;
8397 RtApiAlsa *object = (RtApiAlsa *) info->object;
8398 bool *isRunning = &info->isRunning;
8400 #ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
8401 if ( info->doRealtime ) {
8402 std::cerr << "RtAudio alsa: " <<
8403 (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") <<
8404 "running realtime scheduling" << std::endl;
8408 while ( *isRunning == true ) {
8409 pthread_testcancel();
8410 object->callbackEvent();
8413 pthread_exit( NULL );
8416 //******************** End of __LINUX_ALSA__ *********************//
8419 #if defined(__LINUX_PULSE__)
8421 // Code written by Peter Meerwald, pmeerw@pmeerw.net
8422 // and Tristan Matthews.
8424 #include <pulse/error.h>
8425 #include <pulse/simple.h>
8428 static const unsigned int SUPPORTED_SAMPLERATES[] = { 8000, 16000, 22050, 32000,
8429 44100, 48000, 96000, 0};
8431 struct rtaudio_pa_format_mapping_t {
8432 RtAudioFormat rtaudio_format;
8433 pa_sample_format_t pa_format;
8436 static const rtaudio_pa_format_mapping_t supported_sampleformats[] = {
8437 {RTAUDIO_SINT16, PA_SAMPLE_S16LE},
8438 {RTAUDIO_SINT32, PA_SAMPLE_S32LE},
8439 {RTAUDIO_FLOAT32, PA_SAMPLE_FLOAT32LE},
8440 {0, PA_SAMPLE_INVALID}};
8442 struct PulseAudioHandle {
8446 pthread_cond_t runnable_cv;
8448 PulseAudioHandle() : s_play(0), s_rec(0), runnable(false) { }
8451 RtApiPulse::~RtApiPulse()
8453 if ( stream_.state != STREAM_CLOSED )
8457 unsigned int RtApiPulse::getDeviceCount( void )
8462 RtAudio::DeviceInfo RtApiPulse::getDeviceInfo( unsigned int /*device*/ )
8464 RtAudio::DeviceInfo info;
8466 info.name = "PulseAudio";
8467 info.outputChannels = 2;
8468 info.inputChannels = 2;
8469 info.duplexChannels = 2;
8470 info.isDefaultOutput = true;
8471 info.isDefaultInput = true;
8473 for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr )
8474 info.sampleRates.push_back( *sr );
8476 info.preferredSampleRate = 48000;
8477 info.nativeFormats = RTAUDIO_SINT16 | RTAUDIO_SINT32 | RTAUDIO_FLOAT32;
8482 static void *pulseaudio_callback( void * user )
8484 CallbackInfo *cbi = static_cast<CallbackInfo *>( user );
8485 RtApiPulse *context = static_cast<RtApiPulse *>( cbi->object );
8486 volatile bool *isRunning = &cbi->isRunning;
8488 #ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
8489 if (cbi->doRealtime) {
8490 std::cerr << "RtAudio pulse: " <<
8491 (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") <<
8492 "running realtime scheduling" << std::endl;
8496 while ( *isRunning ) {
8497 pthread_testcancel();
8498 context->callbackEvent();
8501 pthread_exit( NULL );
8504 void RtApiPulse::closeStream( void )
8506 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8508 stream_.callbackInfo.isRunning = false;
8510 MUTEX_LOCK( &stream_.mutex );
8511 if ( stream_.state == STREAM_STOPPED ) {
8512 pah->runnable = true;
8513 pthread_cond_signal( &pah->runnable_cv );
8515 MUTEX_UNLOCK( &stream_.mutex );
8517 pthread_join( pah->thread, 0 );
8518 if ( pah->s_play ) {
8519 pa_simple_flush( pah->s_play, NULL );
8520 pa_simple_free( pah->s_play );
8523 pa_simple_free( pah->s_rec );
8525 pthread_cond_destroy( &pah->runnable_cv );
8527 stream_.apiHandle = 0;
8530 if ( stream_.userBuffer[0] ) {
8531 free( stream_.userBuffer[0] );
8532 stream_.userBuffer[0] = 0;
8534 if ( stream_.userBuffer[1] ) {
8535 free( stream_.userBuffer[1] );
8536 stream_.userBuffer[1] = 0;
8539 stream_.state = STREAM_CLOSED;
8540 stream_.mode = UNINITIALIZED;
8543 void RtApiPulse::callbackEvent( void )
8545 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8547 if ( stream_.state == STREAM_STOPPED ) {
8548 MUTEX_LOCK( &stream_.mutex );
8549 while ( !pah->runnable )
8550 pthread_cond_wait( &pah->runnable_cv, &stream_.mutex );
8552 if ( stream_.state != STREAM_RUNNING ) {
8553 MUTEX_UNLOCK( &stream_.mutex );
8556 MUTEX_UNLOCK( &stream_.mutex );
8559 if ( stream_.state == STREAM_CLOSED ) {
8560 errorText_ = "RtApiPulse::callbackEvent(): the stream is closed ... "
8561 "this shouldn't happen!";
8562 error( RtAudioError::WARNING );
8566 RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
8567 double streamTime = getStreamTime();
8568 RtAudioStreamStatus status = 0;
8569 int doStopStream = callback( stream_.userBuffer[OUTPUT], stream_.userBuffer[INPUT],
8570 stream_.bufferSize, streamTime, status,
8571 stream_.callbackInfo.userData );
8573 if ( doStopStream == 2 ) {
8578 MUTEX_LOCK( &stream_.mutex );
8579 void *pulse_in = stream_.doConvertBuffer[INPUT] ? stream_.deviceBuffer : stream_.userBuffer[INPUT];
8580 void *pulse_out = stream_.doConvertBuffer[OUTPUT] ? stream_.deviceBuffer : stream_.userBuffer[OUTPUT];
8582 if ( stream_.state != STREAM_RUNNING )
8587 if (stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
8588 if ( stream_.doConvertBuffer[OUTPUT] ) {
8589 convertBuffer( stream_.deviceBuffer,
8590 stream_.userBuffer[OUTPUT],
8591 stream_.convertInfo[OUTPUT] );
8592 bytes = stream_.nDeviceChannels[OUTPUT] * stream_.bufferSize *
8593 formatBytes( stream_.deviceFormat[OUTPUT] );
8595 bytes = stream_.nUserChannels[OUTPUT] * stream_.bufferSize *
8596 formatBytes( stream_.userFormat );
8598 if ( pa_simple_write( pah->s_play, pulse_out, bytes, &pa_error ) < 0 ) {
8599 errorStream_ << "RtApiPulse::callbackEvent: audio write error, " <<
8600 pa_strerror( pa_error ) << ".";
8601 errorText_ = errorStream_.str();
8602 error( RtAudioError::WARNING );
8606 if ( stream_.mode == INPUT || stream_.mode == DUPLEX) {
8607 if ( stream_.doConvertBuffer[INPUT] )
8608 bytes = stream_.nDeviceChannels[INPUT] * stream_.bufferSize *
8609 formatBytes( stream_.deviceFormat[INPUT] );
8611 bytes = stream_.nUserChannels[INPUT] * stream_.bufferSize *
8612 formatBytes( stream_.userFormat );
8614 if ( pa_simple_read( pah->s_rec, pulse_in, bytes, &pa_error ) < 0 ) {
8615 errorStream_ << "RtApiPulse::callbackEvent: audio read error, " <<
8616 pa_strerror( pa_error ) << ".";
8617 errorText_ = errorStream_.str();
8618 error( RtAudioError::WARNING );
8620 if ( stream_.doConvertBuffer[INPUT] ) {
8621 convertBuffer( stream_.userBuffer[INPUT],
8622 stream_.deviceBuffer,
8623 stream_.convertInfo[INPUT] );
8628 MUTEX_UNLOCK( &stream_.mutex );
8629 RtApi::tickStreamTime();
8631 if ( doStopStream == 1 )
8635 void RtApiPulse::startStream( void )
8637 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8639 if ( stream_.state == STREAM_CLOSED ) {
8640 errorText_ = "RtApiPulse::startStream(): the stream is not open!";
8641 error( RtAudioError::INVALID_USE );
8644 if ( stream_.state == STREAM_RUNNING ) {
8645 errorText_ = "RtApiPulse::startStream(): the stream is already running!";
8646 error( RtAudioError::WARNING );
8650 MUTEX_LOCK( &stream_.mutex );
8652 stream_.state = STREAM_RUNNING;
8654 pah->runnable = true;
8655 pthread_cond_signal( &pah->runnable_cv );
8656 MUTEX_UNLOCK( &stream_.mutex );
8659 void RtApiPulse::stopStream( void )
8661 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8663 if ( stream_.state == STREAM_CLOSED ) {
8664 errorText_ = "RtApiPulse::stopStream(): the stream is not open!";
8665 error( RtAudioError::INVALID_USE );
8668 if ( stream_.state == STREAM_STOPPED ) {
8669 errorText_ = "RtApiPulse::stopStream(): the stream is already stopped!";
8670 error( RtAudioError::WARNING );
8674 stream_.state = STREAM_STOPPED;
8675 MUTEX_LOCK( &stream_.mutex );
8677 if ( pah && pah->s_play ) {
8679 if ( pa_simple_drain( pah->s_play, &pa_error ) < 0 ) {
8680 errorStream_ << "RtApiPulse::stopStream: error draining output device, " <<
8681 pa_strerror( pa_error ) << ".";
8682 errorText_ = errorStream_.str();
8683 MUTEX_UNLOCK( &stream_.mutex );
8684 error( RtAudioError::SYSTEM_ERROR );
8689 stream_.state = STREAM_STOPPED;
8690 MUTEX_UNLOCK( &stream_.mutex );
8693 void RtApiPulse::abortStream( void )
8695 PulseAudioHandle *pah = static_cast<PulseAudioHandle*>( stream_.apiHandle );
8697 if ( stream_.state == STREAM_CLOSED ) {
8698 errorText_ = "RtApiPulse::abortStream(): the stream is not open!";
8699 error( RtAudioError::INVALID_USE );
8702 if ( stream_.state == STREAM_STOPPED ) {
8703 errorText_ = "RtApiPulse::abortStream(): the stream is already stopped!";
8704 error( RtAudioError::WARNING );
8708 stream_.state = STREAM_STOPPED;
8709 MUTEX_LOCK( &stream_.mutex );
8711 if ( pah && pah->s_play ) {
8713 if ( pa_simple_flush( pah->s_play, &pa_error ) < 0 ) {
8714 errorStream_ << "RtApiPulse::abortStream: error flushing output device, " <<
8715 pa_strerror( pa_error ) << ".";
8716 errorText_ = errorStream_.str();
8717 MUTEX_UNLOCK( &stream_.mutex );
8718 error( RtAudioError::SYSTEM_ERROR );
8723 stream_.state = STREAM_STOPPED;
8724 MUTEX_UNLOCK( &stream_.mutex );
8727 bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode,
8728 unsigned int channels, unsigned int firstChannel,
8729 unsigned int sampleRate, RtAudioFormat format,
8730 unsigned int *bufferSize, RtAudio::StreamOptions *options )
8732 PulseAudioHandle *pah = 0;
8733 unsigned long bufferBytes = 0;
8736 if ( device != 0 ) return false;
8737 if ( mode != INPUT && mode != OUTPUT ) return false;
8738 if ( channels != 1 && channels != 2 ) {
8739 errorText_ = "RtApiPulse::probeDeviceOpen: unsupported number of channels.";
8742 ss.channels = channels;
8744 if ( firstChannel != 0 ) return false;
8746 bool sr_found = false;
8747 for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr ) {
8748 if ( sampleRate == *sr ) {
8750 stream_.sampleRate = sampleRate;
8751 ss.rate = sampleRate;
8756 errorText_ = "RtApiPulse::probeDeviceOpen: unsupported sample rate.";
8761 for ( const rtaudio_pa_format_mapping_t *sf = supported_sampleformats;
8762 sf->rtaudio_format && sf->pa_format != PA_SAMPLE_INVALID; ++sf ) {
8763 if ( format == sf->rtaudio_format ) {
8765 stream_.userFormat = sf->rtaudio_format;
8766 stream_.deviceFormat[mode] = stream_.userFormat;
8767 ss.format = sf->pa_format;
8771 if ( !sf_found ) { // Use internal data format conversion.
8772 stream_.userFormat = format;
8773 stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
8774 ss.format = PA_SAMPLE_FLOAT32LE;
8777 // Set other stream parameters.
8778 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
8779 else stream_.userInterleaved = true;
8780 stream_.deviceInterleaved[mode] = true;
8781 stream_.nBuffers = 1;
8782 stream_.doByteSwap[mode] = false;
8783 stream_.nUserChannels[mode] = channels;
8784 stream_.nDeviceChannels[mode] = channels + firstChannel;
8785 stream_.channelOffset[mode] = 0;
8786 std::string streamName = "RtAudio";
8788 // Set flags for buffer conversion.
8789 stream_.doConvertBuffer[mode] = false;
8790 if ( stream_.userFormat != stream_.deviceFormat[mode] )
8791 stream_.doConvertBuffer[mode] = true;
8792 if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
8793 stream_.doConvertBuffer[mode] = true;
8795 // Allocate necessary internal buffers.
8796 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
8797 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
8798 if ( stream_.userBuffer[mode] == NULL ) {
8799 errorText_ = "RtApiPulse::probeDeviceOpen: error allocating user buffer memory.";
8802 stream_.bufferSize = *bufferSize;
8804 if ( stream_.doConvertBuffer[mode] ) {
8806 bool makeBuffer = true;
8807 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
8808 if ( mode == INPUT ) {
8809 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
8810 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
8811 if ( bufferBytes <= bytesOut ) makeBuffer = false;
8816 bufferBytes *= *bufferSize;
8817 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
8818 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
8819 if ( stream_.deviceBuffer == NULL ) {
8820 errorText_ = "RtApiPulse::probeDeviceOpen: error allocating device buffer memory.";
8826 stream_.device[mode] = device;
8828 // Setup the buffer conversion information structure.
8829 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
8831 if ( !stream_.apiHandle ) {
8832 PulseAudioHandle *pah = new PulseAudioHandle;
8834 errorText_ = "RtApiPulse::probeDeviceOpen: error allocating memory for handle.";
8838 stream_.apiHandle = pah;
8839 if ( pthread_cond_init( &pah->runnable_cv, NULL ) != 0 ) {
8840 errorText_ = "RtApiPulse::probeDeviceOpen: error creating condition variable.";
8844 pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8847 if ( options && !options->streamName.empty() ) streamName = options->streamName;
8850 pa_buffer_attr buffer_attr;
8851 buffer_attr.fragsize = bufferBytes;
8852 buffer_attr.maxlength = -1;
8854 pah->s_rec = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_RECORD, NULL, "Record", &ss, NULL, &buffer_attr, &error );
8855 if ( !pah->s_rec ) {
8856 errorText_ = "RtApiPulse::probeDeviceOpen: error connecting input to PulseAudio server.";
8861 pah->s_play = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_PLAYBACK, NULL, "Playback", &ss, NULL, NULL, &error );
8862 if ( !pah->s_play ) {
8863 errorText_ = "RtApiPulse::probeDeviceOpen: error connecting output to PulseAudio server.";
8871 if ( stream_.mode == UNINITIALIZED )
8872 stream_.mode = mode;
8873 else if ( stream_.mode == mode )
8876 stream_.mode = DUPLEX;
8878 if ( !stream_.callbackInfo.isRunning ) {
8879 stream_.callbackInfo.object = this;
8881 stream_.state = STREAM_STOPPED;
8882 // Set the thread attributes for joinable and realtime scheduling
8883 // priority (optional). The higher priority will only take affect
8884 // if the program is run as root or suid. Note, under Linux
8885 // processes with CAP_SYS_NICE privilege, a user can change
8886 // scheduling policy and priority (thus need not be root). See
8887 // POSIX "capabilities".
8888 pthread_attr_t attr;
8889 pthread_attr_init( &attr );
8890 pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
8891 #ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
8892 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
8893 stream_.callbackInfo.doRealtime = true;
8894 struct sched_param param;
8895 int priority = options->priority;
8896 int min = sched_get_priority_min( SCHED_RR );
8897 int max = sched_get_priority_max( SCHED_RR );
8898 if ( priority < min ) priority = min;
8899 else if ( priority > max ) priority = max;
8900 param.sched_priority = priority;
8902 // Set the policy BEFORE the priority. Otherwise it fails.
8903 pthread_attr_setschedpolicy(&attr, SCHED_RR);
8904 pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM);
8905 // This is definitely required. Otherwise it fails.
8906 pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED);
8907 pthread_attr_setschedparam(&attr, ¶m);
8910 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
8912 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
8915 stream_.callbackInfo.isRunning = true;
8916 int result = pthread_create( &pah->thread, &attr, pulseaudio_callback, (void *)&stream_.callbackInfo);
8917 pthread_attr_destroy(&attr);
8919 // Failed. Try instead with default attributes.
8920 result = pthread_create( &pah->thread, NULL, pulseaudio_callback, (void *)&stream_.callbackInfo);
8922 stream_.callbackInfo.isRunning = false;
8923 errorText_ = "RtApiPulse::probeDeviceOpen: error creating thread.";
8932 if ( pah && stream_.callbackInfo.isRunning ) {
8933 pthread_cond_destroy( &pah->runnable_cv );
8935 stream_.apiHandle = 0;
8938 for ( int i=0; i<2; i++ ) {
8939 if ( stream_.userBuffer[i] ) {
8940 free( stream_.userBuffer[i] );
8941 stream_.userBuffer[i] = 0;
8945 if ( stream_.deviceBuffer ) {
8946 free( stream_.deviceBuffer );
8947 stream_.deviceBuffer = 0;
8950 stream_.state = STREAM_CLOSED;
8954 //******************** End of __LINUX_PULSE__ *********************//
8957 #if defined(__LINUX_OSS__)
8960 #include <sys/ioctl.h>
8963 #include <sys/soundcard.h>
8967 static void *ossCallbackHandler(void * ptr);
8969 // A structure to hold various information related to the OSS API
8972 int id[2]; // device ids
8975 pthread_cond_t runnable;
8978 :triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
8981 RtApiOss :: RtApiOss()
8983 // Nothing to do here.
8986 RtApiOss :: ~RtApiOss()
8988 if ( stream_.state != STREAM_CLOSED ) closeStream();
8991 unsigned int RtApiOss :: getDeviceCount( void )
8993 int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
8994 if ( mixerfd == -1 ) {
8995 errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'.";
8996 error( RtAudioError::WARNING );
9000 oss_sysinfo sysinfo;
9001 if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) {
9003 errorText_ = "RtApiOss::getDeviceCount: error getting sysinfo, OSS version >= 4.0 is required.";
9004 error( RtAudioError::WARNING );
9009 return sysinfo.numaudios;
9012 RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device )
9014 RtAudio::DeviceInfo info;
9015 info.probed = false;
9017 int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
9018 if ( mixerfd == -1 ) {
9019 errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'.";
9020 error( RtAudioError::WARNING );
9024 oss_sysinfo sysinfo;
9025 int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
9026 if ( result == -1 ) {
9028 errorText_ = "RtApiOss::getDeviceInfo: error getting sysinfo, OSS version >= 4.0 is required.";
9029 error( RtAudioError::WARNING );
9033 unsigned nDevices = sysinfo.numaudios;
9034 if ( nDevices == 0 ) {
9036 errorText_ = "RtApiOss::getDeviceInfo: no devices found!";
9037 error( RtAudioError::INVALID_USE );
9041 if ( device >= nDevices ) {
9043 errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!";
9044 error( RtAudioError::INVALID_USE );
9048 oss_audioinfo ainfo;
9050 result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
9052 if ( result == -1 ) {
9053 errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
9054 errorText_ = errorStream_.str();
9055 error( RtAudioError::WARNING );
9060 if ( ainfo.caps & PCM_CAP_OUTPUT ) info.outputChannels = ainfo.max_channels;
9061 if ( ainfo.caps & PCM_CAP_INPUT ) info.inputChannels = ainfo.max_channels;
9062 if ( ainfo.caps & PCM_CAP_DUPLEX ) {
9063 if ( info.outputChannels > 0 && info.inputChannels > 0 && ainfo.caps & PCM_CAP_DUPLEX )
9064 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
9067 // Probe data formats ... do for input
9068 unsigned long mask = ainfo.iformats;
9069 if ( mask & AFMT_S16_LE || mask & AFMT_S16_BE )
9070 info.nativeFormats |= RTAUDIO_SINT16;
9071 if ( mask & AFMT_S8 )
9072 info.nativeFormats |= RTAUDIO_SINT8;
9073 if ( mask & AFMT_S32_LE || mask & AFMT_S32_BE )
9074 info.nativeFormats |= RTAUDIO_SINT32;
9076 if ( mask & AFMT_FLOAT )
9077 info.nativeFormats |= RTAUDIO_FLOAT32;
9079 if ( mask & AFMT_S24_LE || mask & AFMT_S24_BE )
9080 info.nativeFormats |= RTAUDIO_SINT24;
9082 // Check that we have at least one supported format
9083 if ( info.nativeFormats == 0 ) {
9084 errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio.";
9085 errorText_ = errorStream_.str();
9086 error( RtAudioError::WARNING );
9090 // Probe the supported sample rates.
9091 info.sampleRates.clear();
9092 if ( ainfo.nrates ) {
9093 for ( unsigned int i=0; i<ainfo.nrates; i++ ) {
9094 for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
9095 if ( ainfo.rates[i] == SAMPLE_RATES[k] ) {
9096 info.sampleRates.push_back( SAMPLE_RATES[k] );
9098 if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
9099 info.preferredSampleRate = SAMPLE_RATES[k];
9107 // Check min and max rate values;
9108 for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
9109 if ( ainfo.min_rate <= (int) SAMPLE_RATES[k] && ainfo.max_rate >= (int) SAMPLE_RATES[k] ) {
9110 info.sampleRates.push_back( SAMPLE_RATES[k] );
9112 if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
9113 info.preferredSampleRate = SAMPLE_RATES[k];
9118 if ( info.sampleRates.size() == 0 ) {
9119 errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ").";
9120 errorText_ = errorStream_.str();
9121 error( RtAudioError::WARNING );
9125 info.name = ainfo.name;
9132 bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
9133 unsigned int firstChannel, unsigned int sampleRate,
9134 RtAudioFormat format, unsigned int *bufferSize,
9135 RtAudio::StreamOptions *options )
9137 int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
9138 if ( mixerfd == -1 ) {
9139 errorText_ = "RtApiOss::probeDeviceOpen: error opening '/dev/mixer'.";
9143 oss_sysinfo sysinfo;
9144 int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
9145 if ( result == -1 ) {
9147 errorText_ = "RtApiOss::probeDeviceOpen: error getting sysinfo, OSS version >= 4.0 is required.";
9151 unsigned nDevices = sysinfo.numaudios;
9152 if ( nDevices == 0 ) {
9153 // This should not happen because a check is made before this function is called.
9155 errorText_ = "RtApiOss::probeDeviceOpen: no devices found!";
9159 if ( device >= nDevices ) {
9160 // This should not happen because a check is made before this function is called.
9162 errorText_ = "RtApiOss::probeDeviceOpen: device ID is invalid!";
9166 oss_audioinfo ainfo;
9168 result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
9170 if ( result == -1 ) {
9171 errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
9172 errorText_ = errorStream_.str();
9176 // Check if device supports input or output
9177 if ( ( mode == OUTPUT && !( ainfo.caps & PCM_CAP_OUTPUT ) ) ||
9178 ( mode == INPUT && !( ainfo.caps & PCM_CAP_INPUT ) ) ) {
9179 if ( mode == OUTPUT )
9180 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support output.";
9182 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support input.";
9183 errorText_ = errorStream_.str();
9188 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9189 if ( mode == OUTPUT )
9191 else { // mode == INPUT
9192 if (stream_.mode == OUTPUT && stream_.device[0] == device) {
9193 // We just set the same device for playback ... close and reopen for duplex (OSS only).
9194 close( handle->id[0] );
9196 if ( !( ainfo.caps & PCM_CAP_DUPLEX ) ) {
9197 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support duplex mode.";
9198 errorText_ = errorStream_.str();
9201 // Check that the number previously set channels is the same.
9202 if ( stream_.nUserChannels[0] != channels ) {
9203 errorStream_ << "RtApiOss::probeDeviceOpen: input/output channels must be equal for OSS duplex device (" << ainfo.name << ").";
9204 errorText_ = errorStream_.str();
9213 // Set exclusive access if specified.
9214 if ( options && options->flags & RTAUDIO_HOG_DEVICE ) flags |= O_EXCL;
9216 // Try to open the device.
9218 fd = open( ainfo.devnode, flags, 0 );
9220 if ( errno == EBUSY )
9221 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") is busy.";
9223 errorStream_ << "RtApiOss::probeDeviceOpen: error opening device (" << ainfo.name << ").";
9224 errorText_ = errorStream_.str();
9228 // For duplex operation, specifically set this mode (this doesn't seem to work).
9230 if ( flags | O_RDWR ) {
9231 result = ioctl( fd, SNDCTL_DSP_SETDUPLEX, NULL );
9232 if ( result == -1) {
9233 errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ").";
9234 errorText_ = errorStream_.str();
9240 // Check the device channel support.
9241 stream_.nUserChannels[mode] = channels;
9242 if ( ainfo.max_channels < (int)(channels + firstChannel) ) {
9244 errorStream_ << "RtApiOss::probeDeviceOpen: the device (" << ainfo.name << ") does not support requested channel parameters.";
9245 errorText_ = errorStream_.str();
9249 // Set the number of channels.
9250 int deviceChannels = channels + firstChannel;
9251 result = ioctl( fd, SNDCTL_DSP_CHANNELS, &deviceChannels );
9252 if ( result == -1 || deviceChannels < (int)(channels + firstChannel) ) {
9254 errorStream_ << "RtApiOss::probeDeviceOpen: error setting channel parameters on device (" << ainfo.name << ").";
9255 errorText_ = errorStream_.str();
9258 stream_.nDeviceChannels[mode] = deviceChannels;
9260 // Get the data format mask
9262 result = ioctl( fd, SNDCTL_DSP_GETFMTS, &mask );
9263 if ( result == -1 ) {
9265 errorStream_ << "RtApiOss::probeDeviceOpen: error getting device (" << ainfo.name << ") data formats.";
9266 errorText_ = errorStream_.str();
9270 // Determine how to set the device format.
9271 stream_.userFormat = format;
9272 int deviceFormat = -1;
9273 stream_.doByteSwap[mode] = false;
9274 if ( format == RTAUDIO_SINT8 ) {
9275 if ( mask & AFMT_S8 ) {
9276 deviceFormat = AFMT_S8;
9277 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
9280 else if ( format == RTAUDIO_SINT16 ) {
9281 if ( mask & AFMT_S16_NE ) {
9282 deviceFormat = AFMT_S16_NE;
9283 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
9285 else if ( mask & AFMT_S16_OE ) {
9286 deviceFormat = AFMT_S16_OE;
9287 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
9288 stream_.doByteSwap[mode] = true;
9291 else if ( format == RTAUDIO_SINT24 ) {
9292 if ( mask & AFMT_S24_NE ) {
9293 deviceFormat = AFMT_S24_NE;
9294 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
9296 else if ( mask & AFMT_S24_OE ) {
9297 deviceFormat = AFMT_S24_OE;
9298 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
9299 stream_.doByteSwap[mode] = true;
9302 else if ( format == RTAUDIO_SINT32 ) {
9303 if ( mask & AFMT_S32_NE ) {
9304 deviceFormat = AFMT_S32_NE;
9305 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
9307 else if ( mask & AFMT_S32_OE ) {
9308 deviceFormat = AFMT_S32_OE;
9309 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
9310 stream_.doByteSwap[mode] = true;
9314 if ( deviceFormat == -1 ) {
9315 // The user requested format is not natively supported by the device.
9316 if ( mask & AFMT_S16_NE ) {
9317 deviceFormat = AFMT_S16_NE;
9318 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
9320 else if ( mask & AFMT_S32_NE ) {
9321 deviceFormat = AFMT_S32_NE;
9322 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
9324 else if ( mask & AFMT_S24_NE ) {
9325 deviceFormat = AFMT_S24_NE;
9326 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
9328 else if ( mask & AFMT_S16_OE ) {
9329 deviceFormat = AFMT_S16_OE;
9330 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
9331 stream_.doByteSwap[mode] = true;
9333 else if ( mask & AFMT_S32_OE ) {
9334 deviceFormat = AFMT_S32_OE;
9335 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
9336 stream_.doByteSwap[mode] = true;
9338 else if ( mask & AFMT_S24_OE ) {
9339 deviceFormat = AFMT_S24_OE;
9340 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
9341 stream_.doByteSwap[mode] = true;
9343 else if ( mask & AFMT_S8) {
9344 deviceFormat = AFMT_S8;
9345 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
9349 if ( stream_.deviceFormat[mode] == 0 ) {
9350 // This really shouldn't happen ...
9352 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") data format not supported by RtAudio.";
9353 errorText_ = errorStream_.str();
9357 // Set the data format.
9358 int temp = deviceFormat;
9359 result = ioctl( fd, SNDCTL_DSP_SETFMT, &deviceFormat );
9360 if ( result == -1 || deviceFormat != temp ) {
9362 errorStream_ << "RtApiOss::probeDeviceOpen: error setting data format on device (" << ainfo.name << ").";
9363 errorText_ = errorStream_.str();
9367 // Attempt to set the buffer size. According to OSS, the minimum
9368 // number of buffers is two. The supposed minimum buffer size is 16
9369 // bytes, so that will be our lower bound. The argument to this
9370 // call is in the form 0xMMMMSSSS (hex), where the buffer size (in
9371 // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM.
9372 // We'll check the actual value used near the end of the setup
9374 int ossBufferBytes = *bufferSize * formatBytes( stream_.deviceFormat[mode] ) * deviceChannels;
9375 if ( ossBufferBytes < 16 ) ossBufferBytes = 16;
9377 if ( options ) buffers = options->numberOfBuffers;
9378 if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) buffers = 2;
9379 if ( buffers < 2 ) buffers = 3;
9380 temp = ((int) buffers << 16) + (int)( log10( (double)ossBufferBytes ) / log10( 2.0 ) );
9381 result = ioctl( fd, SNDCTL_DSP_SETFRAGMENT, &temp );
9382 if ( result == -1 ) {
9384 errorStream_ << "RtApiOss::probeDeviceOpen: error setting buffer size on device (" << ainfo.name << ").";
9385 errorText_ = errorStream_.str();
9388 stream_.nBuffers = buffers;
9390 // Save buffer size (in sample frames).
9391 *bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels );
9392 stream_.bufferSize = *bufferSize;
9394 // Set the sample rate.
9395 int srate = sampleRate;
9396 result = ioctl( fd, SNDCTL_DSP_SPEED, &srate );
9397 if ( result == -1 ) {
9399 errorStream_ << "RtApiOss::probeDeviceOpen: error setting sample rate (" << sampleRate << ") on device (" << ainfo.name << ").";
9400 errorText_ = errorStream_.str();
9404 // Verify the sample rate setup worked.
9405 if ( abs( srate - (int)sampleRate ) > 100 ) {
9407 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ").";
9408 errorText_ = errorStream_.str();
9411 stream_.sampleRate = sampleRate;
9413 if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device) {
9414 // We're doing duplex setup here.
9415 stream_.deviceFormat[0] = stream_.deviceFormat[1];
9416 stream_.nDeviceChannels[0] = deviceChannels;
9419 // Set interleaving parameters.
9420 stream_.userInterleaved = true;
9421 stream_.deviceInterleaved[mode] = true;
9422 if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
9423 stream_.userInterleaved = false;
9425 // Set flags for buffer conversion
9426 stream_.doConvertBuffer[mode] = false;
9427 if ( stream_.userFormat != stream_.deviceFormat[mode] )
9428 stream_.doConvertBuffer[mode] = true;
9429 if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
9430 stream_.doConvertBuffer[mode] = true;
9431 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
9432 stream_.nUserChannels[mode] > 1 )
9433 stream_.doConvertBuffer[mode] = true;
9435 // Allocate the stream handles if necessary and then save.
9436 if ( stream_.apiHandle == 0 ) {
9438 handle = new OssHandle;
9440 catch ( std::bad_alloc& ) {
9441 errorText_ = "RtApiOss::probeDeviceOpen: error allocating OssHandle memory.";
9445 if ( pthread_cond_init( &handle->runnable, NULL ) ) {
9446 errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable.";
9450 stream_.apiHandle = (void *) handle;
9453 handle = (OssHandle *) stream_.apiHandle;
9455 handle->id[mode] = fd;
9457 // Allocate necessary internal buffers.
9458 unsigned long bufferBytes;
9459 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
9460 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
9461 if ( stream_.userBuffer[mode] == NULL ) {
9462 errorText_ = "RtApiOss::probeDeviceOpen: error allocating user buffer memory.";
9466 if ( stream_.doConvertBuffer[mode] ) {
9468 bool makeBuffer = true;
9469 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
9470 if ( mode == INPUT ) {
9471 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
9472 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
9473 if ( bufferBytes <= bytesOut ) makeBuffer = false;
9478 bufferBytes *= *bufferSize;
9479 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
9480 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
9481 if ( stream_.deviceBuffer == NULL ) {
9482 errorText_ = "RtApiOss::probeDeviceOpen: error allocating device buffer memory.";
9488 stream_.device[mode] = device;
9489 stream_.state = STREAM_STOPPED;
9491 // Setup the buffer conversion information structure.
9492 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
9494 // Setup thread if necessary.
9495 if ( stream_.mode == OUTPUT && mode == INPUT ) {
9496 // We had already set up an output stream.
9497 stream_.mode = DUPLEX;
9498 if ( stream_.device[0] == device ) handle->id[0] = fd;
9501 stream_.mode = mode;
9503 // Setup callback thread.
9504 stream_.callbackInfo.object = (void *) this;
9506 // Set the thread attributes for joinable and realtime scheduling
9507 // priority. The higher priority will only take affect if the
9508 // program is run as root or suid.
9509 pthread_attr_t attr;
9510 pthread_attr_init( &attr );
9511 pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
9512 #ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
9513 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
9514 stream_.callbackInfo.doRealtime = true;
9515 struct sched_param param;
9516 int priority = options->priority;
9517 int min = sched_get_priority_min( SCHED_RR );
9518 int max = sched_get_priority_max( SCHED_RR );
9519 if ( priority < min ) priority = min;
9520 else if ( priority > max ) priority = max;
9521 param.sched_priority = priority;
9523 // Set the policy BEFORE the priority. Otherwise it fails.
9524 pthread_attr_setschedpolicy(&attr, SCHED_RR);
9525 pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM);
9526 // This is definitely required. Otherwise it fails.
9527 pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED);
9528 pthread_attr_setschedparam(&attr, ¶m);
9531 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
9533 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
9536 stream_.callbackInfo.isRunning = true;
9537 result = pthread_create( &stream_.callbackInfo.thread, &attr, ossCallbackHandler, &stream_.callbackInfo );
9538 pthread_attr_destroy( &attr );
9540 // Failed. Try instead with default attributes.
9541 result = pthread_create( &stream_.callbackInfo.thread, NULL, ossCallbackHandler, &stream_.callbackInfo );
9543 stream_.callbackInfo.isRunning = false;
9544 errorText_ = "RtApiOss::error creating callback thread!";
9554 pthread_cond_destroy( &handle->runnable );
9555 if ( handle->id[0] ) close( handle->id[0] );
9556 if ( handle->id[1] ) close( handle->id[1] );
9558 stream_.apiHandle = 0;
9561 for ( int i=0; i<2; i++ ) {
9562 if ( stream_.userBuffer[i] ) {
9563 free( stream_.userBuffer[i] );
9564 stream_.userBuffer[i] = 0;
9568 if ( stream_.deviceBuffer ) {
9569 free( stream_.deviceBuffer );
9570 stream_.deviceBuffer = 0;
9573 stream_.state = STREAM_CLOSED;
9577 void RtApiOss :: closeStream()
9579 if ( stream_.state == STREAM_CLOSED ) {
9580 errorText_ = "RtApiOss::closeStream(): no open stream to close!";
9581 error( RtAudioError::WARNING );
9585 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9586 stream_.callbackInfo.isRunning = false;
9587 MUTEX_LOCK( &stream_.mutex );
9588 if ( stream_.state == STREAM_STOPPED )
9589 pthread_cond_signal( &handle->runnable );
9590 MUTEX_UNLOCK( &stream_.mutex );
9591 pthread_join( stream_.callbackInfo.thread, NULL );
9593 if ( stream_.state == STREAM_RUNNING ) {
9594 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
9595 ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
9597 ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
9598 stream_.state = STREAM_STOPPED;
9602 pthread_cond_destroy( &handle->runnable );
9603 if ( handle->id[0] ) close( handle->id[0] );
9604 if ( handle->id[1] ) close( handle->id[1] );
9606 stream_.apiHandle = 0;
9609 for ( int i=0; i<2; i++ ) {
9610 if ( stream_.userBuffer[i] ) {
9611 free( stream_.userBuffer[i] );
9612 stream_.userBuffer[i] = 0;
9616 if ( stream_.deviceBuffer ) {
9617 free( stream_.deviceBuffer );
9618 stream_.deviceBuffer = 0;
9621 stream_.mode = UNINITIALIZED;
9622 stream_.state = STREAM_CLOSED;
9625 void RtApiOss :: startStream()
9628 if ( stream_.state == STREAM_RUNNING ) {
9629 errorText_ = "RtApiOss::startStream(): the stream is already running!";
9630 error( RtAudioError::WARNING );
9634 MUTEX_LOCK( &stream_.mutex );
9636 stream_.state = STREAM_RUNNING;
9638 // No need to do anything else here ... OSS automatically starts
9639 // when fed samples.
9641 MUTEX_UNLOCK( &stream_.mutex );
9643 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9644 pthread_cond_signal( &handle->runnable );
9647 void RtApiOss :: stopStream()
9650 if ( stream_.state == STREAM_STOPPED ) {
9651 errorText_ = "RtApiOss::stopStream(): the stream is already stopped!";
9652 error( RtAudioError::WARNING );
9656 MUTEX_LOCK( &stream_.mutex );
9658 // The state might change while waiting on a mutex.
9659 if ( stream_.state == STREAM_STOPPED ) {
9660 MUTEX_UNLOCK( &stream_.mutex );
9665 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9666 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
9668 // Flush the output with zeros a few times.
9671 RtAudioFormat format;
9673 if ( stream_.doConvertBuffer[0] ) {
9674 buffer = stream_.deviceBuffer;
9675 samples = stream_.bufferSize * stream_.nDeviceChannels[0];
9676 format = stream_.deviceFormat[0];
9679 buffer = stream_.userBuffer[0];
9680 samples = stream_.bufferSize * stream_.nUserChannels[0];
9681 format = stream_.userFormat;
9684 memset( buffer, 0, samples * formatBytes(format) );
9685 for ( unsigned int i=0; i<stream_.nBuffers+1; i++ ) {
9686 result = write( handle->id[0], buffer, samples * formatBytes(format) );
9687 if ( result == -1 ) {
9688 errorText_ = "RtApiOss::stopStream: audio write error.";
9689 error( RtAudioError::WARNING );
9693 result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
9694 if ( result == -1 ) {
9695 errorStream_ << "RtApiOss::stopStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
9696 errorText_ = errorStream_.str();
9699 handle->triggered = false;
9702 if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
9703 result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
9704 if ( result == -1 ) {
9705 errorStream_ << "RtApiOss::stopStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
9706 errorText_ = errorStream_.str();
9712 stream_.state = STREAM_STOPPED;
9713 MUTEX_UNLOCK( &stream_.mutex );
9715 if ( result != -1 ) return;
9716 error( RtAudioError::SYSTEM_ERROR );
9719 void RtApiOss :: abortStream()
9722 if ( stream_.state == STREAM_STOPPED ) {
9723 errorText_ = "RtApiOss::abortStream(): the stream is already stopped!";
9724 error( RtAudioError::WARNING );
9728 MUTEX_LOCK( &stream_.mutex );
9730 // The state might change while waiting on a mutex.
9731 if ( stream_.state == STREAM_STOPPED ) {
9732 MUTEX_UNLOCK( &stream_.mutex );
9737 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9738 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
9739 result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
9740 if ( result == -1 ) {
9741 errorStream_ << "RtApiOss::abortStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
9742 errorText_ = errorStream_.str();
9745 handle->triggered = false;
9748 if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
9749 result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
9750 if ( result == -1 ) {
9751 errorStream_ << "RtApiOss::abortStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
9752 errorText_ = errorStream_.str();
9758 stream_.state = STREAM_STOPPED;
9759 MUTEX_UNLOCK( &stream_.mutex );
9761 if ( result != -1 ) return;
9762 error( RtAudioError::SYSTEM_ERROR );
9765 void RtApiOss :: callbackEvent()
9767 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9768 if ( stream_.state == STREAM_STOPPED ) {
9769 MUTEX_LOCK( &stream_.mutex );
9770 pthread_cond_wait( &handle->runnable, &stream_.mutex );
9771 if ( stream_.state != STREAM_RUNNING ) {
9772 MUTEX_UNLOCK( &stream_.mutex );
9775 MUTEX_UNLOCK( &stream_.mutex );
9778 if ( stream_.state == STREAM_CLOSED ) {
9779 errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!";
9780 error( RtAudioError::WARNING );
9784 // Invoke user callback to get fresh output data.
9785 int doStopStream = 0;
9786 RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
9787 double streamTime = getStreamTime();
9788 RtAudioStreamStatus status = 0;
9789 if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
9790 status |= RTAUDIO_OUTPUT_UNDERFLOW;
9791 handle->xrun[0] = false;
9793 if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
9794 status |= RTAUDIO_INPUT_OVERFLOW;
9795 handle->xrun[1] = false;
9797 doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
9798 stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
9799 if ( doStopStream == 2 ) {
9800 this->abortStream();
9804 MUTEX_LOCK( &stream_.mutex );
9806 // The state might change while waiting on a mutex.
9807 if ( stream_.state == STREAM_STOPPED ) goto unlock;
9812 RtAudioFormat format;
9814 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
9816 // Setup parameters and do buffer conversion if necessary.
9817 if ( stream_.doConvertBuffer[0] ) {
9818 buffer = stream_.deviceBuffer;
9819 convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
9820 samples = stream_.bufferSize * stream_.nDeviceChannels[0];
9821 format = stream_.deviceFormat[0];
9824 buffer = stream_.userBuffer[0];
9825 samples = stream_.bufferSize * stream_.nUserChannels[0];
9826 format = stream_.userFormat;
9829 // Do byte swapping if necessary.
9830 if ( stream_.doByteSwap[0] )
9831 byteSwapBuffer( buffer, samples, format );
9833 if ( stream_.mode == DUPLEX && handle->triggered == false ) {
9835 ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
9836 result = write( handle->id[0], buffer, samples * formatBytes(format) );
9837 trig = PCM_ENABLE_INPUT|PCM_ENABLE_OUTPUT;
9838 ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
9839 handle->triggered = true;
9842 // Write samples to device.
9843 result = write( handle->id[0], buffer, samples * formatBytes(format) );
9845 if ( result == -1 ) {
9846 // We'll assume this is an underrun, though there isn't a
9847 // specific means for determining that.
9848 handle->xrun[0] = true;
9849 errorText_ = "RtApiOss::callbackEvent: audio write error.";
9850 error( RtAudioError::WARNING );
9851 // Continue on to input section.
9855 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
9857 // Setup parameters.
9858 if ( stream_.doConvertBuffer[1] ) {
9859 buffer = stream_.deviceBuffer;
9860 samples = stream_.bufferSize * stream_.nDeviceChannels[1];
9861 format = stream_.deviceFormat[1];
9864 buffer = stream_.userBuffer[1];
9865 samples = stream_.bufferSize * stream_.nUserChannels[1];
9866 format = stream_.userFormat;
9869 // Read samples from device.
9870 result = read( handle->id[1], buffer, samples * formatBytes(format) );
9872 if ( result == -1 ) {
9873 // We'll assume this is an overrun, though there isn't a
9874 // specific means for determining that.
9875 handle->xrun[1] = true;
9876 errorText_ = "RtApiOss::callbackEvent: audio read error.";
9877 error( RtAudioError::WARNING );
9881 // Do byte swapping if necessary.
9882 if ( stream_.doByteSwap[1] )
9883 byteSwapBuffer( buffer, samples, format );
9885 // Do buffer conversion if necessary.
9886 if ( stream_.doConvertBuffer[1] )
9887 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
9891 MUTEX_UNLOCK( &stream_.mutex );
9893 RtApi::tickStreamTime();
9894 if ( doStopStream == 1 ) this->stopStream();
9897 static void *ossCallbackHandler( void *ptr )
9899 CallbackInfo *info = (CallbackInfo *) ptr;
9900 RtApiOss *object = (RtApiOss *) info->object;
9901 bool *isRunning = &info->isRunning;
9903 #ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
9904 if (info->doRealtime) {
9905 std::cerr << "RtAudio oss: " <<
9906 (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") <<
9907 "running realtime scheduling" << std::endl;
9911 while ( *isRunning == true ) {
9912 pthread_testcancel();
9913 object->callbackEvent();
9916 pthread_exit( NULL );
9919 //******************** End of __LINUX_OSS__ *********************//
9923 // *************************************************** //
9925 // Protected common (OS-independent) RtAudio methods.
9927 // *************************************************** //
9929 // This method can be modified to control the behavior of error
9930 // message printing.
9931 void RtApi :: error( RtAudioError::Type type )
9933 errorStream_.str(""); // clear the ostringstream
9935 RtAudioErrorCallback errorCallback = (RtAudioErrorCallback) stream_.callbackInfo.errorCallback;
9936 if ( errorCallback ) {
9937 // abortStream() can generate new error messages. Ignore them. Just keep original one.
9939 if ( firstErrorOccurred_ )
9942 firstErrorOccurred_ = true;
9943 const std::string errorMessage = errorText_;
9945 if ( type != RtAudioError::WARNING && stream_.state != STREAM_STOPPED) {
9946 stream_.callbackInfo.isRunning = false; // exit from the thread
9950 errorCallback( type, errorMessage );
9951 firstErrorOccurred_ = false;
9955 if ( type == RtAudioError::WARNING && showWarnings_ == true )
9956 std::cerr << '\n' << errorText_ << "\n\n";
9957 else if ( type != RtAudioError::WARNING )
9958 throw( RtAudioError( errorText_, type ) );
9961 void RtApi :: verifyStream()
9963 if ( stream_.state == STREAM_CLOSED ) {
9964 errorText_ = "RtApi:: a stream is not open!";
9965 error( RtAudioError::INVALID_USE );
9969 void RtApi :: clearStreamInfo()
9971 stream_.mode = UNINITIALIZED;
9972 stream_.state = STREAM_CLOSED;
9973 stream_.sampleRate = 0;
9974 stream_.bufferSize = 0;
9975 stream_.nBuffers = 0;
9976 stream_.userFormat = 0;
9977 stream_.userInterleaved = true;
9978 stream_.streamTime = 0.0;
9979 stream_.apiHandle = 0;
9980 stream_.deviceBuffer = 0;
9981 stream_.callbackInfo.callback = 0;
9982 stream_.callbackInfo.userData = 0;
9983 stream_.callbackInfo.isRunning = false;
9984 stream_.callbackInfo.errorCallback = 0;
9985 for ( int i=0; i<2; i++ ) {
9986 stream_.device[i] = 11111;
9987 stream_.doConvertBuffer[i] = false;
9988 stream_.deviceInterleaved[i] = true;
9989 stream_.doByteSwap[i] = false;
9990 stream_.nUserChannels[i] = 0;
9991 stream_.nDeviceChannels[i] = 0;
9992 stream_.channelOffset[i] = 0;
9993 stream_.deviceFormat[i] = 0;
9994 stream_.latency[i] = 0;
9995 stream_.userBuffer[i] = 0;
9996 stream_.convertInfo[i].channels = 0;
9997 stream_.convertInfo[i].inJump = 0;
9998 stream_.convertInfo[i].outJump = 0;
9999 stream_.convertInfo[i].inFormat = 0;
10000 stream_.convertInfo[i].outFormat = 0;
10001 stream_.convertInfo[i].inOffset.clear();
10002 stream_.convertInfo[i].outOffset.clear();
10006 unsigned int RtApi :: formatBytes( RtAudioFormat format )
10008 if ( format == RTAUDIO_SINT16 )
10010 else if ( format == RTAUDIO_SINT32 || format == RTAUDIO_FLOAT32 )
10012 else if ( format == RTAUDIO_FLOAT64 )
10014 else if ( format == RTAUDIO_SINT24 )
10016 else if ( format == RTAUDIO_SINT8 )
10019 errorText_ = "RtApi::formatBytes: undefined format.";
10020 error( RtAudioError::WARNING );
10025 void RtApi :: setConvertInfo( StreamMode mode, unsigned int firstChannel )
10027 if ( mode == INPUT ) { // convert device to user buffer
10028 stream_.convertInfo[mode].inJump = stream_.nDeviceChannels[1];
10029 stream_.convertInfo[mode].outJump = stream_.nUserChannels[1];
10030 stream_.convertInfo[mode].inFormat = stream_.deviceFormat[1];
10031 stream_.convertInfo[mode].outFormat = stream_.userFormat;
10033 else { // convert user to device buffer
10034 stream_.convertInfo[mode].inJump = stream_.nUserChannels[0];
10035 stream_.convertInfo[mode].outJump = stream_.nDeviceChannels[0];
10036 stream_.convertInfo[mode].inFormat = stream_.userFormat;
10037 stream_.convertInfo[mode].outFormat = stream_.deviceFormat[0];
10040 if ( stream_.convertInfo[mode].inJump < stream_.convertInfo[mode].outJump )
10041 stream_.convertInfo[mode].channels = stream_.convertInfo[mode].inJump;
10043 stream_.convertInfo[mode].channels = stream_.convertInfo[mode].outJump;
10045 // Set up the interleave/deinterleave offsets.
10046 if ( stream_.deviceInterleaved[mode] != stream_.userInterleaved ) {
10047 if ( ( mode == OUTPUT && stream_.deviceInterleaved[mode] ) ||
10048 ( mode == INPUT && stream_.userInterleaved ) ) {
10049 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
10050 stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
10051 stream_.convertInfo[mode].outOffset.push_back( k );
10052 stream_.convertInfo[mode].inJump = 1;
10056 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
10057 stream_.convertInfo[mode].inOffset.push_back( k );
10058 stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
10059 stream_.convertInfo[mode].outJump = 1;
10063 else { // no (de)interleaving
10064 if ( stream_.userInterleaved ) {
10065 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
10066 stream_.convertInfo[mode].inOffset.push_back( k );
10067 stream_.convertInfo[mode].outOffset.push_back( k );
10071 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
10072 stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
10073 stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
10074 stream_.convertInfo[mode].inJump = 1;
10075 stream_.convertInfo[mode].outJump = 1;
10080 // Add channel offset.
10081 if ( firstChannel > 0 ) {
10082 if ( stream_.deviceInterleaved[mode] ) {
10083 if ( mode == OUTPUT ) {
10084 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
10085 stream_.convertInfo[mode].outOffset[k] += firstChannel;
10088 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
10089 stream_.convertInfo[mode].inOffset[k] += firstChannel;
10093 if ( mode == OUTPUT ) {
10094 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
10095 stream_.convertInfo[mode].outOffset[k] += ( firstChannel * stream_.bufferSize );
10098 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
10099 stream_.convertInfo[mode].inOffset[k] += ( firstChannel * stream_.bufferSize );
10105 void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info )
10107 // This function does format conversion, input/output channel compensation, and
10108 // data interleaving/deinterleaving. 24-bit integers are assumed to occupy
10109 // the lower three bytes of a 32-bit integer.
10111 // Clear our device buffer when in/out duplex device channels are different
10112 if ( outBuffer == stream_.deviceBuffer && stream_.mode == DUPLEX &&
10113 ( stream_.nDeviceChannels[0] < stream_.nDeviceChannels[1] ) )
10114 memset( outBuffer, 0, stream_.bufferSize * info.outJump * formatBytes( info.outFormat ) );
10117 if (info.outFormat == RTAUDIO_FLOAT64) {
10119 Float64 *out = (Float64 *)outBuffer;
10121 if (info.inFormat == RTAUDIO_SINT8) {
10122 signed char *in = (signed char *)inBuffer;
10123 scale = 1.0 / 127.5;
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_SINT16) {
10135 Int16 *in = (Int16 *)inBuffer;
10136 scale = 1.0 / 32767.5;
10137 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10138 for (j=0; j<info.channels; j++) {
10139 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
10140 out[info.outOffset[j]] += 0.5;
10141 out[info.outOffset[j]] *= scale;
10144 out += info.outJump;
10147 else if (info.inFormat == RTAUDIO_SINT24) {
10148 Int24 *in = (Int24 *)inBuffer;
10149 scale = 1.0 / 8388607.5;
10150 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10151 for (j=0; j<info.channels; j++) {
10152 out[info.outOffset[j]] = (Float64) (in[info.inOffset[j]].asInt());
10153 out[info.outOffset[j]] += 0.5;
10154 out[info.outOffset[j]] *= scale;
10157 out += info.outJump;
10160 else if (info.inFormat == RTAUDIO_SINT32) {
10161 Int32 *in = (Int32 *)inBuffer;
10162 scale = 1.0 / 2147483647.5;
10163 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10164 for (j=0; j<info.channels; j++) {
10165 out[info.outOffset[j]] = (Float64) 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_FLOAT32) {
10174 Float32 *in = (Float32 *)inBuffer;
10175 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10176 for (j=0; j<info.channels; j++) {
10177 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
10180 out += info.outJump;
10183 else if (info.inFormat == RTAUDIO_FLOAT64) {
10184 // Channel compensation and/or (de)interleaving only.
10185 Float64 *in = (Float64 *)inBuffer;
10186 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10187 for (j=0; j<info.channels; j++) {
10188 out[info.outOffset[j]] = in[info.inOffset[j]];
10191 out += info.outJump;
10195 else if (info.outFormat == RTAUDIO_FLOAT32) {
10197 Float32 *out = (Float32 *)outBuffer;
10199 if (info.inFormat == RTAUDIO_SINT8) {
10200 signed char *in = (signed char *)inBuffer;
10201 scale = (Float32) ( 1.0 / 127.5 );
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_SINT16) {
10213 Int16 *in = (Int16 *)inBuffer;
10214 scale = (Float32) ( 1.0 / 32767.5 );
10215 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10216 for (j=0; j<info.channels; j++) {
10217 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
10218 out[info.outOffset[j]] += 0.5;
10219 out[info.outOffset[j]] *= scale;
10222 out += info.outJump;
10225 else if (info.inFormat == RTAUDIO_SINT24) {
10226 Int24 *in = (Int24 *)inBuffer;
10227 scale = (Float32) ( 1.0 / 8388607.5 );
10228 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10229 for (j=0; j<info.channels; j++) {
10230 out[info.outOffset[j]] = (Float32) (in[info.inOffset[j]].asInt());
10231 out[info.outOffset[j]] += 0.5;
10232 out[info.outOffset[j]] *= scale;
10235 out += info.outJump;
10238 else if (info.inFormat == RTAUDIO_SINT32) {
10239 Int32 *in = (Int32 *)inBuffer;
10240 scale = (Float32) ( 1.0 / 2147483647.5 );
10241 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10242 for (j=0; j<info.channels; j++) {
10243 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
10244 out[info.outOffset[j]] += 0.5;
10245 out[info.outOffset[j]] *= scale;
10248 out += info.outJump;
10251 else if (info.inFormat == RTAUDIO_FLOAT32) {
10252 // Channel compensation and/or (de)interleaving only.
10253 Float32 *in = (Float32 *)inBuffer;
10254 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10255 for (j=0; j<info.channels; j++) {
10256 out[info.outOffset[j]] = in[info.inOffset[j]];
10259 out += info.outJump;
10262 else if (info.inFormat == RTAUDIO_FLOAT64) {
10263 Float64 *in = (Float64 *)inBuffer;
10264 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10265 for (j=0; j<info.channels; j++) {
10266 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
10269 out += info.outJump;
10273 else if (info.outFormat == RTAUDIO_SINT32) {
10274 Int32 *out = (Int32 *)outBuffer;
10275 if (info.inFormat == RTAUDIO_SINT8) {
10276 signed char *in = (signed char *)inBuffer;
10277 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10278 for (j=0; j<info.channels; j++) {
10279 out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
10280 out[info.outOffset[j]] <<= 24;
10283 out += info.outJump;
10286 else if (info.inFormat == RTAUDIO_SINT16) {
10287 Int16 *in = (Int16 *)inBuffer;
10288 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10289 for (j=0; j<info.channels; j++) {
10290 out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
10291 out[info.outOffset[j]] <<= 16;
10294 out += info.outJump;
10297 else if (info.inFormat == RTAUDIO_SINT24) {
10298 Int24 *in = (Int24 *)inBuffer;
10299 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10300 for (j=0; j<info.channels; j++) {
10301 out[info.outOffset[j]] = (Int32) in[info.inOffset[j]].asInt();
10302 out[info.outOffset[j]] <<= 8;
10305 out += info.outJump;
10308 else if (info.inFormat == RTAUDIO_SINT32) {
10309 // Channel compensation and/or (de)interleaving only.
10310 Int32 *in = (Int32 *)inBuffer;
10311 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10312 for (j=0; j<info.channels; j++) {
10313 out[info.outOffset[j]] = in[info.inOffset[j]];
10316 out += info.outJump;
10319 else if (info.inFormat == RTAUDIO_FLOAT32) {
10320 Float32 *in = (Float32 *)inBuffer;
10321 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10322 for (j=0; j<info.channels; j++) {
10323 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
10326 out += info.outJump;
10329 else if (info.inFormat == RTAUDIO_FLOAT64) {
10330 Float64 *in = (Float64 *)inBuffer;
10331 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10332 for (j=0; j<info.channels; j++) {
10333 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
10336 out += info.outJump;
10340 else if (info.outFormat == RTAUDIO_SINT24) {
10341 Int24 *out = (Int24 *)outBuffer;
10342 if (info.inFormat == RTAUDIO_SINT8) {
10343 signed char *in = (signed char *)inBuffer;
10344 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10345 for (j=0; j<info.channels; j++) {
10346 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] << 16);
10347 //out[info.outOffset[j]] <<= 16;
10350 out += info.outJump;
10353 else if (info.inFormat == RTAUDIO_SINT16) {
10354 Int16 *in = (Int16 *)inBuffer;
10355 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10356 for (j=0; j<info.channels; j++) {
10357 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] << 8);
10358 //out[info.outOffset[j]] <<= 8;
10361 out += info.outJump;
10364 else if (info.inFormat == RTAUDIO_SINT24) {
10365 // Channel compensation and/or (de)interleaving only.
10366 Int24 *in = (Int24 *)inBuffer;
10367 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10368 for (j=0; j<info.channels; j++) {
10369 out[info.outOffset[j]] = in[info.inOffset[j]];
10372 out += info.outJump;
10375 else if (info.inFormat == RTAUDIO_SINT32) {
10376 Int32 *in = (Int32 *)inBuffer;
10377 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10378 for (j=0; j<info.channels; j++) {
10379 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] >> 8);
10380 //out[info.outOffset[j]] >>= 8;
10383 out += info.outJump;
10386 else if (info.inFormat == RTAUDIO_FLOAT32) {
10387 Float32 *in = (Float32 *)inBuffer;
10388 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10389 for (j=0; j<info.channels; j++) {
10390 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
10393 out += info.outJump;
10396 else if (info.inFormat == RTAUDIO_FLOAT64) {
10397 Float64 *in = (Float64 *)inBuffer;
10398 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10399 for (j=0; j<info.channels; j++) {
10400 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
10403 out += info.outJump;
10407 else if (info.outFormat == RTAUDIO_SINT16) {
10408 Int16 *out = (Int16 *)outBuffer;
10409 if (info.inFormat == RTAUDIO_SINT8) {
10410 signed char *in = (signed char *)inBuffer;
10411 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10412 for (j=0; j<info.channels; j++) {
10413 out[info.outOffset[j]] = (Int16) in[info.inOffset[j]];
10414 out[info.outOffset[j]] <<= 8;
10417 out += info.outJump;
10420 else if (info.inFormat == RTAUDIO_SINT16) {
10421 // Channel compensation and/or (de)interleaving only.
10422 Int16 *in = (Int16 *)inBuffer;
10423 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10424 for (j=0; j<info.channels; j++) {
10425 out[info.outOffset[j]] = in[info.inOffset[j]];
10428 out += info.outJump;
10431 else if (info.inFormat == RTAUDIO_SINT24) {
10432 Int24 *in = (Int24 *)inBuffer;
10433 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10434 for (j=0; j<info.channels; j++) {
10435 out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]].asInt() >> 8);
10438 out += info.outJump;
10441 else if (info.inFormat == RTAUDIO_SINT32) {
10442 Int32 *in = (Int32 *)inBuffer;
10443 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10444 for (j=0; j<info.channels; j++) {
10445 out[info.outOffset[j]] = (Int16) ((in[info.inOffset[j]] >> 16) & 0x0000ffff);
10448 out += info.outJump;
10451 else if (info.inFormat == RTAUDIO_FLOAT32) {
10452 Float32 *in = (Float32 *)inBuffer;
10453 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10454 for (j=0; j<info.channels; j++) {
10455 out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
10458 out += info.outJump;
10461 else if (info.inFormat == RTAUDIO_FLOAT64) {
10462 Float64 *in = (Float64 *)inBuffer;
10463 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10464 for (j=0; j<info.channels; j++) {
10465 out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
10468 out += info.outJump;
10472 else if (info.outFormat == RTAUDIO_SINT8) {
10473 signed char *out = (signed char *)outBuffer;
10474 if (info.inFormat == RTAUDIO_SINT8) {
10475 // Channel compensation and/or (de)interleaving only.
10476 signed char *in = (signed char *)inBuffer;
10477 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10478 for (j=0; j<info.channels; j++) {
10479 out[info.outOffset[j]] = in[info.inOffset[j]];
10482 out += info.outJump;
10485 if (info.inFormat == RTAUDIO_SINT16) {
10486 Int16 *in = (Int16 *)inBuffer;
10487 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10488 for (j=0; j<info.channels; j++) {
10489 out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 8) & 0x00ff);
10492 out += info.outJump;
10495 else if (info.inFormat == RTAUDIO_SINT24) {
10496 Int24 *in = (Int24 *)inBuffer;
10497 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10498 for (j=0; j<info.channels; j++) {
10499 out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]].asInt() >> 16);
10502 out += info.outJump;
10505 else if (info.inFormat == RTAUDIO_SINT32) {
10506 Int32 *in = (Int32 *)inBuffer;
10507 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10508 for (j=0; j<info.channels; j++) {
10509 out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 24) & 0x000000ff);
10512 out += info.outJump;
10515 else if (info.inFormat == RTAUDIO_FLOAT32) {
10516 Float32 *in = (Float32 *)inBuffer;
10517 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10518 for (j=0; j<info.channels; j++) {
10519 out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
10522 out += info.outJump;
10525 else if (info.inFormat == RTAUDIO_FLOAT64) {
10526 Float64 *in = (Float64 *)inBuffer;
10527 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10528 for (j=0; j<info.channels; j++) {
10529 out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
10532 out += info.outJump;
10538 //static inline uint16_t bswap_16(uint16_t x) { return (x>>8) | (x<<8); }
10539 //static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); }
10540 //static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); }
10542 void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format )
10548 if ( format == RTAUDIO_SINT16 ) {
10549 for ( unsigned int i=0; i<samples; i++ ) {
10550 // Swap 1st and 2nd bytes.
10555 // Increment 2 bytes.
10559 else if ( format == RTAUDIO_SINT32 ||
10560 format == RTAUDIO_FLOAT32 ) {
10561 for ( unsigned int i=0; i<samples; i++ ) {
10562 // Swap 1st and 4th bytes.
10567 // Swap 2nd and 3rd bytes.
10573 // Increment 3 more bytes.
10577 else if ( format == RTAUDIO_SINT24 ) {
10578 for ( unsigned int i=0; i<samples; i++ ) {
10579 // Swap 1st and 3rd bytes.
10584 // Increment 2 more bytes.
10588 else if ( format == RTAUDIO_FLOAT64 ) {
10589 for ( unsigned int i=0; i<samples; i++ ) {
10590 // Swap 1st and 8th bytes
10595 // Swap 2nd and 7th bytes
10601 // Swap 3rd and 6th bytes
10607 // Swap 4th and 5th bytes
10613 // Increment 5 more bytes.
10619 // Indentation settings for Vim and Emacs
10621 // Local Variables:
10622 // c-basic-offset: 2
10623 // indent-tabs-mode: nil
10626 // vim: et sts=2 sw=2