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 // retrieve captureAudioClient from devicePtr
4768 IAudioClient*& captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
4770 hr = renderDevices->Item( device, &devicePtr );
4771 if ( FAILED( hr ) ) {
4772 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device handle.";
4776 hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
4777 NULL, ( void** ) &captureAudioClient );
4778 if ( FAILED( hr ) ) {
4779 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device audio client.";
4783 hr = captureAudioClient->GetMixFormat( &deviceFormat );
4784 if ( FAILED( hr ) ) {
4785 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device mix format.";
4789 stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
4790 captureAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
4793 // if device index falls within render devices and is configured for output
4794 if ( device < renderDeviceCount && mode == OUTPUT )
4796 // retrieve renderAudioClient from devicePtr
4797 IAudioClient*& renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
4799 hr = renderDevices->Item( device, &devicePtr );
4800 if ( FAILED( hr ) ) {
4801 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device handle.";
4805 hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
4806 NULL, ( void** ) &renderAudioClient );
4807 if ( FAILED( hr ) ) {
4808 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device audio client.";
4812 hr = renderAudioClient->GetMixFormat( &deviceFormat );
4813 if ( FAILED( hr ) ) {
4814 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device mix format.";
4818 stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
4819 renderAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
4823 if ( ( stream_.mode == OUTPUT && mode == INPUT ) ||
4824 ( stream_.mode == INPUT && mode == OUTPUT ) ) {
4825 stream_.mode = DUPLEX;
4828 stream_.mode = mode;
4831 stream_.device[mode] = device;
4832 stream_.doByteSwap[mode] = false;
4833 stream_.sampleRate = sampleRate;
4834 stream_.bufferSize = *bufferSize;
4835 stream_.nBuffers = 1;
4836 stream_.nUserChannels[mode] = channels;
4837 stream_.channelOffset[mode] = firstChannel;
4838 stream_.userFormat = format;
4839 stream_.deviceFormat[mode] = getDeviceInfo( device ).nativeFormats;
4841 if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
4842 stream_.userInterleaved = false;
4844 stream_.userInterleaved = true;
4845 stream_.deviceInterleaved[mode] = true;
4847 // Set flags for buffer conversion.
4848 stream_.doConvertBuffer[mode] = false;
4849 if ( stream_.userFormat != stream_.deviceFormat[mode] ||
4850 stream_.nUserChannels[0] != stream_.nDeviceChannels[0] ||
4851 stream_.nUserChannels[1] != stream_.nDeviceChannels[1] )
4852 stream_.doConvertBuffer[mode] = true;
4853 else if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
4854 stream_.nUserChannels[mode] > 1 )
4855 stream_.doConvertBuffer[mode] = true;
4857 if ( stream_.doConvertBuffer[mode] )
4858 setConvertInfo( mode, 0 );
4860 // Allocate necessary internal buffers
4861 bufferBytes = stream_.nUserChannels[mode] * stream_.bufferSize * formatBytes( stream_.userFormat );
4863 stream_.userBuffer[mode] = ( char* ) calloc( bufferBytes, 1 );
4864 if ( !stream_.userBuffer[mode] ) {
4865 errorType = RtAudioError::MEMORY_ERROR;
4866 errorText_ = "RtApiWasapi::probeDeviceOpen: Error allocating user buffer memory.";
4870 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME )
4871 stream_.callbackInfo.priority = 15;
4873 stream_.callbackInfo.priority = 0;
4875 ///! TODO: RTAUDIO_MINIMIZE_LATENCY // Provide stream buffers directly to callback
4876 ///! TODO: RTAUDIO_HOG_DEVICE // Exclusive mode
4878 methodResult = SUCCESS;
4882 SAFE_RELEASE( captureDevices );
4883 SAFE_RELEASE( renderDevices );
4884 SAFE_RELEASE( devicePtr );
4885 CoTaskMemFree( deviceFormat );
4887 // if method failed, close the stream
4888 if ( methodResult == FAILURE )
4891 if ( !errorText_.empty() )
4893 return methodResult;
4896 //=============================================================================
4898 DWORD WINAPI RtApiWasapi::runWasapiThread( void* wasapiPtr )
4901 ( ( RtApiWasapi* ) wasapiPtr )->wasapiThread();
4906 DWORD WINAPI RtApiWasapi::stopWasapiThread( void* wasapiPtr )
4909 ( ( RtApiWasapi* ) wasapiPtr )->stopStream();
4914 DWORD WINAPI RtApiWasapi::abortWasapiThread( void* wasapiPtr )
4917 ( ( RtApiWasapi* ) wasapiPtr )->abortStream();
4922 //-----------------------------------------------------------------------------
4924 void RtApiWasapi::wasapiThread()
4926 // as this is a new thread, we must CoInitialize it
4927 CoInitialize( NULL );
4931 IAudioClient* captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
4932 IAudioClient* renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
4933 IAudioCaptureClient* captureClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureClient;
4934 IAudioRenderClient* renderClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderClient;
4935 HANDLE captureEvent = ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent;
4936 HANDLE renderEvent = ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent;
4938 WAVEFORMATEX* captureFormat = NULL;
4939 WAVEFORMATEX* renderFormat = NULL;
4940 float captureSrRatio = 0.0f;
4941 float renderSrRatio = 0.0f;
4942 WasapiBuffer captureBuffer;
4943 WasapiBuffer renderBuffer;
4944 WasapiResampler* captureResampler = NULL;
4945 WasapiResampler* renderResampler = NULL;
4947 // declare local stream variables
4948 RtAudioCallback callback = ( RtAudioCallback ) stream_.callbackInfo.callback;
4949 BYTE* streamBuffer = NULL;
4950 unsigned long captureFlags = 0;
4951 unsigned int bufferFrameCount = 0;
4952 unsigned int numFramesPadding = 0;
4953 unsigned int convBufferSize = 0;
4954 bool loopbackEnabled = stream_.device[INPUT] == stream_.device[OUTPUT];
4955 bool callbackPushed = true;
4956 bool callbackPulled = false;
4957 bool callbackStopped = false;
4958 int callbackResult = 0;
4960 // convBuffer is used to store converted buffers between WASAPI and the user
4961 char* convBuffer = NULL;
4962 unsigned int convBuffSize = 0;
4963 unsigned int deviceBuffSize = 0;
4966 RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
4968 // Attempt to assign "Pro Audio" characteristic to thread
4969 HMODULE AvrtDll = LoadLibrary( (LPCTSTR) "AVRT.dll" );
4971 DWORD taskIndex = 0;
4972 TAvSetMmThreadCharacteristicsPtr AvSetMmThreadCharacteristicsPtr = ( TAvSetMmThreadCharacteristicsPtr ) GetProcAddress( AvrtDll, "AvSetMmThreadCharacteristicsW" );
4973 AvSetMmThreadCharacteristicsPtr( L"Pro Audio", &taskIndex );
4974 FreeLibrary( AvrtDll );
4977 // start capture stream if applicable
4978 if ( captureAudioClient ) {
4979 hr = captureAudioClient->GetMixFormat( &captureFormat );
4980 if ( FAILED( hr ) ) {
4981 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
4985 // init captureResampler
4986 captureResampler = new WasapiResampler( stream_.deviceFormat[INPUT] == RTAUDIO_FLOAT32 || stream_.deviceFormat[INPUT] == RTAUDIO_FLOAT64,
4987 formatBytes( stream_.deviceFormat[INPUT] ) * 8, stream_.nDeviceChannels[INPUT],
4988 captureFormat->nSamplesPerSec, stream_.sampleRate );
4990 captureSrRatio = ( ( float ) captureFormat->nSamplesPerSec / stream_.sampleRate );
4992 if ( !captureClient ) {
4993 hr = captureAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
4994 loopbackEnabled ? AUDCLNT_STREAMFLAGS_LOOPBACK : AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
4999 if ( FAILED( hr ) ) {
5000 errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize capture audio client.";
5004 hr = captureAudioClient->GetService( __uuidof( IAudioCaptureClient ),
5005 ( void** ) &captureClient );
5006 if ( FAILED( hr ) ) {
5007 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture client handle.";
5011 // don't configure captureEvent if in loopback mode
5012 if ( !loopbackEnabled )
5014 // configure captureEvent to trigger on every available capture buffer
5015 captureEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
5016 if ( !captureEvent ) {
5017 errorType = RtAudioError::SYSTEM_ERROR;
5018 errorText_ = "RtApiWasapi::wasapiThread: Unable to create capture event.";
5022 hr = captureAudioClient->SetEventHandle( captureEvent );
5023 if ( FAILED( hr ) ) {
5024 errorText_ = "RtApiWasapi::wasapiThread: Unable to set capture event handle.";
5028 ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent = captureEvent;
5031 ( ( WasapiHandle* ) stream_.apiHandle )->captureClient = captureClient;
5034 unsigned int inBufferSize = 0;
5035 hr = captureAudioClient->GetBufferSize( &inBufferSize );
5036 if ( FAILED( hr ) ) {
5037 errorText_ = "RtApiWasapi::wasapiThread: Unable to get capture buffer size.";
5041 // scale outBufferSize according to stream->user sample rate ratio
5042 unsigned int outBufferSize = ( unsigned int ) ceilf( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT];
5043 inBufferSize *= stream_.nDeviceChannels[INPUT];
5045 // set captureBuffer size
5046 captureBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[INPUT] ) );
5048 // reset the capture stream
5049 hr = captureAudioClient->Reset();
5050 if ( FAILED( hr ) ) {
5051 errorText_ = "RtApiWasapi::wasapiThread: Unable to reset capture stream.";
5055 // start the capture stream
5056 hr = captureAudioClient->Start();
5057 if ( FAILED( hr ) ) {
5058 errorText_ = "RtApiWasapi::wasapiThread: Unable to start capture stream.";
5063 // start render stream if applicable
5064 if ( renderAudioClient ) {
5065 hr = renderAudioClient->GetMixFormat( &renderFormat );
5066 if ( FAILED( hr ) ) {
5067 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
5071 // init renderResampler
5072 renderResampler = new WasapiResampler( stream_.deviceFormat[OUTPUT] == RTAUDIO_FLOAT32 || stream_.deviceFormat[OUTPUT] == RTAUDIO_FLOAT64,
5073 formatBytes( stream_.deviceFormat[OUTPUT] ) * 8, stream_.nDeviceChannels[OUTPUT],
5074 stream_.sampleRate, renderFormat->nSamplesPerSec );
5076 renderSrRatio = ( ( float ) renderFormat->nSamplesPerSec / stream_.sampleRate );
5078 if ( !renderClient ) {
5079 hr = renderAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
5080 AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
5085 if ( FAILED( hr ) ) {
5086 errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize render audio client.";
5090 hr = renderAudioClient->GetService( __uuidof( IAudioRenderClient ),
5091 ( void** ) &renderClient );
5092 if ( FAILED( hr ) ) {
5093 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render client handle.";
5097 // configure renderEvent to trigger on every available render buffer
5098 renderEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
5099 if ( !renderEvent ) {
5100 errorType = RtAudioError::SYSTEM_ERROR;
5101 errorText_ = "RtApiWasapi::wasapiThread: Unable to create render event.";
5105 hr = renderAudioClient->SetEventHandle( renderEvent );
5106 if ( FAILED( hr ) ) {
5107 errorText_ = "RtApiWasapi::wasapiThread: Unable to set render event handle.";
5111 ( ( WasapiHandle* ) stream_.apiHandle )->renderClient = renderClient;
5112 ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent = renderEvent;
5115 unsigned int outBufferSize = 0;
5116 hr = renderAudioClient->GetBufferSize( &outBufferSize );
5117 if ( FAILED( hr ) ) {
5118 errorText_ = "RtApiWasapi::wasapiThread: Unable to get render buffer size.";
5122 // scale inBufferSize according to user->stream sample rate ratio
5123 unsigned int inBufferSize = ( unsigned int ) ceilf( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT];
5124 outBufferSize *= stream_.nDeviceChannels[OUTPUT];
5126 // set renderBuffer size
5127 renderBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[OUTPUT] ) );
5129 // reset the render stream
5130 hr = renderAudioClient->Reset();
5131 if ( FAILED( hr ) ) {
5132 errorText_ = "RtApiWasapi::wasapiThread: Unable to reset render stream.";
5136 // start the render stream
5137 hr = renderAudioClient->Start();
5138 if ( FAILED( hr ) ) {
5139 errorText_ = "RtApiWasapi::wasapiThread: Unable to start render stream.";
5144 // malloc buffer memory
5145 if ( stream_.mode == INPUT )
5147 using namespace std; // for ceilf
5148 convBuffSize = ( size_t ) ( ceilf( stream_.bufferSize * captureSrRatio ) ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
5149 deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
5151 else if ( stream_.mode == OUTPUT )
5153 convBuffSize = ( size_t ) ( ceilf( stream_.bufferSize * renderSrRatio ) ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
5154 deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
5156 else if ( stream_.mode == DUPLEX )
5158 convBuffSize = std::max( ( size_t ) ( ceilf( stream_.bufferSize * captureSrRatio ) ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
5159 ( size_t ) ( ceilf( stream_.bufferSize * renderSrRatio ) ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
5160 deviceBuffSize = std::max( stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
5161 stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
5164 convBuffSize *= 2; // allow overflow for *SrRatio remainders
5165 convBuffer = ( char* ) malloc( convBuffSize );
5166 stream_.deviceBuffer = ( char* ) malloc( deviceBuffSize );
5167 if ( !convBuffer || !stream_.deviceBuffer ) {
5168 errorType = RtAudioError::MEMORY_ERROR;
5169 errorText_ = "RtApiWasapi::wasapiThread: Error allocating device buffer memory.";
5173 // stream process loop
5174 while ( stream_.state != STREAM_STOPPING ) {
5175 if ( !callbackPulled ) {
5178 // 1. Pull callback buffer from inputBuffer
5179 // 2. If 1. was successful: Convert callback buffer to user sample rate and channel count
5180 // Convert callback buffer to user format
5182 if ( captureAudioClient )
5184 int samplesToPull = ( unsigned int ) floorf( stream_.bufferSize * captureSrRatio );
5185 if ( captureSrRatio != 1 )
5187 // account for remainders
5192 while ( convBufferSize < stream_.bufferSize )
5194 // Pull callback buffer from inputBuffer
5195 callbackPulled = captureBuffer.pullBuffer( convBuffer,
5196 samplesToPull * stream_.nDeviceChannels[INPUT],
5197 stream_.deviceFormat[INPUT] );
5199 if ( !callbackPulled )
5204 // Convert callback buffer to user sample rate
5205 unsigned int deviceBufferOffset = convBufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
5206 unsigned int convSamples = 0;
5208 captureResampler->Convert( stream_.deviceBuffer + deviceBufferOffset,
5213 convBufferSize += convSamples;
5214 samplesToPull = 1; // now pull one sample at a time until we have stream_.bufferSize samples
5217 if ( callbackPulled )
5219 if ( stream_.doConvertBuffer[INPUT] ) {
5220 // Convert callback buffer to user format
5221 convertBuffer( stream_.userBuffer[INPUT],
5222 stream_.deviceBuffer,
5223 stream_.convertInfo[INPUT] );
5226 // no further conversion, simple copy deviceBuffer to userBuffer
5227 memcpy( stream_.userBuffer[INPUT],
5228 stream_.deviceBuffer,
5229 stream_.bufferSize * stream_.nUserChannels[INPUT] * formatBytes( stream_.userFormat ) );
5234 // if there is no capture stream, set callbackPulled flag
5235 callbackPulled = true;
5240 // 1. Execute user callback method
5241 // 2. Handle return value from callback
5243 // if callback has not requested the stream to stop
5244 if ( callbackPulled && !callbackStopped ) {
5245 // Execute user callback method
5246 callbackResult = callback( stream_.userBuffer[OUTPUT],
5247 stream_.userBuffer[INPUT],
5250 captureFlags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY ? RTAUDIO_INPUT_OVERFLOW : 0,
5251 stream_.callbackInfo.userData );
5253 // Handle return value from callback
5254 if ( callbackResult == 1 ) {
5255 // instantiate a thread to stop this thread
5256 HANDLE threadHandle = CreateThread( NULL, 0, stopWasapiThread, this, 0, NULL );
5257 if ( !threadHandle ) {
5258 errorType = RtAudioError::THREAD_ERROR;
5259 errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream stop thread.";
5262 else if ( !CloseHandle( threadHandle ) ) {
5263 errorType = RtAudioError::THREAD_ERROR;
5264 errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream stop thread handle.";
5268 callbackStopped = true;
5270 else if ( callbackResult == 2 ) {
5271 // instantiate a thread to stop this thread
5272 HANDLE threadHandle = CreateThread( NULL, 0, abortWasapiThread, this, 0, NULL );
5273 if ( !threadHandle ) {
5274 errorType = RtAudioError::THREAD_ERROR;
5275 errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream abort thread.";
5278 else if ( !CloseHandle( threadHandle ) ) {
5279 errorType = RtAudioError::THREAD_ERROR;
5280 errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream abort thread handle.";
5284 callbackStopped = true;
5291 // 1. Convert callback buffer to stream format
5292 // 2. Convert callback buffer to stream sample rate and channel count
5293 // 3. Push callback buffer into outputBuffer
5295 if ( renderAudioClient && callbackPulled )
5297 // if the last call to renderBuffer.PushBuffer() was successful
5298 if ( callbackPushed || convBufferSize == 0 )
5300 if ( stream_.doConvertBuffer[OUTPUT] )
5302 // Convert callback buffer to stream format
5303 convertBuffer( stream_.deviceBuffer,
5304 stream_.userBuffer[OUTPUT],
5305 stream_.convertInfo[OUTPUT] );
5309 // Convert callback buffer to stream sample rate
5310 renderResampler->Convert( convBuffer,
5311 stream_.deviceBuffer,
5316 // Push callback buffer into outputBuffer
5317 callbackPushed = renderBuffer.pushBuffer( convBuffer,
5318 convBufferSize * stream_.nDeviceChannels[OUTPUT],
5319 stream_.deviceFormat[OUTPUT] );
5322 // if there is no render stream, set callbackPushed flag
5323 callbackPushed = true;
5328 // 1. Get capture buffer from stream
5329 // 2. Push capture buffer into inputBuffer
5330 // 3. If 2. was successful: Release capture buffer
5332 if ( captureAudioClient ) {
5333 // if the callback input buffer was not pulled from captureBuffer, wait for next capture event
5334 if ( !callbackPulled ) {
5335 WaitForSingleObject( loopbackEnabled ? renderEvent : captureEvent, INFINITE );
5338 // Get capture buffer from stream
5339 hr = captureClient->GetBuffer( &streamBuffer,
5341 &captureFlags, NULL, NULL );
5342 if ( FAILED( hr ) ) {
5343 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture buffer.";
5347 if ( bufferFrameCount != 0 ) {
5348 // Push capture buffer into inputBuffer
5349 if ( captureBuffer.pushBuffer( ( char* ) streamBuffer,
5350 bufferFrameCount * stream_.nDeviceChannels[INPUT],
5351 stream_.deviceFormat[INPUT] ) )
5353 // Release capture buffer
5354 hr = captureClient->ReleaseBuffer( bufferFrameCount );
5355 if ( FAILED( hr ) ) {
5356 errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
5362 // Inform WASAPI that capture was unsuccessful
5363 hr = captureClient->ReleaseBuffer( 0 );
5364 if ( FAILED( hr ) ) {
5365 errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
5372 // Inform WASAPI that capture was unsuccessful
5373 hr = captureClient->ReleaseBuffer( 0 );
5374 if ( FAILED( hr ) ) {
5375 errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
5383 // 1. Get render buffer from stream
5384 // 2. Pull next buffer from outputBuffer
5385 // 3. If 2. was successful: Fill render buffer with next buffer
5386 // Release render buffer
5388 if ( renderAudioClient ) {
5389 // if the callback output buffer was not pushed to renderBuffer, wait for next render event
5390 if ( callbackPulled && !callbackPushed ) {
5391 WaitForSingleObject( renderEvent, INFINITE );
5394 // Get render buffer from stream
5395 hr = renderAudioClient->GetBufferSize( &bufferFrameCount );
5396 if ( FAILED( hr ) ) {
5397 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer size.";
5401 hr = renderAudioClient->GetCurrentPadding( &numFramesPadding );
5402 if ( FAILED( hr ) ) {
5403 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer padding.";
5407 bufferFrameCount -= numFramesPadding;
5409 if ( bufferFrameCount != 0 ) {
5410 hr = renderClient->GetBuffer( bufferFrameCount, &streamBuffer );
5411 if ( FAILED( hr ) ) {
5412 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer.";
5416 // Pull next buffer from outputBuffer
5417 // Fill render buffer with next buffer
5418 if ( renderBuffer.pullBuffer( ( char* ) streamBuffer,
5419 bufferFrameCount * stream_.nDeviceChannels[OUTPUT],
5420 stream_.deviceFormat[OUTPUT] ) )
5422 // Release render buffer
5423 hr = renderClient->ReleaseBuffer( bufferFrameCount, 0 );
5424 if ( FAILED( hr ) ) {
5425 errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
5431 // Inform WASAPI that render was unsuccessful
5432 hr = renderClient->ReleaseBuffer( 0, 0 );
5433 if ( FAILED( hr ) ) {
5434 errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
5441 // Inform WASAPI that render was unsuccessful
5442 hr = renderClient->ReleaseBuffer( 0, 0 );
5443 if ( FAILED( hr ) ) {
5444 errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
5450 // if the callback buffer was pushed renderBuffer reset callbackPulled flag
5451 if ( callbackPushed ) {
5452 // unsetting the callbackPulled flag lets the stream know that
5453 // the audio device is ready for another callback output buffer.
5454 callbackPulled = false;
5457 RtApi::tickStreamTime();
5464 CoTaskMemFree( captureFormat );
5465 CoTaskMemFree( renderFormat );
5467 free ( convBuffer );
5468 delete renderResampler;
5469 delete captureResampler;
5473 if ( !errorText_.empty() )
5476 // update stream state
5477 stream_.state = STREAM_STOPPED;
5480 //******************** End of __WINDOWS_WASAPI__ *********************//
5484 #if defined(__WINDOWS_DS__) // Windows DirectSound API
5486 // Modified by Robin Davies, October 2005
5487 // - Improvements to DirectX pointer chasing.
5488 // - Bug fix for non-power-of-two Asio granularity used by Edirol PCR-A30.
5489 // - Auto-call CoInitialize for DSOUND and ASIO platforms.
5490 // Various revisions for RtAudio 4.0 by Gary Scavone, April 2007
5491 // Changed device query structure for RtAudio 4.0.7, January 2010
5493 #include <windows.h>
5494 #include <process.h>
5495 #include <mmsystem.h>
5499 #include <algorithm>
5501 #if defined(__MINGW32__)
5502 // missing from latest mingw winapi
5503 #define WAVE_FORMAT_96M08 0x00010000 /* 96 kHz, Mono, 8-bit */
5504 #define WAVE_FORMAT_96S08 0x00020000 /* 96 kHz, Stereo, 8-bit */
5505 #define WAVE_FORMAT_96M16 0x00040000 /* 96 kHz, Mono, 16-bit */
5506 #define WAVE_FORMAT_96S16 0x00080000 /* 96 kHz, Stereo, 16-bit */
5509 #define MINIMUM_DEVICE_BUFFER_SIZE 32768
5511 #ifdef _MSC_VER // if Microsoft Visual C++
5512 #pragma comment( lib, "winmm.lib" ) // then, auto-link winmm.lib. Otherwise, it has to be added manually.
5515 static inline DWORD dsPointerBetween( DWORD pointer, DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize )
5517 if ( pointer > bufferSize ) pointer -= bufferSize;
5518 if ( laterPointer < earlierPointer ) laterPointer += bufferSize;
5519 if ( pointer < earlierPointer ) pointer += bufferSize;
5520 return pointer >= earlierPointer && pointer < laterPointer;
5523 // A structure to hold various information related to the DirectSound
5524 // API implementation.
5526 unsigned int drainCounter; // Tracks callback counts when draining
5527 bool internalDrain; // Indicates if stop is initiated from callback or not.
5531 UINT bufferPointer[2];
5532 DWORD dsBufferSize[2];
5533 DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by.
5537 :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; }
5540 // Declarations for utility functions, callbacks, and structures
5541 // specific to the DirectSound implementation.
5542 static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
5543 LPCTSTR description,
5547 static const char* getErrorString( int code );
5549 static unsigned __stdcall callbackHandler( void *ptr );
5558 : found(false) { validId[0] = false; validId[1] = false; }
5561 struct DsProbeData {
5563 std::vector<struct DsDevice>* dsDevices;
5566 RtApiDs :: RtApiDs()
5568 // Dsound will run both-threaded. If CoInitialize fails, then just
5569 // accept whatever the mainline chose for a threading model.
5570 coInitialized_ = false;
5571 HRESULT hr = CoInitialize( NULL );
5572 if ( !FAILED( hr ) ) coInitialized_ = true;
5575 RtApiDs :: ~RtApiDs()
5577 if ( stream_.state != STREAM_CLOSED ) closeStream();
5578 if ( coInitialized_ ) CoUninitialize(); // balanced call.
5581 // The DirectSound default output is always the first device.
5582 unsigned int RtApiDs :: getDefaultOutputDevice( void )
5587 // The DirectSound default input is always the first input device,
5588 // which is the first capture device enumerated.
5589 unsigned int RtApiDs :: getDefaultInputDevice( void )
5594 unsigned int RtApiDs :: getDeviceCount( void )
5596 // Set query flag for previously found devices to false, so that we
5597 // can check for any devices that have disappeared.
5598 for ( unsigned int i=0; i<dsDevices.size(); i++ )
5599 dsDevices[i].found = false;
5601 // Query DirectSound devices.
5602 struct DsProbeData probeInfo;
5603 probeInfo.isInput = false;
5604 probeInfo.dsDevices = &dsDevices;
5605 HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );
5606 if ( FAILED( result ) ) {
5607 errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!";
5608 errorText_ = errorStream_.str();
5609 error( RtAudioError::WARNING );
5612 // Query DirectSoundCapture devices.
5613 probeInfo.isInput = true;
5614 result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );
5615 if ( FAILED( result ) ) {
5616 errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!";
5617 errorText_ = errorStream_.str();
5618 error( RtAudioError::WARNING );
5621 // Clean out any devices that may have disappeared (code update submitted by Eli Zehngut).
5622 for ( unsigned int i=0; i<dsDevices.size(); ) {
5623 if ( dsDevices[i].found == false ) dsDevices.erase( dsDevices.begin() + i );
5627 return static_cast<unsigned int>(dsDevices.size());
5630 RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device )
5632 RtAudio::DeviceInfo info;
5633 info.probed = false;
5635 if ( dsDevices.size() == 0 ) {
5636 // Force a query of all devices
5638 if ( dsDevices.size() == 0 ) {
5639 errorText_ = "RtApiDs::getDeviceInfo: no devices found!";
5640 error( RtAudioError::INVALID_USE );
5645 if ( device >= dsDevices.size() ) {
5646 errorText_ = "RtApiDs::getDeviceInfo: device ID is invalid!";
5647 error( RtAudioError::INVALID_USE );
5652 if ( dsDevices[ device ].validId[0] == false ) goto probeInput;
5654 LPDIRECTSOUND output;
5656 result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );
5657 if ( FAILED( result ) ) {
5658 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!";
5659 errorText_ = errorStream_.str();
5660 error( RtAudioError::WARNING );
5664 outCaps.dwSize = sizeof( outCaps );
5665 result = output->GetCaps( &outCaps );
5666 if ( FAILED( result ) ) {
5668 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting capabilities!";
5669 errorText_ = errorStream_.str();
5670 error( RtAudioError::WARNING );
5674 // Get output channel information.
5675 info.outputChannels = ( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1;
5677 // Get sample rate information.
5678 info.sampleRates.clear();
5679 for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
5680 if ( SAMPLE_RATES[k] >= (unsigned int) outCaps.dwMinSecondarySampleRate &&
5681 SAMPLE_RATES[k] <= (unsigned int) outCaps.dwMaxSecondarySampleRate ) {
5682 info.sampleRates.push_back( SAMPLE_RATES[k] );
5684 if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
5685 info.preferredSampleRate = SAMPLE_RATES[k];
5689 // Get format information.
5690 if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT ) info.nativeFormats |= RTAUDIO_SINT16;
5691 if ( outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) info.nativeFormats |= RTAUDIO_SINT8;
5695 if ( getDefaultOutputDevice() == device )
5696 info.isDefaultOutput = true;
5698 if ( dsDevices[ device ].validId[1] == false ) {
5699 info.name = dsDevices[ device ].name;
5706 LPDIRECTSOUNDCAPTURE input;
5707 result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );
5708 if ( FAILED( result ) ) {
5709 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";
5710 errorText_ = errorStream_.str();
5711 error( RtAudioError::WARNING );
5716 inCaps.dwSize = sizeof( inCaps );
5717 result = input->GetCaps( &inCaps );
5718 if ( FAILED( result ) ) {
5720 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsDevices[ device ].name << ")!";
5721 errorText_ = errorStream_.str();
5722 error( RtAudioError::WARNING );
5726 // Get input channel information.
5727 info.inputChannels = inCaps.dwChannels;
5729 // Get sample rate and format information.
5730 std::vector<unsigned int> rates;
5731 if ( inCaps.dwChannels >= 2 ) {
5732 if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.nativeFormats |= RTAUDIO_SINT16;
5733 if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.nativeFormats |= RTAUDIO_SINT16;
5734 if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.nativeFormats |= RTAUDIO_SINT16;
5735 if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.nativeFormats |= RTAUDIO_SINT16;
5736 if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.nativeFormats |= RTAUDIO_SINT8;
5737 if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.nativeFormats |= RTAUDIO_SINT8;
5738 if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.nativeFormats |= RTAUDIO_SINT8;
5739 if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.nativeFormats |= RTAUDIO_SINT8;
5741 if ( info.nativeFormats & RTAUDIO_SINT16 ) {
5742 if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) rates.push_back( 11025 );
5743 if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) rates.push_back( 22050 );
5744 if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) rates.push_back( 44100 );
5745 if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) rates.push_back( 96000 );
5747 else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
5748 if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) rates.push_back( 11025 );
5749 if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) rates.push_back( 22050 );
5750 if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) rates.push_back( 44100 );
5751 if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) rates.push_back( 96000 );
5754 else if ( inCaps.dwChannels == 1 ) {
5755 if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.nativeFormats |= RTAUDIO_SINT16;
5756 if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.nativeFormats |= RTAUDIO_SINT16;
5757 if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.nativeFormats |= RTAUDIO_SINT16;
5758 if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.nativeFormats |= RTAUDIO_SINT16;
5759 if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.nativeFormats |= RTAUDIO_SINT8;
5760 if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.nativeFormats |= RTAUDIO_SINT8;
5761 if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.nativeFormats |= RTAUDIO_SINT8;
5762 if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.nativeFormats |= RTAUDIO_SINT8;
5764 if ( info.nativeFormats & RTAUDIO_SINT16 ) {
5765 if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) rates.push_back( 11025 );
5766 if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) rates.push_back( 22050 );
5767 if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) rates.push_back( 44100 );
5768 if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) rates.push_back( 96000 );
5770 else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
5771 if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) rates.push_back( 11025 );
5772 if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) rates.push_back( 22050 );
5773 if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) rates.push_back( 44100 );
5774 if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) rates.push_back( 96000 );
5777 else info.inputChannels = 0; // technically, this would be an error
5781 if ( info.inputChannels == 0 ) return info;
5783 // Copy the supported rates to the info structure but avoid duplication.
5785 for ( unsigned int i=0; i<rates.size(); i++ ) {
5787 for ( unsigned int j=0; j<info.sampleRates.size(); j++ ) {
5788 if ( rates[i] == info.sampleRates[j] ) {
5793 if ( found == false ) info.sampleRates.push_back( rates[i] );
5795 std::sort( info.sampleRates.begin(), info.sampleRates.end() );
5797 // If device opens for both playback and capture, we determine the channels.
5798 if ( info.outputChannels > 0 && info.inputChannels > 0 )
5799 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
5801 if ( device == 0 ) info.isDefaultInput = true;
5803 // Copy name and return.
5804 info.name = dsDevices[ device ].name;
5809 bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
5810 unsigned int firstChannel, unsigned int sampleRate,
5811 RtAudioFormat format, unsigned int *bufferSize,
5812 RtAudio::StreamOptions *options )
5814 if ( channels + firstChannel > 2 ) {
5815 errorText_ = "RtApiDs::probeDeviceOpen: DirectSound does not support more than 2 channels per device.";
5819 size_t nDevices = dsDevices.size();
5820 if ( nDevices == 0 ) {
5821 // This should not happen because a check is made before this function is called.
5822 errorText_ = "RtApiDs::probeDeviceOpen: no devices found!";
5826 if ( device >= nDevices ) {
5827 // This should not happen because a check is made before this function is called.
5828 errorText_ = "RtApiDs::probeDeviceOpen: device ID is invalid!";
5832 if ( mode == OUTPUT ) {
5833 if ( dsDevices[ device ].validId[0] == false ) {
5834 errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support output!";
5835 errorText_ = errorStream_.str();
5839 else { // mode == INPUT
5840 if ( dsDevices[ device ].validId[1] == false ) {
5841 errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support input!";
5842 errorText_ = errorStream_.str();
5847 // According to a note in PortAudio, using GetDesktopWindow()
5848 // instead of GetForegroundWindow() is supposed to avoid problems
5849 // that occur when the application's window is not the foreground
5850 // window. Also, if the application window closes before the
5851 // DirectSound buffer, DirectSound can crash. In the past, I had
5852 // problems when using GetDesktopWindow() but it seems fine now
5853 // (January 2010). I'll leave it commented here.
5854 // HWND hWnd = GetForegroundWindow();
5855 HWND hWnd = GetDesktopWindow();
5857 // Check the numberOfBuffers parameter and limit the lowest value to
5858 // two. This is a judgement call and a value of two is probably too
5859 // low for capture, but it should work for playback.
5861 if ( options ) nBuffers = options->numberOfBuffers;
5862 if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) nBuffers = 2;
5863 if ( nBuffers < 2 ) nBuffers = 3;
5865 // Check the lower range of the user-specified buffer size and set
5866 // (arbitrarily) to a lower bound of 32.
5867 if ( *bufferSize < 32 ) *bufferSize = 32;
5869 // Create the wave format structure. The data format setting will
5870 // be determined later.
5871 WAVEFORMATEX waveFormat;
5872 ZeroMemory( &waveFormat, sizeof(WAVEFORMATEX) );
5873 waveFormat.wFormatTag = WAVE_FORMAT_PCM;
5874 waveFormat.nChannels = channels + firstChannel;
5875 waveFormat.nSamplesPerSec = (unsigned long) sampleRate;
5877 // Determine the device buffer size. By default, we'll use the value
5878 // defined above (32K), but we will grow it to make allowances for
5879 // very large software buffer sizes.
5880 DWORD dsBufferSize = MINIMUM_DEVICE_BUFFER_SIZE;
5881 DWORD dsPointerLeadTime = 0;
5883 void *ohandle = 0, *bhandle = 0;
5885 if ( mode == OUTPUT ) {
5887 LPDIRECTSOUND output;
5888 result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );
5889 if ( FAILED( result ) ) {
5890 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!";
5891 errorText_ = errorStream_.str();
5896 outCaps.dwSize = sizeof( outCaps );
5897 result = output->GetCaps( &outCaps );
5898 if ( FAILED( result ) ) {
5900 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting capabilities (" << dsDevices[ device ].name << ")!";
5901 errorText_ = errorStream_.str();
5905 // Check channel information.
5906 if ( channels + firstChannel == 2 && !( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ) {
5907 errorStream_ << "RtApiDs::getDeviceInfo: the output device (" << dsDevices[ device ].name << ") does not support stereo playback.";
5908 errorText_ = errorStream_.str();
5912 // Check format information. Use 16-bit format unless not
5913 // supported or user requests 8-bit.
5914 if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT &&
5915 !( format == RTAUDIO_SINT8 && outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) ) {
5916 waveFormat.wBitsPerSample = 16;
5917 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
5920 waveFormat.wBitsPerSample = 8;
5921 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
5923 stream_.userFormat = format;
5925 // Update wave format structure and buffer information.
5926 waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
5927 waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
5928 dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
5930 // If the user wants an even bigger buffer, increase the device buffer size accordingly.
5931 while ( dsPointerLeadTime * 2U > dsBufferSize )
5934 // Set cooperative level to DSSCL_EXCLUSIVE ... sound stops when window focus changes.
5935 // result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE );
5936 // Set cooperative level to DSSCL_PRIORITY ... sound remains when window focus changes.
5937 result = output->SetCooperativeLevel( hWnd, DSSCL_PRIORITY );
5938 if ( FAILED( result ) ) {
5940 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting cooperative level (" << dsDevices[ device ].name << ")!";
5941 errorText_ = errorStream_.str();
5945 // Even though we will write to the secondary buffer, we need to
5946 // access the primary buffer to set the correct output format
5947 // (since the default is 8-bit, 22 kHz!). Setup the DS primary
5948 // buffer description.
5949 DSBUFFERDESC bufferDescription;
5950 ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
5951 bufferDescription.dwSize = sizeof( DSBUFFERDESC );
5952 bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER;
5954 // Obtain the primary buffer
5955 LPDIRECTSOUNDBUFFER buffer;
5956 result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
5957 if ( FAILED( result ) ) {
5959 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") accessing primary buffer (" << dsDevices[ device ].name << ")!";
5960 errorText_ = errorStream_.str();
5964 // Set the primary DS buffer sound format.
5965 result = buffer->SetFormat( &waveFormat );
5966 if ( FAILED( result ) ) {
5968 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting primary buffer format (" << dsDevices[ device ].name << ")!";
5969 errorText_ = errorStream_.str();
5973 // Setup the secondary DS buffer description.
5974 ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
5975 bufferDescription.dwSize = sizeof( DSBUFFERDESC );
5976 bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
5977 DSBCAPS_GLOBALFOCUS |
5978 DSBCAPS_GETCURRENTPOSITION2 |
5979 DSBCAPS_LOCHARDWARE ); // Force hardware mixing
5980 bufferDescription.dwBufferBytes = dsBufferSize;
5981 bufferDescription.lpwfxFormat = &waveFormat;
5983 // Try to create the secondary DS buffer. If that doesn't work,
5984 // try to use software mixing. Otherwise, there's a problem.
5985 result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
5986 if ( FAILED( result ) ) {
5987 bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
5988 DSBCAPS_GLOBALFOCUS |
5989 DSBCAPS_GETCURRENTPOSITION2 |
5990 DSBCAPS_LOCSOFTWARE ); // Force software mixing
5991 result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
5992 if ( FAILED( result ) ) {
5994 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating secondary buffer (" << dsDevices[ device ].name << ")!";
5995 errorText_ = errorStream_.str();
6000 // Get the buffer size ... might be different from what we specified.
6002 dsbcaps.dwSize = sizeof( DSBCAPS );
6003 result = buffer->GetCaps( &dsbcaps );
6004 if ( FAILED( result ) ) {
6007 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";
6008 errorText_ = errorStream_.str();
6012 dsBufferSize = dsbcaps.dwBufferBytes;
6014 // Lock the DS buffer
6017 result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );
6018 if ( FAILED( result ) ) {
6021 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking buffer (" << dsDevices[ device ].name << ")!";
6022 errorText_ = errorStream_.str();
6026 // Zero the DS buffer
6027 ZeroMemory( audioPtr, dataLen );
6029 // Unlock the DS buffer
6030 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
6031 if ( FAILED( result ) ) {
6034 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking buffer (" << dsDevices[ device ].name << ")!";
6035 errorText_ = errorStream_.str();
6039 ohandle = (void *) output;
6040 bhandle = (void *) buffer;
6043 if ( mode == INPUT ) {
6045 LPDIRECTSOUNDCAPTURE input;
6046 result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );
6047 if ( FAILED( result ) ) {
6048 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";
6049 errorText_ = errorStream_.str();
6054 inCaps.dwSize = sizeof( inCaps );
6055 result = input->GetCaps( &inCaps );
6056 if ( FAILED( result ) ) {
6058 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsDevices[ device ].name << ")!";
6059 errorText_ = errorStream_.str();
6063 // Check channel information.
6064 if ( inCaps.dwChannels < channels + firstChannel ) {
6065 errorText_ = "RtApiDs::getDeviceInfo: the input device does not support requested input channels.";
6069 // Check format information. Use 16-bit format unless user
6071 DWORD deviceFormats;
6072 if ( channels + firstChannel == 2 ) {
6073 deviceFormats = WAVE_FORMAT_1S08 | WAVE_FORMAT_2S08 | WAVE_FORMAT_4S08 | WAVE_FORMAT_96S08;
6074 if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
6075 waveFormat.wBitsPerSample = 8;
6076 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
6078 else { // assume 16-bit is supported
6079 waveFormat.wBitsPerSample = 16;
6080 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
6083 else { // channel == 1
6084 deviceFormats = WAVE_FORMAT_1M08 | WAVE_FORMAT_2M08 | WAVE_FORMAT_4M08 | WAVE_FORMAT_96M08;
6085 if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
6086 waveFormat.wBitsPerSample = 8;
6087 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
6089 else { // assume 16-bit is supported
6090 waveFormat.wBitsPerSample = 16;
6091 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
6094 stream_.userFormat = format;
6096 // Update wave format structure and buffer information.
6097 waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
6098 waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
6099 dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
6101 // If the user wants an even bigger buffer, increase the device buffer size accordingly.
6102 while ( dsPointerLeadTime * 2U > dsBufferSize )
6105 // Setup the secondary DS buffer description.
6106 DSCBUFFERDESC bufferDescription;
6107 ZeroMemory( &bufferDescription, sizeof( DSCBUFFERDESC ) );
6108 bufferDescription.dwSize = sizeof( DSCBUFFERDESC );
6109 bufferDescription.dwFlags = 0;
6110 bufferDescription.dwReserved = 0;
6111 bufferDescription.dwBufferBytes = dsBufferSize;
6112 bufferDescription.lpwfxFormat = &waveFormat;
6114 // Create the capture buffer.
6115 LPDIRECTSOUNDCAPTUREBUFFER buffer;
6116 result = input->CreateCaptureBuffer( &bufferDescription, &buffer, NULL );
6117 if ( FAILED( result ) ) {
6119 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating input buffer (" << dsDevices[ device ].name << ")!";
6120 errorText_ = errorStream_.str();
6124 // Get the buffer size ... might be different from what we specified.
6126 dscbcaps.dwSize = sizeof( DSCBCAPS );
6127 result = buffer->GetCaps( &dscbcaps );
6128 if ( FAILED( result ) ) {
6131 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";
6132 errorText_ = errorStream_.str();
6136 dsBufferSize = dscbcaps.dwBufferBytes;
6138 // NOTE: We could have a problem here if this is a duplex stream
6139 // and the play and capture hardware buffer sizes are different
6140 // (I'm actually not sure if that is a problem or not).
6141 // Currently, we are not verifying that.
6143 // Lock the capture buffer
6146 result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );
6147 if ( FAILED( result ) ) {
6150 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking input buffer (" << dsDevices[ device ].name << ")!";
6151 errorText_ = errorStream_.str();
6156 ZeroMemory( audioPtr, dataLen );
6158 // Unlock the buffer
6159 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
6160 if ( FAILED( result ) ) {
6163 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking input buffer (" << dsDevices[ device ].name << ")!";
6164 errorText_ = errorStream_.str();
6168 ohandle = (void *) input;
6169 bhandle = (void *) buffer;
6172 // Set various stream parameters
6173 DsHandle *handle = 0;
6174 stream_.nDeviceChannels[mode] = channels + firstChannel;
6175 stream_.nUserChannels[mode] = channels;
6176 stream_.bufferSize = *bufferSize;
6177 stream_.channelOffset[mode] = firstChannel;
6178 stream_.deviceInterleaved[mode] = true;
6179 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
6180 else stream_.userInterleaved = true;
6182 // Set flag for buffer conversion
6183 stream_.doConvertBuffer[mode] = false;
6184 if (stream_.nUserChannels[mode] != stream_.nDeviceChannels[mode])
6185 stream_.doConvertBuffer[mode] = true;
6186 if (stream_.userFormat != stream_.deviceFormat[mode])
6187 stream_.doConvertBuffer[mode] = true;
6188 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
6189 stream_.nUserChannels[mode] > 1 )
6190 stream_.doConvertBuffer[mode] = true;
6192 // Allocate necessary internal buffers
6193 long bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
6194 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
6195 if ( stream_.userBuffer[mode] == NULL ) {
6196 errorText_ = "RtApiDs::probeDeviceOpen: error allocating user buffer memory.";
6200 if ( stream_.doConvertBuffer[mode] ) {
6202 bool makeBuffer = true;
6203 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
6204 if ( mode == INPUT ) {
6205 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
6206 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
6207 if ( bufferBytes <= (long) bytesOut ) makeBuffer = false;
6212 bufferBytes *= *bufferSize;
6213 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
6214 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
6215 if ( stream_.deviceBuffer == NULL ) {
6216 errorText_ = "RtApiDs::probeDeviceOpen: error allocating device buffer memory.";
6222 // Allocate our DsHandle structures for the stream.
6223 if ( stream_.apiHandle == 0 ) {
6225 handle = new DsHandle;
6227 catch ( std::bad_alloc& ) {
6228 errorText_ = "RtApiDs::probeDeviceOpen: error allocating AsioHandle memory.";
6232 // Create a manual-reset event.
6233 handle->condition = CreateEvent( NULL, // no security
6234 TRUE, // manual-reset
6235 FALSE, // non-signaled initially
6237 stream_.apiHandle = (void *) handle;
6240 handle = (DsHandle *) stream_.apiHandle;
6241 handle->id[mode] = ohandle;
6242 handle->buffer[mode] = bhandle;
6243 handle->dsBufferSize[mode] = dsBufferSize;
6244 handle->dsPointerLeadTime[mode] = dsPointerLeadTime;
6246 stream_.device[mode] = device;
6247 stream_.state = STREAM_STOPPED;
6248 if ( stream_.mode == OUTPUT && mode == INPUT )
6249 // We had already set up an output stream.
6250 stream_.mode = DUPLEX;
6252 stream_.mode = mode;
6253 stream_.nBuffers = nBuffers;
6254 stream_.sampleRate = sampleRate;
6256 // Setup the buffer conversion information structure.
6257 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
6259 // Setup the callback thread.
6260 if ( stream_.callbackInfo.isRunning == false ) {
6262 stream_.callbackInfo.isRunning = true;
6263 stream_.callbackInfo.object = (void *) this;
6264 stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &callbackHandler,
6265 &stream_.callbackInfo, 0, &threadId );
6266 if ( stream_.callbackInfo.thread == 0 ) {
6267 errorText_ = "RtApiDs::probeDeviceOpen: error creating callback thread!";
6271 // Boost DS thread priority
6272 SetThreadPriority( (HANDLE) stream_.callbackInfo.thread, THREAD_PRIORITY_HIGHEST );
6278 if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
6279 LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
6280 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6281 if ( buffer ) buffer->Release();
6284 if ( handle->buffer[1] ) {
6285 LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
6286 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6287 if ( buffer ) buffer->Release();
6290 CloseHandle( handle->condition );
6292 stream_.apiHandle = 0;
6295 for ( int i=0; i<2; i++ ) {
6296 if ( stream_.userBuffer[i] ) {
6297 free( stream_.userBuffer[i] );
6298 stream_.userBuffer[i] = 0;
6302 if ( stream_.deviceBuffer ) {
6303 free( stream_.deviceBuffer );
6304 stream_.deviceBuffer = 0;
6307 stream_.state = STREAM_CLOSED;
6311 void RtApiDs :: closeStream()
6313 if ( stream_.state == STREAM_CLOSED ) {
6314 errorText_ = "RtApiDs::closeStream(): no open stream to close!";
6315 error( RtAudioError::WARNING );
6319 // Stop the callback thread.
6320 stream_.callbackInfo.isRunning = false;
6321 WaitForSingleObject( (HANDLE) stream_.callbackInfo.thread, INFINITE );
6322 CloseHandle( (HANDLE) stream_.callbackInfo.thread );
6324 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6326 if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
6327 LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
6328 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6335 if ( handle->buffer[1] ) {
6336 LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
6337 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6344 CloseHandle( handle->condition );
6346 stream_.apiHandle = 0;
6349 for ( int i=0; i<2; i++ ) {
6350 if ( stream_.userBuffer[i] ) {
6351 free( stream_.userBuffer[i] );
6352 stream_.userBuffer[i] = 0;
6356 if ( stream_.deviceBuffer ) {
6357 free( stream_.deviceBuffer );
6358 stream_.deviceBuffer = 0;
6361 stream_.mode = UNINITIALIZED;
6362 stream_.state = STREAM_CLOSED;
6365 void RtApiDs :: startStream()
6368 if ( stream_.state == STREAM_RUNNING ) {
6369 errorText_ = "RtApiDs::startStream(): the stream is already running!";
6370 error( RtAudioError::WARNING );
6374 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6376 // Increase scheduler frequency on lesser windows (a side-effect of
6377 // increasing timer accuracy). On greater windows (Win2K or later),
6378 // this is already in effect.
6379 timeBeginPeriod( 1 );
6381 buffersRolling = false;
6382 duplexPrerollBytes = 0;
6384 if ( stream_.mode == DUPLEX ) {
6385 // 0.5 seconds of silence in DUPLEX mode while the devices spin up and synchronize.
6386 duplexPrerollBytes = (int) ( 0.5 * stream_.sampleRate * formatBytes( stream_.deviceFormat[1] ) * stream_.nDeviceChannels[1] );
6390 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
6392 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6393 result = buffer->Play( 0, 0, DSBPLAY_LOOPING );
6394 if ( FAILED( result ) ) {
6395 errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting output buffer!";
6396 errorText_ = errorStream_.str();
6401 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
6403 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6404 result = buffer->Start( DSCBSTART_LOOPING );
6405 if ( FAILED( result ) ) {
6406 errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting input buffer!";
6407 errorText_ = errorStream_.str();
6412 handle->drainCounter = 0;
6413 handle->internalDrain = false;
6414 ResetEvent( handle->condition );
6415 stream_.state = STREAM_RUNNING;
6418 if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );
6421 void RtApiDs :: stopStream()
6424 if ( stream_.state == STREAM_STOPPED ) {
6425 errorText_ = "RtApiDs::stopStream(): the stream is already stopped!";
6426 error( RtAudioError::WARNING );
6433 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6434 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
6435 if ( handle->drainCounter == 0 ) {
6436 handle->drainCounter = 2;
6437 WaitForSingleObject( handle->condition, INFINITE ); // block until signaled
6440 stream_.state = STREAM_STOPPED;
6442 MUTEX_LOCK( &stream_.mutex );
6444 // Stop the buffer and clear memory
6445 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6446 result = buffer->Stop();
6447 if ( FAILED( result ) ) {
6448 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping output buffer!";
6449 errorText_ = errorStream_.str();
6453 // Lock the buffer and clear it so that if we start to play again,
6454 // we won't have old data playing.
6455 result = buffer->Lock( 0, handle->dsBufferSize[0], &audioPtr, &dataLen, NULL, NULL, 0 );
6456 if ( FAILED( result ) ) {
6457 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking output buffer!";
6458 errorText_ = errorStream_.str();
6462 // Zero the DS buffer
6463 ZeroMemory( audioPtr, dataLen );
6465 // Unlock the DS buffer
6466 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
6467 if ( FAILED( result ) ) {
6468 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking output buffer!";
6469 errorText_ = errorStream_.str();
6473 // If we start playing again, we must begin at beginning of buffer.
6474 handle->bufferPointer[0] = 0;
6477 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
6478 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6482 stream_.state = STREAM_STOPPED;
6484 if ( stream_.mode != DUPLEX )
6485 MUTEX_LOCK( &stream_.mutex );
6487 result = buffer->Stop();
6488 if ( FAILED( result ) ) {
6489 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping input buffer!";
6490 errorText_ = errorStream_.str();
6494 // Lock the buffer and clear it so that if we start to play again,
6495 // we won't have old data playing.
6496 result = buffer->Lock( 0, handle->dsBufferSize[1], &audioPtr, &dataLen, NULL, NULL, 0 );
6497 if ( FAILED( result ) ) {
6498 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking input buffer!";
6499 errorText_ = errorStream_.str();
6503 // Zero the DS buffer
6504 ZeroMemory( audioPtr, dataLen );
6506 // Unlock the DS buffer
6507 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
6508 if ( FAILED( result ) ) {
6509 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking input buffer!";
6510 errorText_ = errorStream_.str();
6514 // If we start recording again, we must begin at beginning of buffer.
6515 handle->bufferPointer[1] = 0;
6519 timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows.
6520 MUTEX_UNLOCK( &stream_.mutex );
6522 if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );
6525 void RtApiDs :: abortStream()
6528 if ( stream_.state == STREAM_STOPPED ) {
6529 errorText_ = "RtApiDs::abortStream(): the stream is already stopped!";
6530 error( RtAudioError::WARNING );
6534 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6535 handle->drainCounter = 2;
6540 void RtApiDs :: callbackEvent()
6542 if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) {
6543 Sleep( 50 ); // sleep 50 milliseconds
6547 if ( stream_.state == STREAM_CLOSED ) {
6548 errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!";
6549 error( RtAudioError::WARNING );
6553 CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
6554 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6556 // Check if we were draining the stream and signal is finished.
6557 if ( handle->drainCounter > stream_.nBuffers + 2 ) {
6559 stream_.state = STREAM_STOPPING;
6560 if ( handle->internalDrain == false )
6561 SetEvent( handle->condition );
6567 // Invoke user callback to get fresh output data UNLESS we are
6569 if ( handle->drainCounter == 0 ) {
6570 RtAudioCallback callback = (RtAudioCallback) info->callback;
6571 double streamTime = getStreamTime();
6572 RtAudioStreamStatus status = 0;
6573 if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
6574 status |= RTAUDIO_OUTPUT_UNDERFLOW;
6575 handle->xrun[0] = false;
6577 if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
6578 status |= RTAUDIO_INPUT_OVERFLOW;
6579 handle->xrun[1] = false;
6581 int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
6582 stream_.bufferSize, streamTime, status, info->userData );
6583 if ( cbReturnValue == 2 ) {
6584 stream_.state = STREAM_STOPPING;
6585 handle->drainCounter = 2;
6589 else if ( cbReturnValue == 1 ) {
6590 handle->drainCounter = 1;
6591 handle->internalDrain = true;
6596 DWORD currentWritePointer, safeWritePointer;
6597 DWORD currentReadPointer, safeReadPointer;
6598 UINT nextWritePointer;
6600 LPVOID buffer1 = NULL;
6601 LPVOID buffer2 = NULL;
6602 DWORD bufferSize1 = 0;
6603 DWORD bufferSize2 = 0;
6608 MUTEX_LOCK( &stream_.mutex );
6609 if ( stream_.state == STREAM_STOPPED ) {
6610 MUTEX_UNLOCK( &stream_.mutex );
6614 if ( buffersRolling == false ) {
6615 if ( stream_.mode == DUPLEX ) {
6616 //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
6618 // It takes a while for the devices to get rolling. As a result,
6619 // there's no guarantee that the capture and write device pointers
6620 // will move in lockstep. Wait here for both devices to start
6621 // rolling, and then set our buffer pointers accordingly.
6622 // e.g. Crystal Drivers: the capture buffer starts up 5700 to 9600
6623 // bytes later than the write buffer.
6625 // Stub: a serious risk of having a pre-emptive scheduling round
6626 // take place between the two GetCurrentPosition calls... but I'm
6627 // really not sure how to solve the problem. Temporarily boost to
6628 // Realtime priority, maybe; but I'm not sure what priority the
6629 // DirectSound service threads run at. We *should* be roughly
6630 // within a ms or so of correct.
6632 LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6633 LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6635 DWORD startSafeWritePointer, startSafeReadPointer;
6637 result = dsWriteBuffer->GetCurrentPosition( NULL, &startSafeWritePointer );
6638 if ( FAILED( result ) ) {
6639 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
6640 errorText_ = errorStream_.str();
6641 MUTEX_UNLOCK( &stream_.mutex );
6642 error( RtAudioError::SYSTEM_ERROR );
6645 result = dsCaptureBuffer->GetCurrentPosition( NULL, &startSafeReadPointer );
6646 if ( FAILED( result ) ) {
6647 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
6648 errorText_ = errorStream_.str();
6649 MUTEX_UNLOCK( &stream_.mutex );
6650 error( RtAudioError::SYSTEM_ERROR );
6654 result = dsWriteBuffer->GetCurrentPosition( NULL, &safeWritePointer );
6655 if ( FAILED( result ) ) {
6656 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
6657 errorText_ = errorStream_.str();
6658 MUTEX_UNLOCK( &stream_.mutex );
6659 error( RtAudioError::SYSTEM_ERROR );
6662 result = dsCaptureBuffer->GetCurrentPosition( NULL, &safeReadPointer );
6663 if ( FAILED( result ) ) {
6664 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
6665 errorText_ = errorStream_.str();
6666 MUTEX_UNLOCK( &stream_.mutex );
6667 error( RtAudioError::SYSTEM_ERROR );
6670 if ( safeWritePointer != startSafeWritePointer && safeReadPointer != startSafeReadPointer ) break;
6674 //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
6676 handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];
6677 if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];
6678 handle->bufferPointer[1] = safeReadPointer;
6680 else if ( stream_.mode == OUTPUT ) {
6682 // Set the proper nextWritePosition after initial startup.
6683 LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6684 result = dsWriteBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer );
6685 if ( FAILED( result ) ) {
6686 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
6687 errorText_ = errorStream_.str();
6688 MUTEX_UNLOCK( &stream_.mutex );
6689 error( RtAudioError::SYSTEM_ERROR );
6692 handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];
6693 if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];
6696 buffersRolling = true;
6699 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
6701 LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6703 if ( handle->drainCounter > 1 ) { // write zeros to the output stream
6704 bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
6705 bufferBytes *= formatBytes( stream_.userFormat );
6706 memset( stream_.userBuffer[0], 0, bufferBytes );
6709 // Setup parameters and do buffer conversion if necessary.
6710 if ( stream_.doConvertBuffer[0] ) {
6711 buffer = stream_.deviceBuffer;
6712 convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
6713 bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[0];
6714 bufferBytes *= formatBytes( stream_.deviceFormat[0] );
6717 buffer = stream_.userBuffer[0];
6718 bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
6719 bufferBytes *= formatBytes( stream_.userFormat );
6722 // No byte swapping necessary in DirectSound implementation.
6724 // Ahhh ... windoze. 16-bit data is signed but 8-bit data is
6725 // unsigned. So, we need to convert our signed 8-bit data here to
6727 if ( stream_.deviceFormat[0] == RTAUDIO_SINT8 )
6728 for ( int i=0; i<bufferBytes; i++ ) buffer[i] = (unsigned char) ( buffer[i] + 128 );
6730 DWORD dsBufferSize = handle->dsBufferSize[0];
6731 nextWritePointer = handle->bufferPointer[0];
6733 DWORD endWrite, leadPointer;
6735 // Find out where the read and "safe write" pointers are.
6736 result = dsBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer );
6737 if ( FAILED( result ) ) {
6738 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
6739 errorText_ = errorStream_.str();
6740 MUTEX_UNLOCK( &stream_.mutex );
6741 error( RtAudioError::SYSTEM_ERROR );
6745 // We will copy our output buffer into the region between
6746 // safeWritePointer and leadPointer. If leadPointer is not
6747 // beyond the next endWrite position, wait until it is.
6748 leadPointer = safeWritePointer + handle->dsPointerLeadTime[0];
6749 //std::cout << "safeWritePointer = " << safeWritePointer << ", leadPointer = " << leadPointer << ", nextWritePointer = " << nextWritePointer << std::endl;
6750 if ( leadPointer > dsBufferSize ) leadPointer -= dsBufferSize;
6751 if ( leadPointer < nextWritePointer ) leadPointer += dsBufferSize; // unwrap offset
6752 endWrite = nextWritePointer + bufferBytes;
6754 // Check whether the entire write region is behind the play pointer.
6755 if ( leadPointer >= endWrite ) break;
6757 // If we are here, then we must wait until the leadPointer advances
6758 // beyond the end of our next write region. We use the
6759 // Sleep() function to suspend operation until that happens.
6760 double millis = ( endWrite - leadPointer ) * 1000.0;
6761 millis /= ( formatBytes( stream_.deviceFormat[0]) * stream_.nDeviceChannels[0] * stream_.sampleRate);
6762 if ( millis < 1.0 ) millis = 1.0;
6763 Sleep( (DWORD) millis );
6766 if ( dsPointerBetween( nextWritePointer, safeWritePointer, currentWritePointer, dsBufferSize )
6767 || dsPointerBetween( endWrite, safeWritePointer, currentWritePointer, dsBufferSize ) ) {
6768 // We've strayed into the forbidden zone ... resync the read pointer.
6769 handle->xrun[0] = true;
6770 nextWritePointer = safeWritePointer + handle->dsPointerLeadTime[0] - bufferBytes;
6771 if ( nextWritePointer >= dsBufferSize ) nextWritePointer -= dsBufferSize;
6772 handle->bufferPointer[0] = nextWritePointer;
6773 endWrite = nextWritePointer + bufferBytes;
6776 // Lock free space in the buffer
6777 result = dsBuffer->Lock( nextWritePointer, bufferBytes, &buffer1,
6778 &bufferSize1, &buffer2, &bufferSize2, 0 );
6779 if ( FAILED( result ) ) {
6780 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking buffer during playback!";
6781 errorText_ = errorStream_.str();
6782 MUTEX_UNLOCK( &stream_.mutex );
6783 error( RtAudioError::SYSTEM_ERROR );
6787 // Copy our buffer into the DS buffer
6788 CopyMemory( buffer1, buffer, bufferSize1 );
6789 if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 );
6791 // Update our buffer offset and unlock sound buffer
6792 dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
6793 if ( FAILED( result ) ) {
6794 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!";
6795 errorText_ = errorStream_.str();
6796 MUTEX_UNLOCK( &stream_.mutex );
6797 error( RtAudioError::SYSTEM_ERROR );
6800 nextWritePointer = ( nextWritePointer + bufferSize1 + bufferSize2 ) % dsBufferSize;
6801 handle->bufferPointer[0] = nextWritePointer;
6804 // Don't bother draining input
6805 if ( handle->drainCounter ) {
6806 handle->drainCounter++;
6810 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
6812 // Setup parameters.
6813 if ( stream_.doConvertBuffer[1] ) {
6814 buffer = stream_.deviceBuffer;
6815 bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[1];
6816 bufferBytes *= formatBytes( stream_.deviceFormat[1] );
6819 buffer = stream_.userBuffer[1];
6820 bufferBytes = stream_.bufferSize * stream_.nUserChannels[1];
6821 bufferBytes *= formatBytes( stream_.userFormat );
6824 LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6825 long nextReadPointer = handle->bufferPointer[1];
6826 DWORD dsBufferSize = handle->dsBufferSize[1];
6828 // Find out where the write and "safe read" pointers are.
6829 result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer );
6830 if ( FAILED( result ) ) {
6831 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
6832 errorText_ = errorStream_.str();
6833 MUTEX_UNLOCK( &stream_.mutex );
6834 error( RtAudioError::SYSTEM_ERROR );
6838 if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset
6839 DWORD endRead = nextReadPointer + bufferBytes;
6841 // Handling depends on whether we are INPUT or DUPLEX.
6842 // If we're in INPUT mode then waiting is a good thing. If we're in DUPLEX mode,
6843 // then a wait here will drag the write pointers into the forbidden zone.
6845 // In DUPLEX mode, rather than wait, we will back off the read pointer until
6846 // it's in a safe position. This causes dropouts, but it seems to be the only
6847 // practical way to sync up the read and write pointers reliably, given the
6848 // the very complex relationship between phase and increment of the read and write
6851 // In order to minimize audible dropouts in DUPLEX mode, we will
6852 // provide a pre-roll period of 0.5 seconds in which we return
6853 // zeros from the read buffer while the pointers sync up.
6855 if ( stream_.mode == DUPLEX ) {
6856 if ( safeReadPointer < endRead ) {
6857 if ( duplexPrerollBytes <= 0 ) {
6858 // Pre-roll time over. Be more agressive.
6859 int adjustment = endRead-safeReadPointer;
6861 handle->xrun[1] = true;
6863 // - large adjustments: we've probably run out of CPU cycles, so just resync exactly,
6864 // and perform fine adjustments later.
6865 // - small adjustments: back off by twice as much.
6866 if ( adjustment >= 2*bufferBytes )
6867 nextReadPointer = safeReadPointer-2*bufferBytes;
6869 nextReadPointer = safeReadPointer-bufferBytes-adjustment;
6871 if ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;
6875 // In pre=roll time. Just do it.
6876 nextReadPointer = safeReadPointer - bufferBytes;
6877 while ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;
6879 endRead = nextReadPointer + bufferBytes;
6882 else { // mode == INPUT
6883 while ( safeReadPointer < endRead && stream_.callbackInfo.isRunning ) {
6884 // See comments for playback.
6885 double millis = (endRead - safeReadPointer) * 1000.0;
6886 millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate);
6887 if ( millis < 1.0 ) millis = 1.0;
6888 Sleep( (DWORD) millis );
6890 // Wake up and find out where we are now.
6891 result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer );
6892 if ( FAILED( result ) ) {
6893 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
6894 errorText_ = errorStream_.str();
6895 MUTEX_UNLOCK( &stream_.mutex );
6896 error( RtAudioError::SYSTEM_ERROR );
6900 if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset
6904 // Lock free space in the buffer
6905 result = dsBuffer->Lock( nextReadPointer, bufferBytes, &buffer1,
6906 &bufferSize1, &buffer2, &bufferSize2, 0 );
6907 if ( FAILED( result ) ) {
6908 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking capture buffer!";
6909 errorText_ = errorStream_.str();
6910 MUTEX_UNLOCK( &stream_.mutex );
6911 error( RtAudioError::SYSTEM_ERROR );
6915 if ( duplexPrerollBytes <= 0 ) {
6916 // Copy our buffer into the DS buffer
6917 CopyMemory( buffer, buffer1, bufferSize1 );
6918 if ( buffer2 != NULL ) CopyMemory( buffer+bufferSize1, buffer2, bufferSize2 );
6921 memset( buffer, 0, bufferSize1 );
6922 if ( buffer2 != NULL ) memset( buffer + bufferSize1, 0, bufferSize2 );
6923 duplexPrerollBytes -= bufferSize1 + bufferSize2;
6926 // Update our buffer offset and unlock sound buffer
6927 nextReadPointer = ( nextReadPointer + bufferSize1 + bufferSize2 ) % dsBufferSize;
6928 dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
6929 if ( FAILED( result ) ) {
6930 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking capture buffer!";
6931 errorText_ = errorStream_.str();
6932 MUTEX_UNLOCK( &stream_.mutex );
6933 error( RtAudioError::SYSTEM_ERROR );
6936 handle->bufferPointer[1] = nextReadPointer;
6938 // No byte swapping necessary in DirectSound implementation.
6940 // If necessary, convert 8-bit data from unsigned to signed.
6941 if ( stream_.deviceFormat[1] == RTAUDIO_SINT8 )
6942 for ( int j=0; j<bufferBytes; j++ ) buffer[j] = (signed char) ( buffer[j] - 128 );
6944 // Do buffer conversion if necessary.
6945 if ( stream_.doConvertBuffer[1] )
6946 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
6950 MUTEX_UNLOCK( &stream_.mutex );
6951 RtApi::tickStreamTime();
6954 // Definitions for utility functions and callbacks
6955 // specific to the DirectSound implementation.
6957 static unsigned __stdcall callbackHandler( void *ptr )
6959 CallbackInfo *info = (CallbackInfo *) ptr;
6960 RtApiDs *object = (RtApiDs *) info->object;
6961 bool* isRunning = &info->isRunning;
6963 while ( *isRunning == true ) {
6964 object->callbackEvent();
6971 static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
6972 LPCTSTR description,
6976 struct DsProbeData& probeInfo = *(struct DsProbeData*) lpContext;
6977 std::vector<struct DsDevice>& dsDevices = *probeInfo.dsDevices;
6980 bool validDevice = false;
6981 if ( probeInfo.isInput == true ) {
6983 LPDIRECTSOUNDCAPTURE object;
6985 hr = DirectSoundCaptureCreate( lpguid, &object, NULL );
6986 if ( hr != DS_OK ) return TRUE;
6988 caps.dwSize = sizeof(caps);
6989 hr = object->GetCaps( &caps );
6990 if ( hr == DS_OK ) {
6991 if ( caps.dwChannels > 0 && caps.dwFormats > 0 )
6998 LPDIRECTSOUND object;
6999 hr = DirectSoundCreate( lpguid, &object, NULL );
7000 if ( hr != DS_OK ) return TRUE;
7002 caps.dwSize = sizeof(caps);
7003 hr = object->GetCaps( &caps );
7004 if ( hr == DS_OK ) {
7005 if ( caps.dwFlags & DSCAPS_PRIMARYMONO || caps.dwFlags & DSCAPS_PRIMARYSTEREO )
7011 // If good device, then save its name and guid.
7012 std::string name = convertCharPointerToStdString( description );
7013 //if ( name == "Primary Sound Driver" || name == "Primary Sound Capture Driver" )
7014 if ( lpguid == NULL )
7015 name = "Default Device";
7016 if ( validDevice ) {
7017 for ( unsigned int i=0; i<dsDevices.size(); i++ ) {
7018 if ( dsDevices[i].name == name ) {
7019 dsDevices[i].found = true;
7020 if ( probeInfo.isInput ) {
7021 dsDevices[i].id[1] = lpguid;
7022 dsDevices[i].validId[1] = true;
7025 dsDevices[i].id[0] = lpguid;
7026 dsDevices[i].validId[0] = true;
7034 device.found = true;
7035 if ( probeInfo.isInput ) {
7036 device.id[1] = lpguid;
7037 device.validId[1] = true;
7040 device.id[0] = lpguid;
7041 device.validId[0] = true;
7043 dsDevices.push_back( device );
7049 static const char* getErrorString( int code )
7053 case DSERR_ALLOCATED:
7054 return "Already allocated";
7056 case DSERR_CONTROLUNAVAIL:
7057 return "Control unavailable";
7059 case DSERR_INVALIDPARAM:
7060 return "Invalid parameter";
7062 case DSERR_INVALIDCALL:
7063 return "Invalid call";
7066 return "Generic error";
7068 case DSERR_PRIOLEVELNEEDED:
7069 return "Priority level needed";
7071 case DSERR_OUTOFMEMORY:
7072 return "Out of memory";
7074 case DSERR_BADFORMAT:
7075 return "The sample rate or the channel format is not supported";
7077 case DSERR_UNSUPPORTED:
7078 return "Not supported";
7080 case DSERR_NODRIVER:
7083 case DSERR_ALREADYINITIALIZED:
7084 return "Already initialized";
7086 case DSERR_NOAGGREGATION:
7087 return "No aggregation";
7089 case DSERR_BUFFERLOST:
7090 return "Buffer lost";
7092 case DSERR_OTHERAPPHASPRIO:
7093 return "Another application already has priority";
7095 case DSERR_UNINITIALIZED:
7096 return "Uninitialized";
7099 return "DirectSound unknown error";
7102 //******************** End of __WINDOWS_DS__ *********************//
7106 #if defined(__LINUX_ALSA__)
7108 #include <alsa/asoundlib.h>
7111 // A structure to hold various information related to the ALSA API
7114 snd_pcm_t *handles[2];
7117 pthread_cond_t runnable_cv;
7121 :synchronized(false), runnable(false) { xrun[0] = false; xrun[1] = false; }
7124 static void *alsaCallbackHandler( void * ptr );
7126 RtApiAlsa :: RtApiAlsa()
7128 // Nothing to do here.
7131 RtApiAlsa :: ~RtApiAlsa()
7133 if ( stream_.state != STREAM_CLOSED ) closeStream();
7136 unsigned int RtApiAlsa :: getDeviceCount( void )
7138 unsigned nDevices = 0;
7139 int result, subdevice, card;
7143 // Count cards and devices
7145 snd_card_next( &card );
7146 while ( card >= 0 ) {
7147 sprintf( name, "hw:%d", card );
7148 result = snd_ctl_open( &handle, name, 0 );
7150 errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << ".";
7151 errorText_ = errorStream_.str();
7152 error( RtAudioError::WARNING );
7157 result = snd_ctl_pcm_next_device( handle, &subdevice );
7159 errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
7160 errorText_ = errorStream_.str();
7161 error( RtAudioError::WARNING );
7164 if ( subdevice < 0 )
7169 snd_ctl_close( handle );
7170 snd_card_next( &card );
7173 result = snd_ctl_open( &handle, "default", 0 );
7176 snd_ctl_close( handle );
7182 RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device )
7184 RtAudio::DeviceInfo info;
7185 info.probed = false;
7187 unsigned nDevices = 0;
7188 int result, subdevice, card;
7192 // Count cards and devices
7195 snd_card_next( &card );
7196 while ( card >= 0 ) {
7197 sprintf( name, "hw:%d", card );
7198 result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
7200 errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << ".";
7201 errorText_ = errorStream_.str();
7202 error( RtAudioError::WARNING );
7207 result = snd_ctl_pcm_next_device( chandle, &subdevice );
7209 errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
7210 errorText_ = errorStream_.str();
7211 error( RtAudioError::WARNING );
7214 if ( subdevice < 0 ) break;
7215 if ( nDevices == device ) {
7216 sprintf( name, "hw:%d,%d", card, subdevice );
7222 snd_ctl_close( chandle );
7223 snd_card_next( &card );
7226 result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
7227 if ( result == 0 ) {
7228 if ( nDevices == device ) {
7229 strcpy( name, "default" );
7235 if ( nDevices == 0 ) {
7236 errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!";
7237 error( RtAudioError::INVALID_USE );
7241 if ( device >= nDevices ) {
7242 errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!";
7243 error( RtAudioError::INVALID_USE );
7249 // If a stream is already open, we cannot probe the stream devices.
7250 // Thus, use the saved results.
7251 if ( stream_.state != STREAM_CLOSED &&
7252 ( stream_.device[0] == device || stream_.device[1] == device ) ) {
7253 snd_ctl_close( chandle );
7254 if ( device >= devices_.size() ) {
7255 errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened.";
7256 error( RtAudioError::WARNING );
7259 return devices_[ device ];
7262 int openMode = SND_PCM_ASYNC;
7263 snd_pcm_stream_t stream;
7264 snd_pcm_info_t *pcminfo;
7265 snd_pcm_info_alloca( &pcminfo );
7267 snd_pcm_hw_params_t *params;
7268 snd_pcm_hw_params_alloca( ¶ms );
7270 // First try for playback unless default device (which has subdev -1)
7271 stream = SND_PCM_STREAM_PLAYBACK;
7272 snd_pcm_info_set_stream( pcminfo, stream );
7273 if ( subdevice != -1 ) {
7274 snd_pcm_info_set_device( pcminfo, subdevice );
7275 snd_pcm_info_set_subdevice( pcminfo, 0 );
7277 result = snd_ctl_pcm_info( chandle, pcminfo );
7279 // Device probably doesn't support playback.
7284 result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK );
7286 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
7287 errorText_ = errorStream_.str();
7288 error( RtAudioError::WARNING );
7292 // The device is open ... fill the parameter structure.
7293 result = snd_pcm_hw_params_any( phandle, params );
7295 snd_pcm_close( phandle );
7296 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
7297 errorText_ = errorStream_.str();
7298 error( RtAudioError::WARNING );
7302 // Get output channel information.
7304 result = snd_pcm_hw_params_get_channels_max( params, &value );
7306 snd_pcm_close( phandle );
7307 errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << ".";
7308 errorText_ = errorStream_.str();
7309 error( RtAudioError::WARNING );
7312 info.outputChannels = value;
7313 snd_pcm_close( phandle );
7316 stream = SND_PCM_STREAM_CAPTURE;
7317 snd_pcm_info_set_stream( pcminfo, stream );
7319 // Now try for capture unless default device (with subdev = -1)
7320 if ( subdevice != -1 ) {
7321 result = snd_ctl_pcm_info( chandle, pcminfo );
7322 snd_ctl_close( chandle );
7324 // Device probably doesn't support capture.
7325 if ( info.outputChannels == 0 ) return info;
7326 goto probeParameters;
7330 snd_ctl_close( chandle );
7332 result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
7334 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
7335 errorText_ = errorStream_.str();
7336 error( RtAudioError::WARNING );
7337 if ( info.outputChannels == 0 ) return info;
7338 goto probeParameters;
7341 // The device is open ... fill the parameter structure.
7342 result = snd_pcm_hw_params_any( phandle, params );
7344 snd_pcm_close( phandle );
7345 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
7346 errorText_ = errorStream_.str();
7347 error( RtAudioError::WARNING );
7348 if ( info.outputChannels == 0 ) return info;
7349 goto probeParameters;
7352 result = snd_pcm_hw_params_get_channels_max( params, &value );
7354 snd_pcm_close( phandle );
7355 errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << ".";
7356 errorText_ = errorStream_.str();
7357 error( RtAudioError::WARNING );
7358 if ( info.outputChannels == 0 ) return info;
7359 goto probeParameters;
7361 info.inputChannels = value;
7362 snd_pcm_close( phandle );
7364 // If device opens for both playback and capture, we determine the channels.
7365 if ( info.outputChannels > 0 && info.inputChannels > 0 )
7366 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
7368 // ALSA doesn't provide default devices so we'll use the first available one.
7369 if ( device == 0 && info.outputChannels > 0 )
7370 info.isDefaultOutput = true;
7371 if ( device == 0 && info.inputChannels > 0 )
7372 info.isDefaultInput = true;
7375 // At this point, we just need to figure out the supported data
7376 // formats and sample rates. We'll proceed by opening the device in
7377 // the direction with the maximum number of channels, or playback if
7378 // they are equal. This might limit our sample rate options, but so
7381 if ( info.outputChannels >= info.inputChannels )
7382 stream = SND_PCM_STREAM_PLAYBACK;
7384 stream = SND_PCM_STREAM_CAPTURE;
7385 snd_pcm_info_set_stream( pcminfo, stream );
7387 result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
7389 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
7390 errorText_ = errorStream_.str();
7391 error( RtAudioError::WARNING );
7395 // The device is open ... fill the parameter structure.
7396 result = snd_pcm_hw_params_any( phandle, params );
7398 snd_pcm_close( phandle );
7399 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
7400 errorText_ = errorStream_.str();
7401 error( RtAudioError::WARNING );
7405 // Test our discrete set of sample rate values.
7406 info.sampleRates.clear();
7407 for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {
7408 if ( snd_pcm_hw_params_test_rate( phandle, params, SAMPLE_RATES[i], 0 ) == 0 ) {
7409 info.sampleRates.push_back( SAMPLE_RATES[i] );
7411 if ( !info.preferredSampleRate || ( SAMPLE_RATES[i] <= 48000 && SAMPLE_RATES[i] > info.preferredSampleRate ) )
7412 info.preferredSampleRate = SAMPLE_RATES[i];
7415 if ( info.sampleRates.size() == 0 ) {
7416 snd_pcm_close( phandle );
7417 errorStream_ << "RtApiAlsa::getDeviceInfo: no supported sample rates found for device (" << name << ").";
7418 errorText_ = errorStream_.str();
7419 error( RtAudioError::WARNING );
7423 // Probe the supported data formats ... we don't care about endian-ness just yet
7424 snd_pcm_format_t format;
7425 info.nativeFormats = 0;
7426 format = SND_PCM_FORMAT_S8;
7427 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7428 info.nativeFormats |= RTAUDIO_SINT8;
7429 format = SND_PCM_FORMAT_S16;
7430 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7431 info.nativeFormats |= RTAUDIO_SINT16;
7432 format = SND_PCM_FORMAT_S24;
7433 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7434 info.nativeFormats |= RTAUDIO_SINT24;
7435 format = SND_PCM_FORMAT_S32;
7436 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7437 info.nativeFormats |= RTAUDIO_SINT32;
7438 format = SND_PCM_FORMAT_FLOAT;
7439 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7440 info.nativeFormats |= RTAUDIO_FLOAT32;
7441 format = SND_PCM_FORMAT_FLOAT64;
7442 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7443 info.nativeFormats |= RTAUDIO_FLOAT64;
7445 // Check that we have at least one supported format
7446 if ( info.nativeFormats == 0 ) {
7447 snd_pcm_close( phandle );
7448 errorStream_ << "RtApiAlsa::getDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio.";
7449 errorText_ = errorStream_.str();
7450 error( RtAudioError::WARNING );
7454 // Get the device name
7456 result = snd_card_get_name( card, &cardname );
7457 if ( result >= 0 ) {
7458 sprintf( name, "hw:%s,%d", cardname, subdevice );
7463 // That's all ... close the device and return
7464 snd_pcm_close( phandle );
7469 void RtApiAlsa :: saveDeviceInfo( void )
7473 unsigned int nDevices = getDeviceCount();
7474 devices_.resize( nDevices );
7475 for ( unsigned int i=0; i<nDevices; i++ )
7476 devices_[i] = getDeviceInfo( i );
7479 bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
7480 unsigned int firstChannel, unsigned int sampleRate,
7481 RtAudioFormat format, unsigned int *bufferSize,
7482 RtAudio::StreamOptions *options )
7485 #if defined(__RTAUDIO_DEBUG__)
7487 snd_output_stdio_attach(&out, stderr, 0);
7490 // I'm not using the "plug" interface ... too much inconsistent behavior.
7492 unsigned nDevices = 0;
7493 int result, subdevice, card;
7497 if ( options && options->flags & RTAUDIO_ALSA_USE_DEFAULT )
7498 snprintf(name, sizeof(name), "%s", "default");
7500 // Count cards and devices
7502 snd_card_next( &card );
7503 while ( card >= 0 ) {
7504 sprintf( name, "hw:%d", card );
7505 result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
7507 errorStream_ << "RtApiAlsa::probeDeviceOpen: control open, card = " << card << ", " << snd_strerror( result ) << ".";
7508 errorText_ = errorStream_.str();
7513 result = snd_ctl_pcm_next_device( chandle, &subdevice );
7514 if ( result < 0 ) break;
7515 if ( subdevice < 0 ) break;
7516 if ( nDevices == device ) {
7517 sprintf( name, "hw:%d,%d", card, subdevice );
7518 snd_ctl_close( chandle );
7523 snd_ctl_close( chandle );
7524 snd_card_next( &card );
7527 result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
7528 if ( result == 0 ) {
7529 if ( nDevices == device ) {
7530 strcpy( name, "default" );
7531 snd_ctl_close( chandle );
7536 snd_ctl_close( chandle );
7538 if ( nDevices == 0 ) {
7539 // This should not happen because a check is made before this function is called.
7540 errorText_ = "RtApiAlsa::probeDeviceOpen: no devices found!";
7544 if ( device >= nDevices ) {
7545 // This should not happen because a check is made before this function is called.
7546 errorText_ = "RtApiAlsa::probeDeviceOpen: device ID is invalid!";
7553 // The getDeviceInfo() function will not work for a device that is
7554 // already open. Thus, we'll probe the system before opening a
7555 // stream and save the results for use by getDeviceInfo().
7556 if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) // only do once
7557 this->saveDeviceInfo();
7559 snd_pcm_stream_t stream;
7560 if ( mode == OUTPUT )
7561 stream = SND_PCM_STREAM_PLAYBACK;
7563 stream = SND_PCM_STREAM_CAPTURE;
7566 int openMode = SND_PCM_ASYNC;
7567 result = snd_pcm_open( &phandle, name, stream, openMode );
7569 if ( mode == OUTPUT )
7570 errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for output.";
7572 errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for input.";
7573 errorText_ = errorStream_.str();
7577 // Fill the parameter structure.
7578 snd_pcm_hw_params_t *hw_params;
7579 snd_pcm_hw_params_alloca( &hw_params );
7580 result = snd_pcm_hw_params_any( phandle, hw_params );
7582 snd_pcm_close( phandle );
7583 errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") parameters, " << snd_strerror( result ) << ".";
7584 errorText_ = errorStream_.str();
7588 #if defined(__RTAUDIO_DEBUG__)
7589 fprintf( stderr, "\nRtApiAlsa: dump hardware params just after device open:\n\n" );
7590 snd_pcm_hw_params_dump( hw_params, out );
7593 // Set access ... check user preference.
7594 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) {
7595 stream_.userInterleaved = false;
7596 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
7598 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
7599 stream_.deviceInterleaved[mode] = true;
7602 stream_.deviceInterleaved[mode] = false;
7605 stream_.userInterleaved = true;
7606 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
7608 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
7609 stream_.deviceInterleaved[mode] = false;
7612 stream_.deviceInterleaved[mode] = true;
7616 snd_pcm_close( phandle );
7617 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") access, " << snd_strerror( result ) << ".";
7618 errorText_ = errorStream_.str();
7622 // Determine how to set the device format.
7623 stream_.userFormat = format;
7624 snd_pcm_format_t deviceFormat = SND_PCM_FORMAT_UNKNOWN;
7626 if ( format == RTAUDIO_SINT8 )
7627 deviceFormat = SND_PCM_FORMAT_S8;
7628 else if ( format == RTAUDIO_SINT16 )
7629 deviceFormat = SND_PCM_FORMAT_S16;
7630 else if ( format == RTAUDIO_SINT24 )
7631 deviceFormat = SND_PCM_FORMAT_S24;
7632 else if ( format == RTAUDIO_SINT32 )
7633 deviceFormat = SND_PCM_FORMAT_S32;
7634 else if ( format == RTAUDIO_FLOAT32 )
7635 deviceFormat = SND_PCM_FORMAT_FLOAT;
7636 else if ( format == RTAUDIO_FLOAT64 )
7637 deviceFormat = SND_PCM_FORMAT_FLOAT64;
7639 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat) == 0) {
7640 stream_.deviceFormat[mode] = format;
7644 // The user requested format is not natively supported by the device.
7645 deviceFormat = SND_PCM_FORMAT_FLOAT64;
7646 if ( snd_pcm_hw_params_test_format( phandle, hw_params, deviceFormat ) == 0 ) {
7647 stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
7651 deviceFormat = SND_PCM_FORMAT_FLOAT;
7652 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7653 stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
7657 deviceFormat = SND_PCM_FORMAT_S32;
7658 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7659 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
7663 deviceFormat = SND_PCM_FORMAT_S24;
7664 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7665 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
7669 deviceFormat = SND_PCM_FORMAT_S16;
7670 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7671 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
7675 deviceFormat = SND_PCM_FORMAT_S8;
7676 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7677 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
7681 // If we get here, no supported format was found.
7682 snd_pcm_close( phandle );
7683 errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device " << device << " data format not supported by RtAudio.";
7684 errorText_ = errorStream_.str();
7688 result = snd_pcm_hw_params_set_format( phandle, hw_params, deviceFormat );
7690 snd_pcm_close( phandle );
7691 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") data format, " << snd_strerror( result ) << ".";
7692 errorText_ = errorStream_.str();
7696 // Determine whether byte-swaping is necessary.
7697 stream_.doByteSwap[mode] = false;
7698 if ( deviceFormat != SND_PCM_FORMAT_S8 ) {
7699 result = snd_pcm_format_cpu_endian( deviceFormat );
7701 stream_.doByteSwap[mode] = true;
7702 else if (result < 0) {
7703 snd_pcm_close( phandle );
7704 errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") endian-ness, " << snd_strerror( result ) << ".";
7705 errorText_ = errorStream_.str();
7710 // Set the sample rate.
7711 result = snd_pcm_hw_params_set_rate_near( phandle, hw_params, (unsigned int*) &sampleRate, 0 );
7713 snd_pcm_close( phandle );
7714 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting sample rate on device (" << name << "), " << snd_strerror( result ) << ".";
7715 errorText_ = errorStream_.str();
7719 // Determine the number of channels for this device. We support a possible
7720 // minimum device channel number > than the value requested by the user.
7721 stream_.nUserChannels[mode] = channels;
7723 result = snd_pcm_hw_params_get_channels_max( hw_params, &value );
7724 unsigned int deviceChannels = value;
7725 if ( result < 0 || deviceChannels < channels + firstChannel ) {
7726 snd_pcm_close( phandle );
7727 errorStream_ << "RtApiAlsa::probeDeviceOpen: requested channel parameters not supported by device (" << name << "), " << snd_strerror( result ) << ".";
7728 errorText_ = errorStream_.str();
7732 result = snd_pcm_hw_params_get_channels_min( hw_params, &value );
7734 snd_pcm_close( phandle );
7735 errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting minimum channels for device (" << name << "), " << snd_strerror( result ) << ".";
7736 errorText_ = errorStream_.str();
7739 deviceChannels = value;
7740 if ( deviceChannels < channels + firstChannel ) deviceChannels = channels + firstChannel;
7741 stream_.nDeviceChannels[mode] = deviceChannels;
7743 // Set the device channels.
7744 result = snd_pcm_hw_params_set_channels( phandle, hw_params, deviceChannels );
7746 snd_pcm_close( phandle );
7747 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting channels for device (" << name << "), " << snd_strerror( result ) << ".";
7748 errorText_ = errorStream_.str();
7752 // Set the buffer (or period) size.
7754 snd_pcm_uframes_t periodSize = *bufferSize;
7755 result = snd_pcm_hw_params_set_period_size_near( phandle, hw_params, &periodSize, &dir );
7757 snd_pcm_close( phandle );
7758 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting period size for device (" << name << "), " << snd_strerror( result ) << ".";
7759 errorText_ = errorStream_.str();
7762 *bufferSize = periodSize;
7764 // Set the buffer number, which in ALSA is referred to as the "period".
7765 unsigned int periods = 0;
7766 if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2;
7767 if ( options && options->numberOfBuffers > 0 ) periods = options->numberOfBuffers;
7768 if ( periods < 2 ) periods = 4; // a fairly safe default value
7769 result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir );
7771 snd_pcm_close( phandle );
7772 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << ".";
7773 errorText_ = errorStream_.str();
7777 // If attempting to setup a duplex stream, the bufferSize parameter
7778 // MUST be the same in both directions!
7779 if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {
7780 snd_pcm_close( phandle );
7781 errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ").";
7782 errorText_ = errorStream_.str();
7786 stream_.bufferSize = *bufferSize;
7788 // Install the hardware configuration
7789 result = snd_pcm_hw_params( phandle, hw_params );
7791 snd_pcm_close( phandle );
7792 errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing hardware configuration on device (" << name << "), " << snd_strerror( result ) << ".";
7793 errorText_ = errorStream_.str();
7797 #if defined(__RTAUDIO_DEBUG__)
7798 fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n");
7799 snd_pcm_hw_params_dump( hw_params, out );
7802 // Set the software configuration to fill buffers with zeros and prevent device stopping on xruns.
7803 snd_pcm_sw_params_t *sw_params = NULL;
7804 snd_pcm_sw_params_alloca( &sw_params );
7805 snd_pcm_sw_params_current( phandle, sw_params );
7806 snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize );
7807 snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, ULONG_MAX );
7808 snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 );
7810 // The following two settings were suggested by Theo Veenker
7811 //snd_pcm_sw_params_set_avail_min( phandle, sw_params, *bufferSize );
7812 //snd_pcm_sw_params_set_xfer_align( phandle, sw_params, 1 );
7814 // here are two options for a fix
7815 //snd_pcm_sw_params_set_silence_size( phandle, sw_params, ULONG_MAX );
7816 snd_pcm_uframes_t val;
7817 snd_pcm_sw_params_get_boundary( sw_params, &val );
7818 snd_pcm_sw_params_set_silence_size( phandle, sw_params, val );
7820 result = snd_pcm_sw_params( phandle, sw_params );
7822 snd_pcm_close( phandle );
7823 errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing software configuration on device (" << name << "), " << snd_strerror( result ) << ".";
7824 errorText_ = errorStream_.str();
7828 #if defined(__RTAUDIO_DEBUG__)
7829 fprintf(stderr, "\nRtApiAlsa: dump software params after installation:\n\n");
7830 snd_pcm_sw_params_dump( sw_params, out );
7833 // Set flags for buffer conversion
7834 stream_.doConvertBuffer[mode] = false;
7835 if ( stream_.userFormat != stream_.deviceFormat[mode] )
7836 stream_.doConvertBuffer[mode] = true;
7837 if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
7838 stream_.doConvertBuffer[mode] = true;
7839 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
7840 stream_.nUserChannels[mode] > 1 )
7841 stream_.doConvertBuffer[mode] = true;
7843 // Allocate the ApiHandle if necessary and then save.
7844 AlsaHandle *apiInfo = 0;
7845 if ( stream_.apiHandle == 0 ) {
7847 apiInfo = (AlsaHandle *) new AlsaHandle;
7849 catch ( std::bad_alloc& ) {
7850 errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating AlsaHandle memory.";
7854 if ( pthread_cond_init( &apiInfo->runnable_cv, NULL ) ) {
7855 errorText_ = "RtApiAlsa::probeDeviceOpen: error initializing pthread condition variable.";
7859 stream_.apiHandle = (void *) apiInfo;
7860 apiInfo->handles[0] = 0;
7861 apiInfo->handles[1] = 0;
7864 apiInfo = (AlsaHandle *) stream_.apiHandle;
7866 apiInfo->handles[mode] = phandle;
7869 // Allocate necessary internal buffers.
7870 unsigned long bufferBytes;
7871 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
7872 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
7873 if ( stream_.userBuffer[mode] == NULL ) {
7874 errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating user buffer memory.";
7878 if ( stream_.doConvertBuffer[mode] ) {
7880 bool makeBuffer = true;
7881 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
7882 if ( mode == INPUT ) {
7883 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
7884 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
7885 if ( bufferBytes <= bytesOut ) makeBuffer = false;
7890 bufferBytes *= *bufferSize;
7891 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
7892 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
7893 if ( stream_.deviceBuffer == NULL ) {
7894 errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating device buffer memory.";
7900 stream_.sampleRate = sampleRate;
7901 stream_.nBuffers = periods;
7902 stream_.device[mode] = device;
7903 stream_.state = STREAM_STOPPED;
7905 // Setup the buffer conversion information structure.
7906 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
7908 // Setup thread if necessary.
7909 if ( stream_.mode == OUTPUT && mode == INPUT ) {
7910 // We had already set up an output stream.
7911 stream_.mode = DUPLEX;
7912 // Link the streams if possible.
7913 apiInfo->synchronized = false;
7914 if ( snd_pcm_link( apiInfo->handles[0], apiInfo->handles[1] ) == 0 )
7915 apiInfo->synchronized = true;
7917 errorText_ = "RtApiAlsa::probeDeviceOpen: unable to synchronize input and output devices.";
7918 error( RtAudioError::WARNING );
7922 stream_.mode = mode;
7924 // Setup callback thread.
7925 stream_.callbackInfo.object = (void *) this;
7927 // Set the thread attributes for joinable and realtime scheduling
7928 // priority (optional). The higher priority will only take affect
7929 // if the program is run as root or suid. Note, under Linux
7930 // processes with CAP_SYS_NICE privilege, a user can change
7931 // scheduling policy and priority (thus need not be root). See
7932 // POSIX "capabilities".
7933 pthread_attr_t attr;
7934 pthread_attr_init( &attr );
7935 pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
7936 #ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
7937 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
7938 stream_.callbackInfo.doRealtime = true;
7939 struct sched_param param;
7940 int priority = options->priority;
7941 int min = sched_get_priority_min( SCHED_RR );
7942 int max = sched_get_priority_max( SCHED_RR );
7943 if ( priority < min ) priority = min;
7944 else if ( priority > max ) priority = max;
7945 param.sched_priority = priority;
7947 // Set the policy BEFORE the priority. Otherwise it fails.
7948 pthread_attr_setschedpolicy(&attr, SCHED_RR);
7949 pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM);
7950 // This is definitely required. Otherwise it fails.
7951 pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED);
7952 pthread_attr_setschedparam(&attr, ¶m);
7955 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
7957 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
7960 stream_.callbackInfo.isRunning = true;
7961 result = pthread_create( &stream_.callbackInfo.thread, &attr, alsaCallbackHandler, &stream_.callbackInfo );
7962 pthread_attr_destroy( &attr );
7964 // Failed. Try instead with default attributes.
7965 result = pthread_create( &stream_.callbackInfo.thread, NULL, alsaCallbackHandler, &stream_.callbackInfo );
7967 stream_.callbackInfo.isRunning = false;
7968 errorText_ = "RtApiAlsa::error creating callback thread!";
7978 pthread_cond_destroy( &apiInfo->runnable_cv );
7979 if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
7980 if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
7982 stream_.apiHandle = 0;
7985 if ( phandle) snd_pcm_close( phandle );
7987 for ( int i=0; i<2; i++ ) {
7988 if ( stream_.userBuffer[i] ) {
7989 free( stream_.userBuffer[i] );
7990 stream_.userBuffer[i] = 0;
7994 if ( stream_.deviceBuffer ) {
7995 free( stream_.deviceBuffer );
7996 stream_.deviceBuffer = 0;
7999 stream_.state = STREAM_CLOSED;
8003 void RtApiAlsa :: closeStream()
8005 if ( stream_.state == STREAM_CLOSED ) {
8006 errorText_ = "RtApiAlsa::closeStream(): no open stream to close!";
8007 error( RtAudioError::WARNING );
8011 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
8012 stream_.callbackInfo.isRunning = false;
8013 MUTEX_LOCK( &stream_.mutex );
8014 if ( stream_.state == STREAM_STOPPED ) {
8015 apiInfo->runnable = true;
8016 pthread_cond_signal( &apiInfo->runnable_cv );
8018 MUTEX_UNLOCK( &stream_.mutex );
8019 pthread_join( stream_.callbackInfo.thread, NULL );
8021 if ( stream_.state == STREAM_RUNNING ) {
8022 stream_.state = STREAM_STOPPED;
8023 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
8024 snd_pcm_drop( apiInfo->handles[0] );
8025 if ( stream_.mode == INPUT || stream_.mode == DUPLEX )
8026 snd_pcm_drop( apiInfo->handles[1] );
8030 pthread_cond_destroy( &apiInfo->runnable_cv );
8031 if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
8032 if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
8034 stream_.apiHandle = 0;
8037 for ( int i=0; i<2; i++ ) {
8038 if ( stream_.userBuffer[i] ) {
8039 free( stream_.userBuffer[i] );
8040 stream_.userBuffer[i] = 0;
8044 if ( stream_.deviceBuffer ) {
8045 free( stream_.deviceBuffer );
8046 stream_.deviceBuffer = 0;
8049 stream_.mode = UNINITIALIZED;
8050 stream_.state = STREAM_CLOSED;
8053 void RtApiAlsa :: startStream()
8055 // This method calls snd_pcm_prepare if the device isn't already in that state.
8058 if ( stream_.state == STREAM_RUNNING ) {
8059 errorText_ = "RtApiAlsa::startStream(): the stream is already running!";
8060 error( RtAudioError::WARNING );
8064 MUTEX_LOCK( &stream_.mutex );
8067 snd_pcm_state_t state;
8068 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
8069 snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
8070 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
8071 state = snd_pcm_state( handle[0] );
8072 if ( state != SND_PCM_STATE_PREPARED ) {
8073 result = snd_pcm_prepare( handle[0] );
8075 errorStream_ << "RtApiAlsa::startStream: error preparing output pcm device, " << snd_strerror( result ) << ".";
8076 errorText_ = errorStream_.str();
8082 if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
8083 result = snd_pcm_drop(handle[1]); // fix to remove stale data received since device has been open
8084 state = snd_pcm_state( handle[1] );
8085 if ( state != SND_PCM_STATE_PREPARED ) {
8086 result = snd_pcm_prepare( handle[1] );
8088 errorStream_ << "RtApiAlsa::startStream: error preparing input pcm device, " << snd_strerror( result ) << ".";
8089 errorText_ = errorStream_.str();
8095 stream_.state = STREAM_RUNNING;
8098 apiInfo->runnable = true;
8099 pthread_cond_signal( &apiInfo->runnable_cv );
8100 MUTEX_UNLOCK( &stream_.mutex );
8102 if ( result >= 0 ) return;
8103 error( RtAudioError::SYSTEM_ERROR );
8106 void RtApiAlsa :: stopStream()
8109 if ( stream_.state == STREAM_STOPPED ) {
8110 errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!";
8111 error( RtAudioError::WARNING );
8115 stream_.state = STREAM_STOPPED;
8116 MUTEX_LOCK( &stream_.mutex );
8119 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
8120 snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
8121 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
8122 if ( apiInfo->synchronized )
8123 result = snd_pcm_drop( handle[0] );
8125 result = snd_pcm_drain( handle[0] );
8127 errorStream_ << "RtApiAlsa::stopStream: error draining output pcm device, " << snd_strerror( result ) << ".";
8128 errorText_ = errorStream_.str();
8133 if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
8134 result = snd_pcm_drop( handle[1] );
8136 errorStream_ << "RtApiAlsa::stopStream: error stopping input pcm device, " << snd_strerror( result ) << ".";
8137 errorText_ = errorStream_.str();
8143 apiInfo->runnable = false; // fixes high CPU usage when stopped
8144 MUTEX_UNLOCK( &stream_.mutex );
8146 if ( result >= 0 ) return;
8147 error( RtAudioError::SYSTEM_ERROR );
8150 void RtApiAlsa :: abortStream()
8153 if ( stream_.state == STREAM_STOPPED ) {
8154 errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!";
8155 error( RtAudioError::WARNING );
8159 stream_.state = STREAM_STOPPED;
8160 MUTEX_LOCK( &stream_.mutex );
8163 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
8164 snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
8165 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
8166 result = snd_pcm_drop( handle[0] );
8168 errorStream_ << "RtApiAlsa::abortStream: error aborting output pcm device, " << snd_strerror( result ) << ".";
8169 errorText_ = errorStream_.str();
8174 if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
8175 result = snd_pcm_drop( handle[1] );
8177 errorStream_ << "RtApiAlsa::abortStream: error aborting input pcm device, " << snd_strerror( result ) << ".";
8178 errorText_ = errorStream_.str();
8184 apiInfo->runnable = false; // fixes high CPU usage when stopped
8185 MUTEX_UNLOCK( &stream_.mutex );
8187 if ( result >= 0 ) return;
8188 error( RtAudioError::SYSTEM_ERROR );
8191 void RtApiAlsa :: callbackEvent()
8193 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
8194 if ( stream_.state == STREAM_STOPPED ) {
8195 MUTEX_LOCK( &stream_.mutex );
8196 while ( !apiInfo->runnable )
8197 pthread_cond_wait( &apiInfo->runnable_cv, &stream_.mutex );
8199 if ( stream_.state != STREAM_RUNNING ) {
8200 MUTEX_UNLOCK( &stream_.mutex );
8203 MUTEX_UNLOCK( &stream_.mutex );
8206 if ( stream_.state == STREAM_CLOSED ) {
8207 errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!";
8208 error( RtAudioError::WARNING );
8212 int doStopStream = 0;
8213 RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
8214 double streamTime = getStreamTime();
8215 RtAudioStreamStatus status = 0;
8216 if ( stream_.mode != INPUT && apiInfo->xrun[0] == true ) {
8217 status |= RTAUDIO_OUTPUT_UNDERFLOW;
8218 apiInfo->xrun[0] = false;
8220 if ( stream_.mode != OUTPUT && apiInfo->xrun[1] == true ) {
8221 status |= RTAUDIO_INPUT_OVERFLOW;
8222 apiInfo->xrun[1] = false;
8224 doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
8225 stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
8227 if ( doStopStream == 2 ) {
8232 MUTEX_LOCK( &stream_.mutex );
8234 // The state might change while waiting on a mutex.
8235 if ( stream_.state == STREAM_STOPPED ) goto unlock;
8241 snd_pcm_sframes_t frames;
8242 RtAudioFormat format;
8243 handle = (snd_pcm_t **) apiInfo->handles;
8245 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
8247 // Setup parameters.
8248 if ( stream_.doConvertBuffer[1] ) {
8249 buffer = stream_.deviceBuffer;
8250 channels = stream_.nDeviceChannels[1];
8251 format = stream_.deviceFormat[1];
8254 buffer = stream_.userBuffer[1];
8255 channels = stream_.nUserChannels[1];
8256 format = stream_.userFormat;
8259 // Read samples from device in interleaved/non-interleaved format.
8260 if ( stream_.deviceInterleaved[1] )
8261 result = snd_pcm_readi( handle[1], buffer, stream_.bufferSize );
8263 void *bufs[channels];
8264 size_t offset = stream_.bufferSize * formatBytes( format );
8265 for ( int i=0; i<channels; i++ )
8266 bufs[i] = (void *) (buffer + (i * offset));
8267 result = snd_pcm_readn( handle[1], bufs, stream_.bufferSize );
8270 if ( result < (int) stream_.bufferSize ) {
8271 // Either an error or overrun occured.
8272 if ( result == -EPIPE ) {
8273 snd_pcm_state_t state = snd_pcm_state( handle[1] );
8274 if ( state == SND_PCM_STATE_XRUN ) {
8275 apiInfo->xrun[1] = true;
8276 result = snd_pcm_prepare( handle[1] );
8278 errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after overrun, " << snd_strerror( result ) << ".";
8279 errorText_ = errorStream_.str();
8283 errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
8284 errorText_ = errorStream_.str();
8288 errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << ".";
8289 errorText_ = errorStream_.str();
8291 error( RtAudioError::WARNING );
8295 // Do byte swapping if necessary.
8296 if ( stream_.doByteSwap[1] )
8297 byteSwapBuffer( buffer, stream_.bufferSize * channels, format );
8299 // Do buffer conversion if necessary.
8300 if ( stream_.doConvertBuffer[1] )
8301 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
8303 // Check stream latency
8304 result = snd_pcm_delay( handle[1], &frames );
8305 if ( result == 0 && frames > 0 ) stream_.latency[1] = frames;
8310 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
8312 // Setup parameters and do buffer conversion if necessary.
8313 if ( stream_.doConvertBuffer[0] ) {
8314 buffer = stream_.deviceBuffer;
8315 convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
8316 channels = stream_.nDeviceChannels[0];
8317 format = stream_.deviceFormat[0];
8320 buffer = stream_.userBuffer[0];
8321 channels = stream_.nUserChannels[0];
8322 format = stream_.userFormat;
8325 // Do byte swapping if necessary.
8326 if ( stream_.doByteSwap[0] )
8327 byteSwapBuffer(buffer, stream_.bufferSize * channels, format);
8329 // Write samples to device in interleaved/non-interleaved format.
8330 if ( stream_.deviceInterleaved[0] )
8331 result = snd_pcm_writei( handle[0], buffer, stream_.bufferSize );
8333 void *bufs[channels];
8334 size_t offset = stream_.bufferSize * formatBytes( format );
8335 for ( int i=0; i<channels; i++ )
8336 bufs[i] = (void *) (buffer + (i * offset));
8337 result = snd_pcm_writen( handle[0], bufs, stream_.bufferSize );
8340 if ( result < (int) stream_.bufferSize ) {
8341 // Either an error or underrun occured.
8342 if ( result == -EPIPE ) {
8343 snd_pcm_state_t state = snd_pcm_state( handle[0] );
8344 if ( state == SND_PCM_STATE_XRUN ) {
8345 apiInfo->xrun[0] = true;
8346 result = snd_pcm_prepare( handle[0] );
8348 errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after underrun, " << snd_strerror( result ) << ".";
8349 errorText_ = errorStream_.str();
8352 errorText_ = "RtApiAlsa::callbackEvent: audio write error, underrun.";
8355 errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
8356 errorText_ = errorStream_.str();
8360 errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << ".";
8361 errorText_ = errorStream_.str();
8363 error( RtAudioError::WARNING );
8367 // Check stream latency
8368 result = snd_pcm_delay( handle[0], &frames );
8369 if ( result == 0 && frames > 0 ) stream_.latency[0] = frames;
8373 MUTEX_UNLOCK( &stream_.mutex );
8375 RtApi::tickStreamTime();
8376 if ( doStopStream == 1 ) this->stopStream();
8379 static void *alsaCallbackHandler( void *ptr )
8381 CallbackInfo *info = (CallbackInfo *) ptr;
8382 RtApiAlsa *object = (RtApiAlsa *) info->object;
8383 bool *isRunning = &info->isRunning;
8385 #ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
8386 if ( info->doRealtime ) {
8387 std::cerr << "RtAudio alsa: " <<
8388 (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") <<
8389 "running realtime scheduling" << std::endl;
8393 while ( *isRunning == true ) {
8394 pthread_testcancel();
8395 object->callbackEvent();
8398 pthread_exit( NULL );
8401 //******************** End of __LINUX_ALSA__ *********************//
8404 #if defined(__LINUX_PULSE__)
8406 // Code written by Peter Meerwald, pmeerw@pmeerw.net
8407 // and Tristan Matthews.
8409 #include <pulse/error.h>
8410 #include <pulse/simple.h>
8413 static const unsigned int SUPPORTED_SAMPLERATES[] = { 8000, 16000, 22050, 32000,
8414 44100, 48000, 96000, 0};
8416 struct rtaudio_pa_format_mapping_t {
8417 RtAudioFormat rtaudio_format;
8418 pa_sample_format_t pa_format;
8421 static const rtaudio_pa_format_mapping_t supported_sampleformats[] = {
8422 {RTAUDIO_SINT16, PA_SAMPLE_S16LE},
8423 {RTAUDIO_SINT32, PA_SAMPLE_S32LE},
8424 {RTAUDIO_FLOAT32, PA_SAMPLE_FLOAT32LE},
8425 {0, PA_SAMPLE_INVALID}};
8427 struct PulseAudioHandle {
8431 pthread_cond_t runnable_cv;
8433 PulseAudioHandle() : s_play(0), s_rec(0), runnable(false) { }
8436 RtApiPulse::~RtApiPulse()
8438 if ( stream_.state != STREAM_CLOSED )
8442 unsigned int RtApiPulse::getDeviceCount( void )
8447 RtAudio::DeviceInfo RtApiPulse::getDeviceInfo( unsigned int /*device*/ )
8449 RtAudio::DeviceInfo info;
8451 info.name = "PulseAudio";
8452 info.outputChannels = 2;
8453 info.inputChannels = 2;
8454 info.duplexChannels = 2;
8455 info.isDefaultOutput = true;
8456 info.isDefaultInput = true;
8458 for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr )
8459 info.sampleRates.push_back( *sr );
8461 info.preferredSampleRate = 48000;
8462 info.nativeFormats = RTAUDIO_SINT16 | RTAUDIO_SINT32 | RTAUDIO_FLOAT32;
8467 static void *pulseaudio_callback( void * user )
8469 CallbackInfo *cbi = static_cast<CallbackInfo *>( user );
8470 RtApiPulse *context = static_cast<RtApiPulse *>( cbi->object );
8471 volatile bool *isRunning = &cbi->isRunning;
8473 #ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
8474 if (cbi->doRealtime) {
8475 std::cerr << "RtAudio pulse: " <<
8476 (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") <<
8477 "running realtime scheduling" << std::endl;
8481 while ( *isRunning ) {
8482 pthread_testcancel();
8483 context->callbackEvent();
8486 pthread_exit( NULL );
8489 void RtApiPulse::closeStream( void )
8491 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8493 stream_.callbackInfo.isRunning = false;
8495 MUTEX_LOCK( &stream_.mutex );
8496 if ( stream_.state == STREAM_STOPPED ) {
8497 pah->runnable = true;
8498 pthread_cond_signal( &pah->runnable_cv );
8500 MUTEX_UNLOCK( &stream_.mutex );
8502 pthread_join( pah->thread, 0 );
8503 if ( pah->s_play ) {
8504 pa_simple_flush( pah->s_play, NULL );
8505 pa_simple_free( pah->s_play );
8508 pa_simple_free( pah->s_rec );
8510 pthread_cond_destroy( &pah->runnable_cv );
8512 stream_.apiHandle = 0;
8515 if ( stream_.userBuffer[0] ) {
8516 free( stream_.userBuffer[0] );
8517 stream_.userBuffer[0] = 0;
8519 if ( stream_.userBuffer[1] ) {
8520 free( stream_.userBuffer[1] );
8521 stream_.userBuffer[1] = 0;
8524 stream_.state = STREAM_CLOSED;
8525 stream_.mode = UNINITIALIZED;
8528 void RtApiPulse::callbackEvent( void )
8530 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8532 if ( stream_.state == STREAM_STOPPED ) {
8533 MUTEX_LOCK( &stream_.mutex );
8534 while ( !pah->runnable )
8535 pthread_cond_wait( &pah->runnable_cv, &stream_.mutex );
8537 if ( stream_.state != STREAM_RUNNING ) {
8538 MUTEX_UNLOCK( &stream_.mutex );
8541 MUTEX_UNLOCK( &stream_.mutex );
8544 if ( stream_.state == STREAM_CLOSED ) {
8545 errorText_ = "RtApiPulse::callbackEvent(): the stream is closed ... "
8546 "this shouldn't happen!";
8547 error( RtAudioError::WARNING );
8551 RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
8552 double streamTime = getStreamTime();
8553 RtAudioStreamStatus status = 0;
8554 int doStopStream = callback( stream_.userBuffer[OUTPUT], stream_.userBuffer[INPUT],
8555 stream_.bufferSize, streamTime, status,
8556 stream_.callbackInfo.userData );
8558 if ( doStopStream == 2 ) {
8563 MUTEX_LOCK( &stream_.mutex );
8564 void *pulse_in = stream_.doConvertBuffer[INPUT] ? stream_.deviceBuffer : stream_.userBuffer[INPUT];
8565 void *pulse_out = stream_.doConvertBuffer[OUTPUT] ? stream_.deviceBuffer : stream_.userBuffer[OUTPUT];
8567 if ( stream_.state != STREAM_RUNNING )
8572 if (stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
8573 if ( stream_.doConvertBuffer[OUTPUT] ) {
8574 convertBuffer( stream_.deviceBuffer,
8575 stream_.userBuffer[OUTPUT],
8576 stream_.convertInfo[OUTPUT] );
8577 bytes = stream_.nDeviceChannels[OUTPUT] * stream_.bufferSize *
8578 formatBytes( stream_.deviceFormat[OUTPUT] );
8580 bytes = stream_.nUserChannels[OUTPUT] * stream_.bufferSize *
8581 formatBytes( stream_.userFormat );
8583 if ( pa_simple_write( pah->s_play, pulse_out, bytes, &pa_error ) < 0 ) {
8584 errorStream_ << "RtApiPulse::callbackEvent: audio write error, " <<
8585 pa_strerror( pa_error ) << ".";
8586 errorText_ = errorStream_.str();
8587 error( RtAudioError::WARNING );
8591 if ( stream_.mode == INPUT || stream_.mode == DUPLEX) {
8592 if ( stream_.doConvertBuffer[INPUT] )
8593 bytes = stream_.nDeviceChannels[INPUT] * stream_.bufferSize *
8594 formatBytes( stream_.deviceFormat[INPUT] );
8596 bytes = stream_.nUserChannels[INPUT] * stream_.bufferSize *
8597 formatBytes( stream_.userFormat );
8599 if ( pa_simple_read( pah->s_rec, pulse_in, bytes, &pa_error ) < 0 ) {
8600 errorStream_ << "RtApiPulse::callbackEvent: audio read error, " <<
8601 pa_strerror( pa_error ) << ".";
8602 errorText_ = errorStream_.str();
8603 error( RtAudioError::WARNING );
8605 if ( stream_.doConvertBuffer[INPUT] ) {
8606 convertBuffer( stream_.userBuffer[INPUT],
8607 stream_.deviceBuffer,
8608 stream_.convertInfo[INPUT] );
8613 MUTEX_UNLOCK( &stream_.mutex );
8614 RtApi::tickStreamTime();
8616 if ( doStopStream == 1 )
8620 void RtApiPulse::startStream( void )
8622 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8624 if ( stream_.state == STREAM_CLOSED ) {
8625 errorText_ = "RtApiPulse::startStream(): the stream is not open!";
8626 error( RtAudioError::INVALID_USE );
8629 if ( stream_.state == STREAM_RUNNING ) {
8630 errorText_ = "RtApiPulse::startStream(): the stream is already running!";
8631 error( RtAudioError::WARNING );
8635 MUTEX_LOCK( &stream_.mutex );
8637 stream_.state = STREAM_RUNNING;
8639 pah->runnable = true;
8640 pthread_cond_signal( &pah->runnable_cv );
8641 MUTEX_UNLOCK( &stream_.mutex );
8644 void RtApiPulse::stopStream( void )
8646 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8648 if ( stream_.state == STREAM_CLOSED ) {
8649 errorText_ = "RtApiPulse::stopStream(): the stream is not open!";
8650 error( RtAudioError::INVALID_USE );
8653 if ( stream_.state == STREAM_STOPPED ) {
8654 errorText_ = "RtApiPulse::stopStream(): the stream is already stopped!";
8655 error( RtAudioError::WARNING );
8659 stream_.state = STREAM_STOPPED;
8660 MUTEX_LOCK( &stream_.mutex );
8662 if ( pah && pah->s_play ) {
8664 if ( pa_simple_drain( pah->s_play, &pa_error ) < 0 ) {
8665 errorStream_ << "RtApiPulse::stopStream: error draining output device, " <<
8666 pa_strerror( pa_error ) << ".";
8667 errorText_ = errorStream_.str();
8668 MUTEX_UNLOCK( &stream_.mutex );
8669 error( RtAudioError::SYSTEM_ERROR );
8674 stream_.state = STREAM_STOPPED;
8675 MUTEX_UNLOCK( &stream_.mutex );
8678 void RtApiPulse::abortStream( void )
8680 PulseAudioHandle *pah = static_cast<PulseAudioHandle*>( stream_.apiHandle );
8682 if ( stream_.state == STREAM_CLOSED ) {
8683 errorText_ = "RtApiPulse::abortStream(): the stream is not open!";
8684 error( RtAudioError::INVALID_USE );
8687 if ( stream_.state == STREAM_STOPPED ) {
8688 errorText_ = "RtApiPulse::abortStream(): the stream is already stopped!";
8689 error( RtAudioError::WARNING );
8693 stream_.state = STREAM_STOPPED;
8694 MUTEX_LOCK( &stream_.mutex );
8696 if ( pah && pah->s_play ) {
8698 if ( pa_simple_flush( pah->s_play, &pa_error ) < 0 ) {
8699 errorStream_ << "RtApiPulse::abortStream: error flushing output device, " <<
8700 pa_strerror( pa_error ) << ".";
8701 errorText_ = errorStream_.str();
8702 MUTEX_UNLOCK( &stream_.mutex );
8703 error( RtAudioError::SYSTEM_ERROR );
8708 stream_.state = STREAM_STOPPED;
8709 MUTEX_UNLOCK( &stream_.mutex );
8712 bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode,
8713 unsigned int channels, unsigned int firstChannel,
8714 unsigned int sampleRate, RtAudioFormat format,
8715 unsigned int *bufferSize, RtAudio::StreamOptions *options )
8717 PulseAudioHandle *pah = 0;
8718 unsigned long bufferBytes = 0;
8721 if ( device != 0 ) return false;
8722 if ( mode != INPUT && mode != OUTPUT ) return false;
8723 if ( channels != 1 && channels != 2 ) {
8724 errorText_ = "RtApiPulse::probeDeviceOpen: unsupported number of channels.";
8727 ss.channels = channels;
8729 if ( firstChannel != 0 ) return false;
8731 bool sr_found = false;
8732 for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr ) {
8733 if ( sampleRate == *sr ) {
8735 stream_.sampleRate = sampleRate;
8736 ss.rate = sampleRate;
8741 errorText_ = "RtApiPulse::probeDeviceOpen: unsupported sample rate.";
8746 for ( const rtaudio_pa_format_mapping_t *sf = supported_sampleformats;
8747 sf->rtaudio_format && sf->pa_format != PA_SAMPLE_INVALID; ++sf ) {
8748 if ( format == sf->rtaudio_format ) {
8750 stream_.userFormat = sf->rtaudio_format;
8751 stream_.deviceFormat[mode] = stream_.userFormat;
8752 ss.format = sf->pa_format;
8756 if ( !sf_found ) { // Use internal data format conversion.
8757 stream_.userFormat = format;
8758 stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
8759 ss.format = PA_SAMPLE_FLOAT32LE;
8762 // Set other stream parameters.
8763 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
8764 else stream_.userInterleaved = true;
8765 stream_.deviceInterleaved[mode] = true;
8766 stream_.nBuffers = 1;
8767 stream_.doByteSwap[mode] = false;
8768 stream_.nUserChannels[mode] = channels;
8769 stream_.nDeviceChannels[mode] = channels + firstChannel;
8770 stream_.channelOffset[mode] = 0;
8771 std::string streamName = "RtAudio";
8773 // Set flags for buffer conversion.
8774 stream_.doConvertBuffer[mode] = false;
8775 if ( stream_.userFormat != stream_.deviceFormat[mode] )
8776 stream_.doConvertBuffer[mode] = true;
8777 if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
8778 stream_.doConvertBuffer[mode] = true;
8780 // Allocate necessary internal buffers.
8781 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
8782 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
8783 if ( stream_.userBuffer[mode] == NULL ) {
8784 errorText_ = "RtApiPulse::probeDeviceOpen: error allocating user buffer memory.";
8787 stream_.bufferSize = *bufferSize;
8789 if ( stream_.doConvertBuffer[mode] ) {
8791 bool makeBuffer = true;
8792 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
8793 if ( mode == INPUT ) {
8794 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
8795 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
8796 if ( bufferBytes <= bytesOut ) makeBuffer = false;
8801 bufferBytes *= *bufferSize;
8802 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
8803 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
8804 if ( stream_.deviceBuffer == NULL ) {
8805 errorText_ = "RtApiPulse::probeDeviceOpen: error allocating device buffer memory.";
8811 stream_.device[mode] = device;
8813 // Setup the buffer conversion information structure.
8814 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
8816 if ( !stream_.apiHandle ) {
8817 PulseAudioHandle *pah = new PulseAudioHandle;
8819 errorText_ = "RtApiPulse::probeDeviceOpen: error allocating memory for handle.";
8823 stream_.apiHandle = pah;
8824 if ( pthread_cond_init( &pah->runnable_cv, NULL ) != 0 ) {
8825 errorText_ = "RtApiPulse::probeDeviceOpen: error creating condition variable.";
8829 pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8832 if ( options && !options->streamName.empty() ) streamName = options->streamName;
8835 pa_buffer_attr buffer_attr;
8836 buffer_attr.fragsize = bufferBytes;
8837 buffer_attr.maxlength = -1;
8839 pah->s_rec = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_RECORD, NULL, "Record", &ss, NULL, &buffer_attr, &error );
8840 if ( !pah->s_rec ) {
8841 errorText_ = "RtApiPulse::probeDeviceOpen: error connecting input to PulseAudio server.";
8846 pah->s_play = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_PLAYBACK, NULL, "Playback", &ss, NULL, NULL, &error );
8847 if ( !pah->s_play ) {
8848 errorText_ = "RtApiPulse::probeDeviceOpen: error connecting output to PulseAudio server.";
8856 if ( stream_.mode == UNINITIALIZED )
8857 stream_.mode = mode;
8858 else if ( stream_.mode == mode )
8861 stream_.mode = DUPLEX;
8863 if ( !stream_.callbackInfo.isRunning ) {
8864 stream_.callbackInfo.object = this;
8866 stream_.state = STREAM_STOPPED;
8867 // Set the thread attributes for joinable and realtime scheduling
8868 // priority (optional). The higher priority will only take affect
8869 // if the program is run as root or suid. Note, under Linux
8870 // processes with CAP_SYS_NICE privilege, a user can change
8871 // scheduling policy and priority (thus need not be root). See
8872 // POSIX "capabilities".
8873 pthread_attr_t attr;
8874 pthread_attr_init( &attr );
8875 pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
8876 #ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
8877 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
8878 stream_.callbackInfo.doRealtime = true;
8879 struct sched_param param;
8880 int priority = options->priority;
8881 int min = sched_get_priority_min( SCHED_RR );
8882 int max = sched_get_priority_max( SCHED_RR );
8883 if ( priority < min ) priority = min;
8884 else if ( priority > max ) priority = max;
8885 param.sched_priority = priority;
8887 // Set the policy BEFORE the priority. Otherwise it fails.
8888 pthread_attr_setschedpolicy(&attr, SCHED_RR);
8889 pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM);
8890 // This is definitely required. Otherwise it fails.
8891 pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED);
8892 pthread_attr_setschedparam(&attr, ¶m);
8895 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
8897 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
8900 stream_.callbackInfo.isRunning = true;
8901 int result = pthread_create( &pah->thread, &attr, pulseaudio_callback, (void *)&stream_.callbackInfo);
8902 pthread_attr_destroy(&attr);
8904 // Failed. Try instead with default attributes.
8905 result = pthread_create( &pah->thread, NULL, pulseaudio_callback, (void *)&stream_.callbackInfo);
8907 stream_.callbackInfo.isRunning = false;
8908 errorText_ = "RtApiPulse::probeDeviceOpen: error creating thread.";
8917 if ( pah && stream_.callbackInfo.isRunning ) {
8918 pthread_cond_destroy( &pah->runnable_cv );
8920 stream_.apiHandle = 0;
8923 for ( int i=0; i<2; i++ ) {
8924 if ( stream_.userBuffer[i] ) {
8925 free( stream_.userBuffer[i] );
8926 stream_.userBuffer[i] = 0;
8930 if ( stream_.deviceBuffer ) {
8931 free( stream_.deviceBuffer );
8932 stream_.deviceBuffer = 0;
8935 stream_.state = STREAM_CLOSED;
8939 //******************** End of __LINUX_PULSE__ *********************//
8942 #if defined(__LINUX_OSS__)
8945 #include <sys/ioctl.h>
8948 #include <sys/soundcard.h>
8952 static void *ossCallbackHandler(void * ptr);
8954 // A structure to hold various information related to the OSS API
8957 int id[2]; // device ids
8960 pthread_cond_t runnable;
8963 :triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
8966 RtApiOss :: RtApiOss()
8968 // Nothing to do here.
8971 RtApiOss :: ~RtApiOss()
8973 if ( stream_.state != STREAM_CLOSED ) closeStream();
8976 unsigned int RtApiOss :: getDeviceCount( void )
8978 int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
8979 if ( mixerfd == -1 ) {
8980 errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'.";
8981 error( RtAudioError::WARNING );
8985 oss_sysinfo sysinfo;
8986 if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) {
8988 errorText_ = "RtApiOss::getDeviceCount: error getting sysinfo, OSS version >= 4.0 is required.";
8989 error( RtAudioError::WARNING );
8994 return sysinfo.numaudios;
8997 RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device )
8999 RtAudio::DeviceInfo info;
9000 info.probed = false;
9002 int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
9003 if ( mixerfd == -1 ) {
9004 errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'.";
9005 error( RtAudioError::WARNING );
9009 oss_sysinfo sysinfo;
9010 int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
9011 if ( result == -1 ) {
9013 errorText_ = "RtApiOss::getDeviceInfo: error getting sysinfo, OSS version >= 4.0 is required.";
9014 error( RtAudioError::WARNING );
9018 unsigned nDevices = sysinfo.numaudios;
9019 if ( nDevices == 0 ) {
9021 errorText_ = "RtApiOss::getDeviceInfo: no devices found!";
9022 error( RtAudioError::INVALID_USE );
9026 if ( device >= nDevices ) {
9028 errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!";
9029 error( RtAudioError::INVALID_USE );
9033 oss_audioinfo ainfo;
9035 result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
9037 if ( result == -1 ) {
9038 errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
9039 errorText_ = errorStream_.str();
9040 error( RtAudioError::WARNING );
9045 if ( ainfo.caps & PCM_CAP_OUTPUT ) info.outputChannels = ainfo.max_channels;
9046 if ( ainfo.caps & PCM_CAP_INPUT ) info.inputChannels = ainfo.max_channels;
9047 if ( ainfo.caps & PCM_CAP_DUPLEX ) {
9048 if ( info.outputChannels > 0 && info.inputChannels > 0 && ainfo.caps & PCM_CAP_DUPLEX )
9049 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
9052 // Probe data formats ... do for input
9053 unsigned long mask = ainfo.iformats;
9054 if ( mask & AFMT_S16_LE || mask & AFMT_S16_BE )
9055 info.nativeFormats |= RTAUDIO_SINT16;
9056 if ( mask & AFMT_S8 )
9057 info.nativeFormats |= RTAUDIO_SINT8;
9058 if ( mask & AFMT_S32_LE || mask & AFMT_S32_BE )
9059 info.nativeFormats |= RTAUDIO_SINT32;
9061 if ( mask & AFMT_FLOAT )
9062 info.nativeFormats |= RTAUDIO_FLOAT32;
9064 if ( mask & AFMT_S24_LE || mask & AFMT_S24_BE )
9065 info.nativeFormats |= RTAUDIO_SINT24;
9067 // Check that we have at least one supported format
9068 if ( info.nativeFormats == 0 ) {
9069 errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio.";
9070 errorText_ = errorStream_.str();
9071 error( RtAudioError::WARNING );
9075 // Probe the supported sample rates.
9076 info.sampleRates.clear();
9077 if ( ainfo.nrates ) {
9078 for ( unsigned int i=0; i<ainfo.nrates; i++ ) {
9079 for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
9080 if ( ainfo.rates[i] == SAMPLE_RATES[k] ) {
9081 info.sampleRates.push_back( SAMPLE_RATES[k] );
9083 if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
9084 info.preferredSampleRate = SAMPLE_RATES[k];
9092 // Check min and max rate values;
9093 for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
9094 if ( ainfo.min_rate <= (int) SAMPLE_RATES[k] && ainfo.max_rate >= (int) SAMPLE_RATES[k] ) {
9095 info.sampleRates.push_back( SAMPLE_RATES[k] );
9097 if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
9098 info.preferredSampleRate = SAMPLE_RATES[k];
9103 if ( info.sampleRates.size() == 0 ) {
9104 errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ").";
9105 errorText_ = errorStream_.str();
9106 error( RtAudioError::WARNING );
9110 info.name = ainfo.name;
9117 bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
9118 unsigned int firstChannel, unsigned int sampleRate,
9119 RtAudioFormat format, unsigned int *bufferSize,
9120 RtAudio::StreamOptions *options )
9122 int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
9123 if ( mixerfd == -1 ) {
9124 errorText_ = "RtApiOss::probeDeviceOpen: error opening '/dev/mixer'.";
9128 oss_sysinfo sysinfo;
9129 int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
9130 if ( result == -1 ) {
9132 errorText_ = "RtApiOss::probeDeviceOpen: error getting sysinfo, OSS version >= 4.0 is required.";
9136 unsigned nDevices = sysinfo.numaudios;
9137 if ( nDevices == 0 ) {
9138 // This should not happen because a check is made before this function is called.
9140 errorText_ = "RtApiOss::probeDeviceOpen: no devices found!";
9144 if ( device >= nDevices ) {
9145 // This should not happen because a check is made before this function is called.
9147 errorText_ = "RtApiOss::probeDeviceOpen: device ID is invalid!";
9151 oss_audioinfo ainfo;
9153 result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
9155 if ( result == -1 ) {
9156 errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
9157 errorText_ = errorStream_.str();
9161 // Check if device supports input or output
9162 if ( ( mode == OUTPUT && !( ainfo.caps & PCM_CAP_OUTPUT ) ) ||
9163 ( mode == INPUT && !( ainfo.caps & PCM_CAP_INPUT ) ) ) {
9164 if ( mode == OUTPUT )
9165 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support output.";
9167 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support input.";
9168 errorText_ = errorStream_.str();
9173 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9174 if ( mode == OUTPUT )
9176 else { // mode == INPUT
9177 if (stream_.mode == OUTPUT && stream_.device[0] == device) {
9178 // We just set the same device for playback ... close and reopen for duplex (OSS only).
9179 close( handle->id[0] );
9181 if ( !( ainfo.caps & PCM_CAP_DUPLEX ) ) {
9182 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support duplex mode.";
9183 errorText_ = errorStream_.str();
9186 // Check that the number previously set channels is the same.
9187 if ( stream_.nUserChannels[0] != channels ) {
9188 errorStream_ << "RtApiOss::probeDeviceOpen: input/output channels must be equal for OSS duplex device (" << ainfo.name << ").";
9189 errorText_ = errorStream_.str();
9198 // Set exclusive access if specified.
9199 if ( options && options->flags & RTAUDIO_HOG_DEVICE ) flags |= O_EXCL;
9201 // Try to open the device.
9203 fd = open( ainfo.devnode, flags, 0 );
9205 if ( errno == EBUSY )
9206 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") is busy.";
9208 errorStream_ << "RtApiOss::probeDeviceOpen: error opening device (" << ainfo.name << ").";
9209 errorText_ = errorStream_.str();
9213 // For duplex operation, specifically set this mode (this doesn't seem to work).
9215 if ( flags | O_RDWR ) {
9216 result = ioctl( fd, SNDCTL_DSP_SETDUPLEX, NULL );
9217 if ( result == -1) {
9218 errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ").";
9219 errorText_ = errorStream_.str();
9225 // Check the device channel support.
9226 stream_.nUserChannels[mode] = channels;
9227 if ( ainfo.max_channels < (int)(channels + firstChannel) ) {
9229 errorStream_ << "RtApiOss::probeDeviceOpen: the device (" << ainfo.name << ") does not support requested channel parameters.";
9230 errorText_ = errorStream_.str();
9234 // Set the number of channels.
9235 int deviceChannels = channels + firstChannel;
9236 result = ioctl( fd, SNDCTL_DSP_CHANNELS, &deviceChannels );
9237 if ( result == -1 || deviceChannels < (int)(channels + firstChannel) ) {
9239 errorStream_ << "RtApiOss::probeDeviceOpen: error setting channel parameters on device (" << ainfo.name << ").";
9240 errorText_ = errorStream_.str();
9243 stream_.nDeviceChannels[mode] = deviceChannels;
9245 // Get the data format mask
9247 result = ioctl( fd, SNDCTL_DSP_GETFMTS, &mask );
9248 if ( result == -1 ) {
9250 errorStream_ << "RtApiOss::probeDeviceOpen: error getting device (" << ainfo.name << ") data formats.";
9251 errorText_ = errorStream_.str();
9255 // Determine how to set the device format.
9256 stream_.userFormat = format;
9257 int deviceFormat = -1;
9258 stream_.doByteSwap[mode] = false;
9259 if ( format == RTAUDIO_SINT8 ) {
9260 if ( mask & AFMT_S8 ) {
9261 deviceFormat = AFMT_S8;
9262 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
9265 else if ( format == RTAUDIO_SINT16 ) {
9266 if ( mask & AFMT_S16_NE ) {
9267 deviceFormat = AFMT_S16_NE;
9268 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
9270 else if ( mask & AFMT_S16_OE ) {
9271 deviceFormat = AFMT_S16_OE;
9272 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
9273 stream_.doByteSwap[mode] = true;
9276 else if ( format == RTAUDIO_SINT24 ) {
9277 if ( mask & AFMT_S24_NE ) {
9278 deviceFormat = AFMT_S24_NE;
9279 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
9281 else if ( mask & AFMT_S24_OE ) {
9282 deviceFormat = AFMT_S24_OE;
9283 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
9284 stream_.doByteSwap[mode] = true;
9287 else if ( format == RTAUDIO_SINT32 ) {
9288 if ( mask & AFMT_S32_NE ) {
9289 deviceFormat = AFMT_S32_NE;
9290 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
9292 else if ( mask & AFMT_S32_OE ) {
9293 deviceFormat = AFMT_S32_OE;
9294 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
9295 stream_.doByteSwap[mode] = true;
9299 if ( deviceFormat == -1 ) {
9300 // The user requested format is not natively supported by the device.
9301 if ( mask & AFMT_S16_NE ) {
9302 deviceFormat = AFMT_S16_NE;
9303 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
9305 else if ( mask & AFMT_S32_NE ) {
9306 deviceFormat = AFMT_S32_NE;
9307 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
9309 else if ( mask & AFMT_S24_NE ) {
9310 deviceFormat = AFMT_S24_NE;
9311 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
9313 else if ( mask & AFMT_S16_OE ) {
9314 deviceFormat = AFMT_S16_OE;
9315 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
9316 stream_.doByteSwap[mode] = true;
9318 else if ( mask & AFMT_S32_OE ) {
9319 deviceFormat = AFMT_S32_OE;
9320 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
9321 stream_.doByteSwap[mode] = true;
9323 else if ( mask & AFMT_S24_OE ) {
9324 deviceFormat = AFMT_S24_OE;
9325 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
9326 stream_.doByteSwap[mode] = true;
9328 else if ( mask & AFMT_S8) {
9329 deviceFormat = AFMT_S8;
9330 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
9334 if ( stream_.deviceFormat[mode] == 0 ) {
9335 // This really shouldn't happen ...
9337 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") data format not supported by RtAudio.";
9338 errorText_ = errorStream_.str();
9342 // Set the data format.
9343 int temp = deviceFormat;
9344 result = ioctl( fd, SNDCTL_DSP_SETFMT, &deviceFormat );
9345 if ( result == -1 || deviceFormat != temp ) {
9347 errorStream_ << "RtApiOss::probeDeviceOpen: error setting data format on device (" << ainfo.name << ").";
9348 errorText_ = errorStream_.str();
9352 // Attempt to set the buffer size. According to OSS, the minimum
9353 // number of buffers is two. The supposed minimum buffer size is 16
9354 // bytes, so that will be our lower bound. The argument to this
9355 // call is in the form 0xMMMMSSSS (hex), where the buffer size (in
9356 // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM.
9357 // We'll check the actual value used near the end of the setup
9359 int ossBufferBytes = *bufferSize * formatBytes( stream_.deviceFormat[mode] ) * deviceChannels;
9360 if ( ossBufferBytes < 16 ) ossBufferBytes = 16;
9362 if ( options ) buffers = options->numberOfBuffers;
9363 if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) buffers = 2;
9364 if ( buffers < 2 ) buffers = 3;
9365 temp = ((int) buffers << 16) + (int)( log10( (double)ossBufferBytes ) / log10( 2.0 ) );
9366 result = ioctl( fd, SNDCTL_DSP_SETFRAGMENT, &temp );
9367 if ( result == -1 ) {
9369 errorStream_ << "RtApiOss::probeDeviceOpen: error setting buffer size on device (" << ainfo.name << ").";
9370 errorText_ = errorStream_.str();
9373 stream_.nBuffers = buffers;
9375 // Save buffer size (in sample frames).
9376 *bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels );
9377 stream_.bufferSize = *bufferSize;
9379 // Set the sample rate.
9380 int srate = sampleRate;
9381 result = ioctl( fd, SNDCTL_DSP_SPEED, &srate );
9382 if ( result == -1 ) {
9384 errorStream_ << "RtApiOss::probeDeviceOpen: error setting sample rate (" << sampleRate << ") on device (" << ainfo.name << ").";
9385 errorText_ = errorStream_.str();
9389 // Verify the sample rate setup worked.
9390 if ( abs( srate - (int)sampleRate ) > 100 ) {
9392 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ").";
9393 errorText_ = errorStream_.str();
9396 stream_.sampleRate = sampleRate;
9398 if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device) {
9399 // We're doing duplex setup here.
9400 stream_.deviceFormat[0] = stream_.deviceFormat[1];
9401 stream_.nDeviceChannels[0] = deviceChannels;
9404 // Set interleaving parameters.
9405 stream_.userInterleaved = true;
9406 stream_.deviceInterleaved[mode] = true;
9407 if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
9408 stream_.userInterleaved = false;
9410 // Set flags for buffer conversion
9411 stream_.doConvertBuffer[mode] = false;
9412 if ( stream_.userFormat != stream_.deviceFormat[mode] )
9413 stream_.doConvertBuffer[mode] = true;
9414 if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
9415 stream_.doConvertBuffer[mode] = true;
9416 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
9417 stream_.nUserChannels[mode] > 1 )
9418 stream_.doConvertBuffer[mode] = true;
9420 // Allocate the stream handles if necessary and then save.
9421 if ( stream_.apiHandle == 0 ) {
9423 handle = new OssHandle;
9425 catch ( std::bad_alloc& ) {
9426 errorText_ = "RtApiOss::probeDeviceOpen: error allocating OssHandle memory.";
9430 if ( pthread_cond_init( &handle->runnable, NULL ) ) {
9431 errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable.";
9435 stream_.apiHandle = (void *) handle;
9438 handle = (OssHandle *) stream_.apiHandle;
9440 handle->id[mode] = fd;
9442 // Allocate necessary internal buffers.
9443 unsigned long bufferBytes;
9444 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
9445 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
9446 if ( stream_.userBuffer[mode] == NULL ) {
9447 errorText_ = "RtApiOss::probeDeviceOpen: error allocating user buffer memory.";
9451 if ( stream_.doConvertBuffer[mode] ) {
9453 bool makeBuffer = true;
9454 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
9455 if ( mode == INPUT ) {
9456 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
9457 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
9458 if ( bufferBytes <= bytesOut ) makeBuffer = false;
9463 bufferBytes *= *bufferSize;
9464 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
9465 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
9466 if ( stream_.deviceBuffer == NULL ) {
9467 errorText_ = "RtApiOss::probeDeviceOpen: error allocating device buffer memory.";
9473 stream_.device[mode] = device;
9474 stream_.state = STREAM_STOPPED;
9476 // Setup the buffer conversion information structure.
9477 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
9479 // Setup thread if necessary.
9480 if ( stream_.mode == OUTPUT && mode == INPUT ) {
9481 // We had already set up an output stream.
9482 stream_.mode = DUPLEX;
9483 if ( stream_.device[0] == device ) handle->id[0] = fd;
9486 stream_.mode = mode;
9488 // Setup callback thread.
9489 stream_.callbackInfo.object = (void *) this;
9491 // Set the thread attributes for joinable and realtime scheduling
9492 // priority. The higher priority will only take affect if the
9493 // program is run as root or suid.
9494 pthread_attr_t attr;
9495 pthread_attr_init( &attr );
9496 pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
9497 #ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
9498 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
9499 stream_.callbackInfo.doRealtime = true;
9500 struct sched_param param;
9501 int priority = options->priority;
9502 int min = sched_get_priority_min( SCHED_RR );
9503 int max = sched_get_priority_max( SCHED_RR );
9504 if ( priority < min ) priority = min;
9505 else if ( priority > max ) priority = max;
9506 param.sched_priority = priority;
9508 // Set the policy BEFORE the priority. Otherwise it fails.
9509 pthread_attr_setschedpolicy(&attr, SCHED_RR);
9510 pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM);
9511 // This is definitely required. Otherwise it fails.
9512 pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED);
9513 pthread_attr_setschedparam(&attr, ¶m);
9516 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
9518 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
9521 stream_.callbackInfo.isRunning = true;
9522 result = pthread_create( &stream_.callbackInfo.thread, &attr, ossCallbackHandler, &stream_.callbackInfo );
9523 pthread_attr_destroy( &attr );
9525 // Failed. Try instead with default attributes.
9526 result = pthread_create( &stream_.callbackInfo.thread, NULL, ossCallbackHandler, &stream_.callbackInfo );
9528 stream_.callbackInfo.isRunning = false;
9529 errorText_ = "RtApiOss::error creating callback thread!";
9539 pthread_cond_destroy( &handle->runnable );
9540 if ( handle->id[0] ) close( handle->id[0] );
9541 if ( handle->id[1] ) close( handle->id[1] );
9543 stream_.apiHandle = 0;
9546 for ( int i=0; i<2; i++ ) {
9547 if ( stream_.userBuffer[i] ) {
9548 free( stream_.userBuffer[i] );
9549 stream_.userBuffer[i] = 0;
9553 if ( stream_.deviceBuffer ) {
9554 free( stream_.deviceBuffer );
9555 stream_.deviceBuffer = 0;
9558 stream_.state = STREAM_CLOSED;
9562 void RtApiOss :: closeStream()
9564 if ( stream_.state == STREAM_CLOSED ) {
9565 errorText_ = "RtApiOss::closeStream(): no open stream to close!";
9566 error( RtAudioError::WARNING );
9570 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9571 stream_.callbackInfo.isRunning = false;
9572 MUTEX_LOCK( &stream_.mutex );
9573 if ( stream_.state == STREAM_STOPPED )
9574 pthread_cond_signal( &handle->runnable );
9575 MUTEX_UNLOCK( &stream_.mutex );
9576 pthread_join( stream_.callbackInfo.thread, NULL );
9578 if ( stream_.state == STREAM_RUNNING ) {
9579 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
9580 ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
9582 ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
9583 stream_.state = STREAM_STOPPED;
9587 pthread_cond_destroy( &handle->runnable );
9588 if ( handle->id[0] ) close( handle->id[0] );
9589 if ( handle->id[1] ) close( handle->id[1] );
9591 stream_.apiHandle = 0;
9594 for ( int i=0; i<2; i++ ) {
9595 if ( stream_.userBuffer[i] ) {
9596 free( stream_.userBuffer[i] );
9597 stream_.userBuffer[i] = 0;
9601 if ( stream_.deviceBuffer ) {
9602 free( stream_.deviceBuffer );
9603 stream_.deviceBuffer = 0;
9606 stream_.mode = UNINITIALIZED;
9607 stream_.state = STREAM_CLOSED;
9610 void RtApiOss :: startStream()
9613 if ( stream_.state == STREAM_RUNNING ) {
9614 errorText_ = "RtApiOss::startStream(): the stream is already running!";
9615 error( RtAudioError::WARNING );
9619 MUTEX_LOCK( &stream_.mutex );
9621 stream_.state = STREAM_RUNNING;
9623 // No need to do anything else here ... OSS automatically starts
9624 // when fed samples.
9626 MUTEX_UNLOCK( &stream_.mutex );
9628 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9629 pthread_cond_signal( &handle->runnable );
9632 void RtApiOss :: stopStream()
9635 if ( stream_.state == STREAM_STOPPED ) {
9636 errorText_ = "RtApiOss::stopStream(): the stream is already stopped!";
9637 error( RtAudioError::WARNING );
9641 MUTEX_LOCK( &stream_.mutex );
9643 // The state might change while waiting on a mutex.
9644 if ( stream_.state == STREAM_STOPPED ) {
9645 MUTEX_UNLOCK( &stream_.mutex );
9650 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9651 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
9653 // Flush the output with zeros a few times.
9656 RtAudioFormat format;
9658 if ( stream_.doConvertBuffer[0] ) {
9659 buffer = stream_.deviceBuffer;
9660 samples = stream_.bufferSize * stream_.nDeviceChannels[0];
9661 format = stream_.deviceFormat[0];
9664 buffer = stream_.userBuffer[0];
9665 samples = stream_.bufferSize * stream_.nUserChannels[0];
9666 format = stream_.userFormat;
9669 memset( buffer, 0, samples * formatBytes(format) );
9670 for ( unsigned int i=0; i<stream_.nBuffers+1; i++ ) {
9671 result = write( handle->id[0], buffer, samples * formatBytes(format) );
9672 if ( result == -1 ) {
9673 errorText_ = "RtApiOss::stopStream: audio write error.";
9674 error( RtAudioError::WARNING );
9678 result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
9679 if ( result == -1 ) {
9680 errorStream_ << "RtApiOss::stopStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
9681 errorText_ = errorStream_.str();
9684 handle->triggered = false;
9687 if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
9688 result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
9689 if ( result == -1 ) {
9690 errorStream_ << "RtApiOss::stopStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
9691 errorText_ = errorStream_.str();
9697 stream_.state = STREAM_STOPPED;
9698 MUTEX_UNLOCK( &stream_.mutex );
9700 if ( result != -1 ) return;
9701 error( RtAudioError::SYSTEM_ERROR );
9704 void RtApiOss :: abortStream()
9707 if ( stream_.state == STREAM_STOPPED ) {
9708 errorText_ = "RtApiOss::abortStream(): the stream is already stopped!";
9709 error( RtAudioError::WARNING );
9713 MUTEX_LOCK( &stream_.mutex );
9715 // The state might change while waiting on a mutex.
9716 if ( stream_.state == STREAM_STOPPED ) {
9717 MUTEX_UNLOCK( &stream_.mutex );
9722 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9723 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
9724 result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
9725 if ( result == -1 ) {
9726 errorStream_ << "RtApiOss::abortStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
9727 errorText_ = errorStream_.str();
9730 handle->triggered = false;
9733 if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
9734 result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
9735 if ( result == -1 ) {
9736 errorStream_ << "RtApiOss::abortStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
9737 errorText_ = errorStream_.str();
9743 stream_.state = STREAM_STOPPED;
9744 MUTEX_UNLOCK( &stream_.mutex );
9746 if ( result != -1 ) return;
9747 error( RtAudioError::SYSTEM_ERROR );
9750 void RtApiOss :: callbackEvent()
9752 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9753 if ( stream_.state == STREAM_STOPPED ) {
9754 MUTEX_LOCK( &stream_.mutex );
9755 pthread_cond_wait( &handle->runnable, &stream_.mutex );
9756 if ( stream_.state != STREAM_RUNNING ) {
9757 MUTEX_UNLOCK( &stream_.mutex );
9760 MUTEX_UNLOCK( &stream_.mutex );
9763 if ( stream_.state == STREAM_CLOSED ) {
9764 errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!";
9765 error( RtAudioError::WARNING );
9769 // Invoke user callback to get fresh output data.
9770 int doStopStream = 0;
9771 RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
9772 double streamTime = getStreamTime();
9773 RtAudioStreamStatus status = 0;
9774 if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
9775 status |= RTAUDIO_OUTPUT_UNDERFLOW;
9776 handle->xrun[0] = false;
9778 if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
9779 status |= RTAUDIO_INPUT_OVERFLOW;
9780 handle->xrun[1] = false;
9782 doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
9783 stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
9784 if ( doStopStream == 2 ) {
9785 this->abortStream();
9789 MUTEX_LOCK( &stream_.mutex );
9791 // The state might change while waiting on a mutex.
9792 if ( stream_.state == STREAM_STOPPED ) goto unlock;
9797 RtAudioFormat format;
9799 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
9801 // Setup parameters and do buffer conversion if necessary.
9802 if ( stream_.doConvertBuffer[0] ) {
9803 buffer = stream_.deviceBuffer;
9804 convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
9805 samples = stream_.bufferSize * stream_.nDeviceChannels[0];
9806 format = stream_.deviceFormat[0];
9809 buffer = stream_.userBuffer[0];
9810 samples = stream_.bufferSize * stream_.nUserChannels[0];
9811 format = stream_.userFormat;
9814 // Do byte swapping if necessary.
9815 if ( stream_.doByteSwap[0] )
9816 byteSwapBuffer( buffer, samples, format );
9818 if ( stream_.mode == DUPLEX && handle->triggered == false ) {
9820 ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
9821 result = write( handle->id[0], buffer, samples * formatBytes(format) );
9822 trig = PCM_ENABLE_INPUT|PCM_ENABLE_OUTPUT;
9823 ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
9824 handle->triggered = true;
9827 // Write samples to device.
9828 result = write( handle->id[0], buffer, samples * formatBytes(format) );
9830 if ( result == -1 ) {
9831 // We'll assume this is an underrun, though there isn't a
9832 // specific means for determining that.
9833 handle->xrun[0] = true;
9834 errorText_ = "RtApiOss::callbackEvent: audio write error.";
9835 error( RtAudioError::WARNING );
9836 // Continue on to input section.
9840 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
9842 // Setup parameters.
9843 if ( stream_.doConvertBuffer[1] ) {
9844 buffer = stream_.deviceBuffer;
9845 samples = stream_.bufferSize * stream_.nDeviceChannels[1];
9846 format = stream_.deviceFormat[1];
9849 buffer = stream_.userBuffer[1];
9850 samples = stream_.bufferSize * stream_.nUserChannels[1];
9851 format = stream_.userFormat;
9854 // Read samples from device.
9855 result = read( handle->id[1], buffer, samples * formatBytes(format) );
9857 if ( result == -1 ) {
9858 // We'll assume this is an overrun, though there isn't a
9859 // specific means for determining that.
9860 handle->xrun[1] = true;
9861 errorText_ = "RtApiOss::callbackEvent: audio read error.";
9862 error( RtAudioError::WARNING );
9866 // Do byte swapping if necessary.
9867 if ( stream_.doByteSwap[1] )
9868 byteSwapBuffer( buffer, samples, format );
9870 // Do buffer conversion if necessary.
9871 if ( stream_.doConvertBuffer[1] )
9872 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
9876 MUTEX_UNLOCK( &stream_.mutex );
9878 RtApi::tickStreamTime();
9879 if ( doStopStream == 1 ) this->stopStream();
9882 static void *ossCallbackHandler( void *ptr )
9884 CallbackInfo *info = (CallbackInfo *) ptr;
9885 RtApiOss *object = (RtApiOss *) info->object;
9886 bool *isRunning = &info->isRunning;
9888 #ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
9889 if (info->doRealtime) {
9890 std::cerr << "RtAudio oss: " <<
9891 (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") <<
9892 "running realtime scheduling" << std::endl;
9896 while ( *isRunning == true ) {
9897 pthread_testcancel();
9898 object->callbackEvent();
9901 pthread_exit( NULL );
9904 //******************** End of __LINUX_OSS__ *********************//
9908 // *************************************************** //
9910 // Protected common (OS-independent) RtAudio methods.
9912 // *************************************************** //
9914 // This method can be modified to control the behavior of error
9915 // message printing.
9916 void RtApi :: error( RtAudioError::Type type )
9918 errorStream_.str(""); // clear the ostringstream
9920 RtAudioErrorCallback errorCallback = (RtAudioErrorCallback) stream_.callbackInfo.errorCallback;
9921 if ( errorCallback ) {
9922 // abortStream() can generate new error messages. Ignore them. Just keep original one.
9924 if ( firstErrorOccurred_ )
9927 firstErrorOccurred_ = true;
9928 const std::string errorMessage = errorText_;
9930 if ( type != RtAudioError::WARNING && stream_.state != STREAM_STOPPED) {
9931 stream_.callbackInfo.isRunning = false; // exit from the thread
9935 errorCallback( type, errorMessage );
9936 firstErrorOccurred_ = false;
9940 if ( type == RtAudioError::WARNING && showWarnings_ == true )
9941 std::cerr << '\n' << errorText_ << "\n\n";
9942 else if ( type != RtAudioError::WARNING )
9943 throw( RtAudioError( errorText_, type ) );
9946 void RtApi :: verifyStream()
9948 if ( stream_.state == STREAM_CLOSED ) {
9949 errorText_ = "RtApi:: a stream is not open!";
9950 error( RtAudioError::INVALID_USE );
9954 void RtApi :: clearStreamInfo()
9956 stream_.mode = UNINITIALIZED;
9957 stream_.state = STREAM_CLOSED;
9958 stream_.sampleRate = 0;
9959 stream_.bufferSize = 0;
9960 stream_.nBuffers = 0;
9961 stream_.userFormat = 0;
9962 stream_.userInterleaved = true;
9963 stream_.streamTime = 0.0;
9964 stream_.apiHandle = 0;
9965 stream_.deviceBuffer = 0;
9966 stream_.callbackInfo.callback = 0;
9967 stream_.callbackInfo.userData = 0;
9968 stream_.callbackInfo.isRunning = false;
9969 stream_.callbackInfo.errorCallback = 0;
9970 for ( int i=0; i<2; i++ ) {
9971 stream_.device[i] = 11111;
9972 stream_.doConvertBuffer[i] = false;
9973 stream_.deviceInterleaved[i] = true;
9974 stream_.doByteSwap[i] = false;
9975 stream_.nUserChannels[i] = 0;
9976 stream_.nDeviceChannels[i] = 0;
9977 stream_.channelOffset[i] = 0;
9978 stream_.deviceFormat[i] = 0;
9979 stream_.latency[i] = 0;
9980 stream_.userBuffer[i] = 0;
9981 stream_.convertInfo[i].channels = 0;
9982 stream_.convertInfo[i].inJump = 0;
9983 stream_.convertInfo[i].outJump = 0;
9984 stream_.convertInfo[i].inFormat = 0;
9985 stream_.convertInfo[i].outFormat = 0;
9986 stream_.convertInfo[i].inOffset.clear();
9987 stream_.convertInfo[i].outOffset.clear();
9991 unsigned int RtApi :: formatBytes( RtAudioFormat format )
9993 if ( format == RTAUDIO_SINT16 )
9995 else if ( format == RTAUDIO_SINT32 || format == RTAUDIO_FLOAT32 )
9997 else if ( format == RTAUDIO_FLOAT64 )
9999 else if ( format == RTAUDIO_SINT24 )
10001 else if ( format == RTAUDIO_SINT8 )
10004 errorText_ = "RtApi::formatBytes: undefined format.";
10005 error( RtAudioError::WARNING );
10010 void RtApi :: setConvertInfo( StreamMode mode, unsigned int firstChannel )
10012 if ( mode == INPUT ) { // convert device to user buffer
10013 stream_.convertInfo[mode].inJump = stream_.nDeviceChannels[1];
10014 stream_.convertInfo[mode].outJump = stream_.nUserChannels[1];
10015 stream_.convertInfo[mode].inFormat = stream_.deviceFormat[1];
10016 stream_.convertInfo[mode].outFormat = stream_.userFormat;
10018 else { // convert user to device buffer
10019 stream_.convertInfo[mode].inJump = stream_.nUserChannels[0];
10020 stream_.convertInfo[mode].outJump = stream_.nDeviceChannels[0];
10021 stream_.convertInfo[mode].inFormat = stream_.userFormat;
10022 stream_.convertInfo[mode].outFormat = stream_.deviceFormat[0];
10025 if ( stream_.convertInfo[mode].inJump < stream_.convertInfo[mode].outJump )
10026 stream_.convertInfo[mode].channels = stream_.convertInfo[mode].inJump;
10028 stream_.convertInfo[mode].channels = stream_.convertInfo[mode].outJump;
10030 // Set up the interleave/deinterleave offsets.
10031 if ( stream_.deviceInterleaved[mode] != stream_.userInterleaved ) {
10032 if ( ( mode == OUTPUT && stream_.deviceInterleaved[mode] ) ||
10033 ( mode == INPUT && stream_.userInterleaved ) ) {
10034 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
10035 stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
10036 stream_.convertInfo[mode].outOffset.push_back( k );
10037 stream_.convertInfo[mode].inJump = 1;
10041 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
10042 stream_.convertInfo[mode].inOffset.push_back( k );
10043 stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
10044 stream_.convertInfo[mode].outJump = 1;
10048 else { // no (de)interleaving
10049 if ( stream_.userInterleaved ) {
10050 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
10051 stream_.convertInfo[mode].inOffset.push_back( k );
10052 stream_.convertInfo[mode].outOffset.push_back( k );
10056 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
10057 stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
10058 stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
10059 stream_.convertInfo[mode].inJump = 1;
10060 stream_.convertInfo[mode].outJump = 1;
10065 // Add channel offset.
10066 if ( firstChannel > 0 ) {
10067 if ( stream_.deviceInterleaved[mode] ) {
10068 if ( mode == OUTPUT ) {
10069 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
10070 stream_.convertInfo[mode].outOffset[k] += firstChannel;
10073 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
10074 stream_.convertInfo[mode].inOffset[k] += firstChannel;
10078 if ( mode == OUTPUT ) {
10079 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
10080 stream_.convertInfo[mode].outOffset[k] += ( firstChannel * stream_.bufferSize );
10083 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
10084 stream_.convertInfo[mode].inOffset[k] += ( firstChannel * stream_.bufferSize );
10090 void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info )
10092 // This function does format conversion, input/output channel compensation, and
10093 // data interleaving/deinterleaving. 24-bit integers are assumed to occupy
10094 // the lower three bytes of a 32-bit integer.
10096 // Clear our device buffer when in/out duplex device channels are different
10097 if ( outBuffer == stream_.deviceBuffer && stream_.mode == DUPLEX &&
10098 ( stream_.nDeviceChannels[0] < stream_.nDeviceChannels[1] ) )
10099 memset( outBuffer, 0, stream_.bufferSize * info.outJump * formatBytes( info.outFormat ) );
10102 if (info.outFormat == RTAUDIO_FLOAT64) {
10104 Float64 *out = (Float64 *)outBuffer;
10106 if (info.inFormat == RTAUDIO_SINT8) {
10107 signed char *in = (signed char *)inBuffer;
10108 scale = 1.0 / 127.5;
10109 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10110 for (j=0; j<info.channels; j++) {
10111 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
10112 out[info.outOffset[j]] += 0.5;
10113 out[info.outOffset[j]] *= scale;
10116 out += info.outJump;
10119 else if (info.inFormat == RTAUDIO_SINT16) {
10120 Int16 *in = (Int16 *)inBuffer;
10121 scale = 1.0 / 32767.5;
10122 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10123 for (j=0; j<info.channels; j++) {
10124 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
10125 out[info.outOffset[j]] += 0.5;
10126 out[info.outOffset[j]] *= scale;
10129 out += info.outJump;
10132 else if (info.inFormat == RTAUDIO_SINT24) {
10133 Int24 *in = (Int24 *)inBuffer;
10134 scale = 1.0 / 8388607.5;
10135 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10136 for (j=0; j<info.channels; j++) {
10137 out[info.outOffset[j]] = (Float64) (in[info.inOffset[j]].asInt());
10138 out[info.outOffset[j]] += 0.5;
10139 out[info.outOffset[j]] *= scale;
10142 out += info.outJump;
10145 else if (info.inFormat == RTAUDIO_SINT32) {
10146 Int32 *in = (Int32 *)inBuffer;
10147 scale = 1.0 / 2147483647.5;
10148 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10149 for (j=0; j<info.channels; j++) {
10150 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
10151 out[info.outOffset[j]] += 0.5;
10152 out[info.outOffset[j]] *= scale;
10155 out += info.outJump;
10158 else if (info.inFormat == RTAUDIO_FLOAT32) {
10159 Float32 *in = (Float32 *)inBuffer;
10160 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10161 for (j=0; j<info.channels; j++) {
10162 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
10165 out += info.outJump;
10168 else if (info.inFormat == RTAUDIO_FLOAT64) {
10169 // Channel compensation and/or (de)interleaving only.
10170 Float64 *in = (Float64 *)inBuffer;
10171 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10172 for (j=0; j<info.channels; j++) {
10173 out[info.outOffset[j]] = in[info.inOffset[j]];
10176 out += info.outJump;
10180 else if (info.outFormat == RTAUDIO_FLOAT32) {
10182 Float32 *out = (Float32 *)outBuffer;
10184 if (info.inFormat == RTAUDIO_SINT8) {
10185 signed char *in = (signed char *)inBuffer;
10186 scale = (Float32) ( 1.0 / 127.5 );
10187 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10188 for (j=0; j<info.channels; j++) {
10189 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
10190 out[info.outOffset[j]] += 0.5;
10191 out[info.outOffset[j]] *= scale;
10194 out += info.outJump;
10197 else if (info.inFormat == RTAUDIO_SINT16) {
10198 Int16 *in = (Int16 *)inBuffer;
10199 scale = (Float32) ( 1.0 / 32767.5 );
10200 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10201 for (j=0; j<info.channels; j++) {
10202 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
10203 out[info.outOffset[j]] += 0.5;
10204 out[info.outOffset[j]] *= scale;
10207 out += info.outJump;
10210 else if (info.inFormat == RTAUDIO_SINT24) {
10211 Int24 *in = (Int24 *)inBuffer;
10212 scale = (Float32) ( 1.0 / 8388607.5 );
10213 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10214 for (j=0; j<info.channels; j++) {
10215 out[info.outOffset[j]] = (Float32) (in[info.inOffset[j]].asInt());
10216 out[info.outOffset[j]] += 0.5;
10217 out[info.outOffset[j]] *= scale;
10220 out += info.outJump;
10223 else if (info.inFormat == RTAUDIO_SINT32) {
10224 Int32 *in = (Int32 *)inBuffer;
10225 scale = (Float32) ( 1.0 / 2147483647.5 );
10226 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10227 for (j=0; j<info.channels; j++) {
10228 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
10229 out[info.outOffset[j]] += 0.5;
10230 out[info.outOffset[j]] *= scale;
10233 out += info.outJump;
10236 else if (info.inFormat == RTAUDIO_FLOAT32) {
10237 // Channel compensation and/or (de)interleaving only.
10238 Float32 *in = (Float32 *)inBuffer;
10239 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10240 for (j=0; j<info.channels; j++) {
10241 out[info.outOffset[j]] = in[info.inOffset[j]];
10244 out += info.outJump;
10247 else if (info.inFormat == RTAUDIO_FLOAT64) {
10248 Float64 *in = (Float64 *)inBuffer;
10249 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10250 for (j=0; j<info.channels; j++) {
10251 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
10254 out += info.outJump;
10258 else if (info.outFormat == RTAUDIO_SINT32) {
10259 Int32 *out = (Int32 *)outBuffer;
10260 if (info.inFormat == RTAUDIO_SINT8) {
10261 signed char *in = (signed char *)inBuffer;
10262 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10263 for (j=0; j<info.channels; j++) {
10264 out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
10265 out[info.outOffset[j]] <<= 24;
10268 out += info.outJump;
10271 else if (info.inFormat == RTAUDIO_SINT16) {
10272 Int16 *in = (Int16 *)inBuffer;
10273 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10274 for (j=0; j<info.channels; j++) {
10275 out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
10276 out[info.outOffset[j]] <<= 16;
10279 out += info.outJump;
10282 else if (info.inFormat == RTAUDIO_SINT24) {
10283 Int24 *in = (Int24 *)inBuffer;
10284 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10285 for (j=0; j<info.channels; j++) {
10286 out[info.outOffset[j]] = (Int32) in[info.inOffset[j]].asInt();
10287 out[info.outOffset[j]] <<= 8;
10290 out += info.outJump;
10293 else if (info.inFormat == RTAUDIO_SINT32) {
10294 // Channel compensation and/or (de)interleaving only.
10295 Int32 *in = (Int32 *)inBuffer;
10296 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10297 for (j=0; j<info.channels; j++) {
10298 out[info.outOffset[j]] = in[info.inOffset[j]];
10301 out += info.outJump;
10304 else if (info.inFormat == RTAUDIO_FLOAT32) {
10305 Float32 *in = (Float32 *)inBuffer;
10306 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10307 for (j=0; j<info.channels; j++) {
10308 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
10311 out += info.outJump;
10314 else if (info.inFormat == RTAUDIO_FLOAT64) {
10315 Float64 *in = (Float64 *)inBuffer;
10316 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10317 for (j=0; j<info.channels; j++) {
10318 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
10321 out += info.outJump;
10325 else if (info.outFormat == RTAUDIO_SINT24) {
10326 Int24 *out = (Int24 *)outBuffer;
10327 if (info.inFormat == RTAUDIO_SINT8) {
10328 signed char *in = (signed char *)inBuffer;
10329 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10330 for (j=0; j<info.channels; j++) {
10331 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] << 16);
10332 //out[info.outOffset[j]] <<= 16;
10335 out += info.outJump;
10338 else if (info.inFormat == RTAUDIO_SINT16) {
10339 Int16 *in = (Int16 *)inBuffer;
10340 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10341 for (j=0; j<info.channels; j++) {
10342 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] << 8);
10343 //out[info.outOffset[j]] <<= 8;
10346 out += info.outJump;
10349 else if (info.inFormat == RTAUDIO_SINT24) {
10350 // Channel compensation and/or (de)interleaving only.
10351 Int24 *in = (Int24 *)inBuffer;
10352 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10353 for (j=0; j<info.channels; j++) {
10354 out[info.outOffset[j]] = in[info.inOffset[j]];
10357 out += info.outJump;
10360 else if (info.inFormat == RTAUDIO_SINT32) {
10361 Int32 *in = (Int32 *)inBuffer;
10362 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10363 for (j=0; j<info.channels; j++) {
10364 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] >> 8);
10365 //out[info.outOffset[j]] >>= 8;
10368 out += info.outJump;
10371 else if (info.inFormat == RTAUDIO_FLOAT32) {
10372 Float32 *in = (Float32 *)inBuffer;
10373 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10374 for (j=0; j<info.channels; j++) {
10375 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
10378 out += info.outJump;
10381 else if (info.inFormat == RTAUDIO_FLOAT64) {
10382 Float64 *in = (Float64 *)inBuffer;
10383 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10384 for (j=0; j<info.channels; j++) {
10385 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
10388 out += info.outJump;
10392 else if (info.outFormat == RTAUDIO_SINT16) {
10393 Int16 *out = (Int16 *)outBuffer;
10394 if (info.inFormat == RTAUDIO_SINT8) {
10395 signed char *in = (signed char *)inBuffer;
10396 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10397 for (j=0; j<info.channels; j++) {
10398 out[info.outOffset[j]] = (Int16) in[info.inOffset[j]];
10399 out[info.outOffset[j]] <<= 8;
10402 out += info.outJump;
10405 else if (info.inFormat == RTAUDIO_SINT16) {
10406 // Channel compensation and/or (de)interleaving only.
10407 Int16 *in = (Int16 *)inBuffer;
10408 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10409 for (j=0; j<info.channels; j++) {
10410 out[info.outOffset[j]] = in[info.inOffset[j]];
10413 out += info.outJump;
10416 else if (info.inFormat == RTAUDIO_SINT24) {
10417 Int24 *in = (Int24 *)inBuffer;
10418 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10419 for (j=0; j<info.channels; j++) {
10420 out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]].asInt() >> 8);
10423 out += info.outJump;
10426 else if (info.inFormat == RTAUDIO_SINT32) {
10427 Int32 *in = (Int32 *)inBuffer;
10428 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10429 for (j=0; j<info.channels; j++) {
10430 out[info.outOffset[j]] = (Int16) ((in[info.inOffset[j]] >> 16) & 0x0000ffff);
10433 out += info.outJump;
10436 else if (info.inFormat == RTAUDIO_FLOAT32) {
10437 Float32 *in = (Float32 *)inBuffer;
10438 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10439 for (j=0; j<info.channels; j++) {
10440 out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
10443 out += info.outJump;
10446 else if (info.inFormat == RTAUDIO_FLOAT64) {
10447 Float64 *in = (Float64 *)inBuffer;
10448 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10449 for (j=0; j<info.channels; j++) {
10450 out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
10453 out += info.outJump;
10457 else if (info.outFormat == RTAUDIO_SINT8) {
10458 signed char *out = (signed char *)outBuffer;
10459 if (info.inFormat == RTAUDIO_SINT8) {
10460 // Channel compensation and/or (de)interleaving only.
10461 signed char *in = (signed char *)inBuffer;
10462 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10463 for (j=0; j<info.channels; j++) {
10464 out[info.outOffset[j]] = in[info.inOffset[j]];
10467 out += info.outJump;
10470 if (info.inFormat == RTAUDIO_SINT16) {
10471 Int16 *in = (Int16 *)inBuffer;
10472 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10473 for (j=0; j<info.channels; j++) {
10474 out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 8) & 0x00ff);
10477 out += info.outJump;
10480 else if (info.inFormat == RTAUDIO_SINT24) {
10481 Int24 *in = (Int24 *)inBuffer;
10482 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10483 for (j=0; j<info.channels; j++) {
10484 out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]].asInt() >> 16);
10487 out += info.outJump;
10490 else if (info.inFormat == RTAUDIO_SINT32) {
10491 Int32 *in = (Int32 *)inBuffer;
10492 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10493 for (j=0; j<info.channels; j++) {
10494 out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 24) & 0x000000ff);
10497 out += info.outJump;
10500 else if (info.inFormat == RTAUDIO_FLOAT32) {
10501 Float32 *in = (Float32 *)inBuffer;
10502 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10503 for (j=0; j<info.channels; j++) {
10504 out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
10507 out += info.outJump;
10510 else if (info.inFormat == RTAUDIO_FLOAT64) {
10511 Float64 *in = (Float64 *)inBuffer;
10512 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10513 for (j=0; j<info.channels; j++) {
10514 out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
10517 out += info.outJump;
10523 //static inline uint16_t bswap_16(uint16_t x) { return (x>>8) | (x<<8); }
10524 //static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); }
10525 //static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); }
10527 void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format )
10533 if ( format == RTAUDIO_SINT16 ) {
10534 for ( unsigned int i=0; i<samples; i++ ) {
10535 // Swap 1st and 2nd bytes.
10540 // Increment 2 bytes.
10544 else if ( format == RTAUDIO_SINT32 ||
10545 format == RTAUDIO_FLOAT32 ) {
10546 for ( unsigned int i=0; i<samples; i++ ) {
10547 // Swap 1st and 4th bytes.
10552 // Swap 2nd and 3rd bytes.
10558 // Increment 3 more bytes.
10562 else if ( format == RTAUDIO_SINT24 ) {
10563 for ( unsigned int i=0; i<samples; i++ ) {
10564 // Swap 1st and 3rd bytes.
10569 // Increment 2 more bytes.
10573 else if ( format == RTAUDIO_FLOAT64 ) {
10574 for ( unsigned int i=0; i<samples; i++ ) {
10575 // Swap 1st and 8th bytes
10580 // Swap 2nd and 7th bytes
10586 // Swap 3rd and 6th bytes
10592 // Swap 4th and 5th bytes
10598 // Increment 5 more bytes.
10604 // Indentation settings for Vim and Emacs
10606 // Local Variables:
10607 // c-basic-offset: 2
10608 // indent-tabs-mode: nil
10611 // vim: et sts=2 sw=2