1 /************************************************************************/
3 \brief Realtime audio i/o C++ classes.
5 RtAudio provides a common API (Application Programming Interface)
6 for realtime audio input/output across Linux (native ALSA, Jack,
7 and OSS), Macintosh OS X (CoreAudio and Jack), and Windows
8 (DirectSound, ASIO and WASAPI) operating systems.
10 RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/
12 RtAudio: realtime audio i/o C++ classes
13 Copyright (c) 2001-2017 Gary P. Scavone
15 Permission is hereby granted, free of charge, to any person
16 obtaining a copy of this software and associated documentation files
17 (the "Software"), to deal in the Software without restriction,
18 including without limitation the rights to use, copy, modify, merge,
19 publish, distribute, sublicense, and/or sell copies of the Software,
20 and to permit persons to whom the Software is furnished to do so,
21 subject to the following conditions:
23 The above copyright notice and this permission notice shall be
24 included in all copies or substantial portions of the Software.
26 Any person wishing to distribute modifications to the Software is
27 asked to send the modifications to the original developer so that
28 they can be incorporated into the canonical version. This is,
29 however, not a binding provision of this license.
31 THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
32 EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
33 MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
34 IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
35 ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
36 CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
37 WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
39 /************************************************************************/
41 // RtAudio: Version 5.0.0
51 // Static variable definitions.
52 const unsigned int RtApi::MAX_SAMPLE_RATES = 14;
53 const unsigned int RtApi::SAMPLE_RATES[] = {
54 4000, 5512, 8000, 9600, 11025, 16000, 22050,
55 32000, 44100, 48000, 88200, 96000, 176400, 192000
58 #if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) || defined(__WINDOWS_WASAPI__)
59 #define MUTEX_INITIALIZE(A) InitializeCriticalSection(A)
60 #define MUTEX_DESTROY(A) DeleteCriticalSection(A)
61 #define MUTEX_LOCK(A) EnterCriticalSection(A)
62 #define MUTEX_UNLOCK(A) LeaveCriticalSection(A)
66 static std::string convertCharPointerToStdString(const char *text)
68 return std::string(text);
71 static std::string convertCharPointerToStdString(const wchar_t *text)
73 int length = WideCharToMultiByte(CP_UTF8, 0, text, -1, NULL, 0, NULL, NULL);
74 std::string s( length-1, '\0' );
75 WideCharToMultiByte(CP_UTF8, 0, text, -1, &s[0], length, NULL, NULL);
79 #elif defined(__LINUX_ALSA__) || defined(__LINUX_PULSE__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__)
81 #define MUTEX_INITIALIZE(A) pthread_mutex_init(A, NULL)
82 #define MUTEX_DESTROY(A) pthread_mutex_destroy(A)
83 #define MUTEX_LOCK(A) pthread_mutex_lock(A)
84 #define MUTEX_UNLOCK(A) pthread_mutex_unlock(A)
86 #define MUTEX_INITIALIZE(A) abs(*A) // dummy definitions
87 #define MUTEX_DESTROY(A) abs(*A) // dummy definitions
90 // *************************************************** //
92 // RtAudio definitions.
94 // *************************************************** //
96 std::string RtAudio :: getVersion( void )
98 return RTAUDIO_VERSION;
101 // Define API names and display names.
102 // Must be in same order as API enum.
104 const char* rtaudio_api_names[][2] = {
105 { "unspecified" , "Unknown" },
107 { "pulse" , "Pulse" },
108 { "oss" , "OpenSoundSystem" },
110 { "core" , "CoreAudio" },
111 { "wasapi" , "WASAPI" },
113 { "ds" , "DirectSound" },
114 { "dummy" , "Dummy" },
116 const unsigned int rtaudio_num_api_names =
117 sizeof(rtaudio_api_names)/sizeof(rtaudio_api_names[0]);
119 // The order here will control the order of RtAudio's API search in
121 extern "C" const RtAudio::Api rtaudio_compiled_apis[] = {
122 #if defined(__UNIX_JACK__)
125 #if defined(__LINUX_PULSE__)
126 RtAudio::LINUX_PULSE,
128 #if defined(__LINUX_ALSA__)
131 #if defined(__LINUX_OSS__)
134 #if defined(__WINDOWS_ASIO__)
135 RtAudio::WINDOWS_ASIO,
137 #if defined(__WINDOWS_WASAPI__)
138 RtAudio::WINDOWS_WASAPI,
140 #if defined(__WINDOWS_DS__)
143 #if defined(__MACOSX_CORE__)
144 RtAudio::MACOSX_CORE,
146 #if defined(__RTAUDIO_DUMMY__)
147 RtAudio::RTAUDIO_DUMMY,
149 RtAudio::UNSPECIFIED,
151 extern "C" const unsigned int rtaudio_num_compiled_apis =
152 sizeof(rtaudio_compiled_apis)/sizeof(rtaudio_compiled_apis[0])-1;
155 // This is a compile-time check that rtaudio_num_api_names == RtAudio::NUM_APIS.
156 // If the build breaks here, check that they match.
157 template<bool b> class StaticAssert { private: StaticAssert() {} };
158 template<> class StaticAssert<true>{ public: StaticAssert() {} };
159 class StaticAssertions { StaticAssertions() {
160 StaticAssert<rtaudio_num_api_names == RtAudio::NUM_APIS>();
163 void RtAudio :: getCompiledApi( std::vector<RtAudio::Api> &apis )
165 apis = std::vector<RtAudio::Api>(rtaudio_compiled_apis,
166 rtaudio_compiled_apis + rtaudio_num_compiled_apis);
169 std::string RtAudio :: getApiName( RtAudio::Api api )
171 if (api < 0 || api >= RtAudio::NUM_APIS)
173 return rtaudio_api_names[api][0];
176 std::string RtAudio :: getApiDisplayName( RtAudio::Api api )
178 if (api < 0 || api >= RtAudio::NUM_APIS)
180 return rtaudio_api_names[api][1];
183 RtAudio::Api RtAudio :: getCompiledApiByName( const std::string &name )
186 for (i = 0; i < rtaudio_num_compiled_apis; ++i)
187 if (name == rtaudio_api_names[rtaudio_compiled_apis[i]][0])
188 return rtaudio_compiled_apis[i];
189 return RtAudio::UNSPECIFIED;
192 void RtAudio :: openRtApi( RtAudio::Api api )
198 #if defined(__UNIX_JACK__)
199 if ( api == UNIX_JACK )
200 rtapi_ = new RtApiJack();
202 #if defined(__LINUX_ALSA__)
203 if ( api == LINUX_ALSA )
204 rtapi_ = new RtApiAlsa();
206 #if defined(__LINUX_PULSE__)
207 if ( api == LINUX_PULSE )
208 rtapi_ = new RtApiPulse();
210 #if defined(__LINUX_OSS__)
211 if ( api == LINUX_OSS )
212 rtapi_ = new RtApiOss();
214 #if defined(__WINDOWS_ASIO__)
215 if ( api == WINDOWS_ASIO )
216 rtapi_ = new RtApiAsio();
218 #if defined(__WINDOWS_WASAPI__)
219 if ( api == WINDOWS_WASAPI )
220 rtapi_ = new RtApiWasapi();
222 #if defined(__WINDOWS_DS__)
223 if ( api == WINDOWS_DS )
224 rtapi_ = new RtApiDs();
226 #if defined(__MACOSX_CORE__)
227 if ( api == MACOSX_CORE )
228 rtapi_ = new RtApiCore();
230 #if defined(__RTAUDIO_DUMMY__)
231 if ( api == RTAUDIO_DUMMY )
232 rtapi_ = new RtApiDummy();
236 RtAudio :: RtAudio( RtAudio::Api api )
240 if ( api != UNSPECIFIED ) {
241 // Attempt to open the specified API.
243 if ( rtapi_ ) return;
245 // No compiled support for specified API value. Issue a debug
246 // warning and continue as if no API was specified.
247 std::cerr << "\nRtAudio: no compiled support for specified API argument!\n" << std::endl;
250 // Iterate through the compiled APIs and return as soon as we find
251 // one with at least one device or we reach the end of the list.
252 std::vector< RtAudio::Api > apis;
253 getCompiledApi( apis );
254 for ( unsigned int i=0; i<apis.size(); i++ ) {
255 openRtApi( apis[i] );
256 if ( rtapi_ && rtapi_->getDeviceCount() ) break;
259 if ( rtapi_ ) return;
261 // It should not be possible to get here because the preprocessor
262 // definition __RTAUDIO_DUMMY__ is automatically defined if no
263 // API-specific definitions are passed to the compiler. But just in
264 // case something weird happens, we'll thow an error.
265 std::string errorText = "\nRtAudio: no compiled API support found ... critical error!!\n\n";
266 throw( RtAudioError( errorText, RtAudioError::UNSPECIFIED ) );
269 RtAudio :: ~RtAudio()
275 void RtAudio :: openStream( RtAudio::StreamParameters *outputParameters,
276 RtAudio::StreamParameters *inputParameters,
277 RtAudioFormat format, unsigned int sampleRate,
278 unsigned int *bufferFrames,
279 RtAudioCallback callback, void *userData,
280 RtAudio::StreamOptions *options,
281 RtAudioErrorCallback errorCallback )
283 return rtapi_->openStream( outputParameters, inputParameters, format,
284 sampleRate, bufferFrames, callback,
285 userData, options, errorCallback );
288 // *************************************************** //
290 // Public RtApi definitions (see end of file for
291 // private or protected utility functions).
293 // *************************************************** //
297 stream_.state = STREAM_CLOSED;
298 stream_.mode = UNINITIALIZED;
299 stream_.apiHandle = 0;
300 stream_.userBuffer[0] = 0;
301 stream_.userBuffer[1] = 0;
302 MUTEX_INITIALIZE( &stream_.mutex );
303 showWarnings_ = true;
304 firstErrorOccurred_ = false;
309 MUTEX_DESTROY( &stream_.mutex );
312 void RtApi :: openStream( RtAudio::StreamParameters *oParams,
313 RtAudio::StreamParameters *iParams,
314 RtAudioFormat format, unsigned int sampleRate,
315 unsigned int *bufferFrames,
316 RtAudioCallback callback, void *userData,
317 RtAudio::StreamOptions *options,
318 RtAudioErrorCallback errorCallback )
320 if ( stream_.state != STREAM_CLOSED ) {
321 errorText_ = "RtApi::openStream: a stream is already open!";
322 error( RtAudioError::INVALID_USE );
326 // Clear stream information potentially left from a previously open stream.
329 if ( oParams && oParams->nChannels < 1 ) {
330 errorText_ = "RtApi::openStream: a non-NULL output StreamParameters structure cannot have an nChannels value less than one.";
331 error( RtAudioError::INVALID_USE );
335 if ( iParams && iParams->nChannels < 1 ) {
336 errorText_ = "RtApi::openStream: a non-NULL input StreamParameters structure cannot have an nChannels value less than one.";
337 error( RtAudioError::INVALID_USE );
341 if ( oParams == NULL && iParams == NULL ) {
342 errorText_ = "RtApi::openStream: input and output StreamParameters structures are both NULL!";
343 error( RtAudioError::INVALID_USE );
347 if ( formatBytes(format) == 0 ) {
348 errorText_ = "RtApi::openStream: 'format' parameter value is undefined.";
349 error( RtAudioError::INVALID_USE );
353 unsigned int nDevices = getDeviceCount();
354 unsigned int oChannels = 0;
356 oChannels = oParams->nChannels;
357 if ( oParams->deviceId >= nDevices ) {
358 errorText_ = "RtApi::openStream: output device parameter value is invalid.";
359 error( RtAudioError::INVALID_USE );
364 unsigned int iChannels = 0;
366 iChannels = iParams->nChannels;
367 if ( iParams->deviceId >= nDevices ) {
368 errorText_ = "RtApi::openStream: input device parameter value is invalid.";
369 error( RtAudioError::INVALID_USE );
376 if ( oChannels > 0 ) {
378 result = probeDeviceOpen( oParams->deviceId, OUTPUT, oChannels, oParams->firstChannel,
379 sampleRate, format, bufferFrames, options );
380 if ( result == false ) {
381 error( RtAudioError::SYSTEM_ERROR );
386 if ( iChannels > 0 ) {
388 result = probeDeviceOpen( iParams->deviceId, INPUT, iChannels, iParams->firstChannel,
389 sampleRate, format, bufferFrames, options );
390 if ( result == false ) {
391 if ( oChannels > 0 ) closeStream();
392 error( RtAudioError::SYSTEM_ERROR );
397 stream_.callbackInfo.callback = (void *) callback;
398 stream_.callbackInfo.userData = userData;
399 stream_.callbackInfo.errorCallback = (void *) errorCallback;
401 if ( options ) options->numberOfBuffers = stream_.nBuffers;
402 stream_.state = STREAM_STOPPED;
405 unsigned int RtApi :: getDefaultInputDevice( void )
407 // Should be implemented in subclasses if possible.
411 unsigned int RtApi :: getDefaultOutputDevice( void )
413 // Should be implemented in subclasses if possible.
417 void RtApi :: closeStream( void )
419 // MUST be implemented in subclasses!
423 bool RtApi :: probeDeviceOpen( unsigned int /*device*/, StreamMode /*mode*/, unsigned int /*channels*/,
424 unsigned int /*firstChannel*/, unsigned int /*sampleRate*/,
425 RtAudioFormat /*format*/, unsigned int * /*bufferSize*/,
426 RtAudio::StreamOptions * /*options*/ )
428 // MUST be implemented in subclasses!
432 void RtApi :: tickStreamTime( void )
434 // Subclasses that do not provide their own implementation of
435 // getStreamTime should call this function once per buffer I/O to
436 // provide basic stream time support.
438 stream_.streamTime += ( stream_.bufferSize * 1.0 / stream_.sampleRate );
440 #if defined( HAVE_GETTIMEOFDAY )
441 gettimeofday( &stream_.lastTickTimestamp, NULL );
445 long RtApi :: getStreamLatency( void )
449 long totalLatency = 0;
450 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
451 totalLatency = stream_.latency[0];
452 if ( stream_.mode == INPUT || stream_.mode == DUPLEX )
453 totalLatency += stream_.latency[1];
458 double RtApi :: getStreamTime( void )
462 #if defined( HAVE_GETTIMEOFDAY )
463 // Return a very accurate estimate of the stream time by
464 // adding in the elapsed time since the last tick.
468 if ( stream_.state != STREAM_RUNNING || stream_.streamTime == 0.0 )
469 return stream_.streamTime;
471 gettimeofday( &now, NULL );
472 then = stream_.lastTickTimestamp;
473 return stream_.streamTime +
474 ((now.tv_sec + 0.000001 * now.tv_usec) -
475 (then.tv_sec + 0.000001 * then.tv_usec));
477 return stream_.streamTime;
481 void RtApi :: setStreamTime( double time )
486 stream_.streamTime = time;
487 #if defined( HAVE_GETTIMEOFDAY )
488 gettimeofday( &stream_.lastTickTimestamp, NULL );
492 unsigned int RtApi :: getStreamSampleRate( void )
496 return stream_.sampleRate;
500 // *************************************************** //
502 // OS/API-specific methods.
504 // *************************************************** //
506 #if defined(__MACOSX_CORE__)
508 // The OS X CoreAudio API is designed to use a separate callback
509 // procedure for each of its audio devices. A single RtAudio duplex
510 // stream using two different devices is supported here, though it
511 // cannot be guaranteed to always behave correctly because we cannot
512 // synchronize these two callbacks.
514 // A property listener is installed for over/underrun information.
515 // However, no functionality is currently provided to allow property
516 // listeners to trigger user handlers because it is unclear what could
517 // be done if a critical stream parameter (buffer size, sample rate,
518 // device disconnect) notification arrived. The listeners entail
519 // quite a bit of extra code and most likely, a user program wouldn't
520 // be prepared for the result anyway. However, we do provide a flag
521 // to the client callback function to inform of an over/underrun.
523 // A structure to hold various information related to the CoreAudio API
526 AudioDeviceID id[2]; // device ids
527 #if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
528 AudioDeviceIOProcID procId[2];
530 UInt32 iStream[2]; // device stream index (or first if using multiple)
531 UInt32 nStreams[2]; // number of streams to use
534 pthread_cond_t condition;
535 int drainCounter; // Tracks callback counts when draining
536 bool internalDrain; // Indicates if stop is initiated from callback or not.
539 :deviceBuffer(0), drainCounter(0), internalDrain(false) { nStreams[0] = 1; nStreams[1] = 1; id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
542 RtApiCore:: RtApiCore()
544 #if defined( AVAILABLE_MAC_OS_X_VERSION_10_6_AND_LATER )
545 // This is a largely undocumented but absolutely necessary
546 // requirement starting with OS-X 10.6. If not called, queries and
547 // updates to various audio device properties are not handled
549 CFRunLoopRef theRunLoop = NULL;
550 AudioObjectPropertyAddress property = { kAudioHardwarePropertyRunLoop,
551 kAudioObjectPropertyScopeGlobal,
552 kAudioObjectPropertyElementMaster };
553 OSStatus result = AudioObjectSetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, sizeof(CFRunLoopRef), &theRunLoop);
554 if ( result != noErr ) {
555 errorText_ = "RtApiCore::RtApiCore: error setting run loop property!";
556 error( RtAudioError::WARNING );
561 RtApiCore :: ~RtApiCore()
563 // The subclass destructor gets called before the base class
564 // destructor, so close an existing stream before deallocating
565 // apiDeviceId memory.
566 if ( stream_.state != STREAM_CLOSED ) closeStream();
569 unsigned int RtApiCore :: getDeviceCount( void )
571 // Find out how many audio devices there are, if any.
573 AudioObjectPropertyAddress propertyAddress = { kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
574 OSStatus result = AudioObjectGetPropertyDataSize( kAudioObjectSystemObject, &propertyAddress, 0, NULL, &dataSize );
575 if ( result != noErr ) {
576 errorText_ = "RtApiCore::getDeviceCount: OS-X error getting device info!";
577 error( RtAudioError::WARNING );
581 return dataSize / sizeof( AudioDeviceID );
584 unsigned int RtApiCore :: getDefaultInputDevice( void )
586 unsigned int nDevices = getDeviceCount();
587 if ( nDevices <= 1 ) return 0;
590 UInt32 dataSize = sizeof( AudioDeviceID );
591 AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultInputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
592 OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id );
593 if ( result != noErr ) {
594 errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device.";
595 error( RtAudioError::WARNING );
599 dataSize *= nDevices;
600 AudioDeviceID deviceList[ nDevices ];
601 property.mSelector = kAudioHardwarePropertyDevices;
602 result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList );
603 if ( result != noErr ) {
604 errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device IDs.";
605 error( RtAudioError::WARNING );
609 for ( unsigned int i=0; i<nDevices; i++ )
610 if ( id == deviceList[i] ) return i;
612 errorText_ = "RtApiCore::getDefaultInputDevice: No default device found!";
613 error( RtAudioError::WARNING );
617 unsigned int RtApiCore :: getDefaultOutputDevice( void )
619 unsigned int nDevices = getDeviceCount();
620 if ( nDevices <= 1 ) return 0;
623 UInt32 dataSize = sizeof( AudioDeviceID );
624 AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultOutputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
625 OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id );
626 if ( result != noErr ) {
627 errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device.";
628 error( RtAudioError::WARNING );
632 dataSize = sizeof( AudioDeviceID ) * nDevices;
633 AudioDeviceID deviceList[ nDevices ];
634 property.mSelector = kAudioHardwarePropertyDevices;
635 result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList );
636 if ( result != noErr ) {
637 errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device IDs.";
638 error( RtAudioError::WARNING );
642 for ( unsigned int i=0; i<nDevices; i++ )
643 if ( id == deviceList[i] ) return i;
645 errorText_ = "RtApiCore::getDefaultOutputDevice: No default device found!";
646 error( RtAudioError::WARNING );
650 RtAudio::DeviceInfo RtApiCore :: getDeviceInfo( unsigned int device )
652 RtAudio::DeviceInfo info;
656 unsigned int nDevices = getDeviceCount();
657 if ( nDevices == 0 ) {
658 errorText_ = "RtApiCore::getDeviceInfo: no devices found!";
659 error( RtAudioError::INVALID_USE );
663 if ( device >= nDevices ) {
664 errorText_ = "RtApiCore::getDeviceInfo: device ID is invalid!";
665 error( RtAudioError::INVALID_USE );
669 AudioDeviceID deviceList[ nDevices ];
670 UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;
671 AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
672 kAudioObjectPropertyScopeGlobal,
673 kAudioObjectPropertyElementMaster };
674 OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property,
675 0, NULL, &dataSize, (void *) &deviceList );
676 if ( result != noErr ) {
677 errorText_ = "RtApiCore::getDeviceInfo: OS-X system error getting device IDs.";
678 error( RtAudioError::WARNING );
682 AudioDeviceID id = deviceList[ device ];
684 // Get the device name.
687 dataSize = sizeof( CFStringRef );
688 property.mSelector = kAudioObjectPropertyManufacturer;
689 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname );
690 if ( result != noErr ) {
691 errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device manufacturer.";
692 errorText_ = errorStream_.str();
693 error( RtAudioError::WARNING );
697 //const char *mname = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() );
698 int length = CFStringGetLength(cfname);
699 char *mname = (char *)malloc(length * 3 + 1);
700 #if defined( UNICODE ) || defined( _UNICODE )
701 CFStringGetCString(cfname, mname, length * 3 + 1, kCFStringEncodingUTF8);
703 CFStringGetCString(cfname, mname, length * 3 + 1, CFStringGetSystemEncoding());
705 info.name.append( (const char *)mname, strlen(mname) );
706 info.name.append( ": " );
710 property.mSelector = kAudioObjectPropertyName;
711 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname );
712 if ( result != noErr ) {
713 errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device name.";
714 errorText_ = errorStream_.str();
715 error( RtAudioError::WARNING );
719 //const char *name = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() );
720 length = CFStringGetLength(cfname);
721 char *name = (char *)malloc(length * 3 + 1);
722 #if defined( UNICODE ) || defined( _UNICODE )
723 CFStringGetCString(cfname, name, length * 3 + 1, kCFStringEncodingUTF8);
725 CFStringGetCString(cfname, name, length * 3 + 1, CFStringGetSystemEncoding());
727 info.name.append( (const char *)name, strlen(name) );
731 // Get the output stream "configuration".
732 AudioBufferList *bufferList = nil;
733 property.mSelector = kAudioDevicePropertyStreamConfiguration;
734 property.mScope = kAudioDevicePropertyScopeOutput;
735 // property.mElement = kAudioObjectPropertyElementWildcard;
737 result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
738 if ( result != noErr || dataSize == 0 ) {
739 errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration info for device (" << device << ").";
740 errorText_ = errorStream_.str();
741 error( RtAudioError::WARNING );
745 // Allocate the AudioBufferList.
746 bufferList = (AudioBufferList *) malloc( dataSize );
747 if ( bufferList == NULL ) {
748 errorText_ = "RtApiCore::getDeviceInfo: memory error allocating output AudioBufferList.";
749 error( RtAudioError::WARNING );
753 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
754 if ( result != noErr || dataSize == 0 ) {
756 errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration for device (" << device << ").";
757 errorText_ = errorStream_.str();
758 error( RtAudioError::WARNING );
762 // Get output channel information.
763 unsigned int i, nStreams = bufferList->mNumberBuffers;
764 for ( i=0; i<nStreams; i++ )
765 info.outputChannels += bufferList->mBuffers[i].mNumberChannels;
768 // Get the input stream "configuration".
769 property.mScope = kAudioDevicePropertyScopeInput;
770 result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
771 if ( result != noErr || dataSize == 0 ) {
772 errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration info for device (" << device << ").";
773 errorText_ = errorStream_.str();
774 error( RtAudioError::WARNING );
778 // Allocate the AudioBufferList.
779 bufferList = (AudioBufferList *) malloc( dataSize );
780 if ( bufferList == NULL ) {
781 errorText_ = "RtApiCore::getDeviceInfo: memory error allocating input AudioBufferList.";
782 error( RtAudioError::WARNING );
786 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
787 if (result != noErr || dataSize == 0) {
789 errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration for device (" << device << ").";
790 errorText_ = errorStream_.str();
791 error( RtAudioError::WARNING );
795 // Get input channel information.
796 nStreams = bufferList->mNumberBuffers;
797 for ( i=0; i<nStreams; i++ )
798 info.inputChannels += bufferList->mBuffers[i].mNumberChannels;
801 // If device opens for both playback and capture, we determine the channels.
802 if ( info.outputChannels > 0 && info.inputChannels > 0 )
803 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
805 // Probe the device sample rates.
806 bool isInput = false;
807 if ( info.outputChannels == 0 ) isInput = true;
809 // Determine the supported sample rates.
810 property.mSelector = kAudioDevicePropertyAvailableNominalSampleRates;
811 if ( isInput == false ) property.mScope = kAudioDevicePropertyScopeOutput;
812 result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
813 if ( result != kAudioHardwareNoError || dataSize == 0 ) {
814 errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rate info.";
815 errorText_ = errorStream_.str();
816 error( RtAudioError::WARNING );
820 UInt32 nRanges = dataSize / sizeof( AudioValueRange );
821 AudioValueRange rangeList[ nRanges ];
822 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &rangeList );
823 if ( result != kAudioHardwareNoError ) {
824 errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rates.";
825 errorText_ = errorStream_.str();
826 error( RtAudioError::WARNING );
830 // The sample rate reporting mechanism is a bit of a mystery. It
831 // seems that it can either return individual rates or a range of
832 // rates. I assume that if the min / max range values are the same,
833 // then that represents a single supported rate and if the min / max
834 // range values are different, the device supports an arbitrary
835 // range of values (though there might be multiple ranges, so we'll
836 // use the most conservative range).
837 Float64 minimumRate = 1.0, maximumRate = 10000000000.0;
838 bool haveValueRange = false;
839 info.sampleRates.clear();
840 for ( UInt32 i=0; i<nRanges; i++ ) {
841 if ( rangeList[i].mMinimum == rangeList[i].mMaximum ) {
842 unsigned int tmpSr = (unsigned int) rangeList[i].mMinimum;
843 info.sampleRates.push_back( tmpSr );
845 if ( !info.preferredSampleRate || ( tmpSr <= 48000 && tmpSr > info.preferredSampleRate ) )
846 info.preferredSampleRate = tmpSr;
849 haveValueRange = true;
850 if ( rangeList[i].mMinimum > minimumRate ) minimumRate = rangeList[i].mMinimum;
851 if ( rangeList[i].mMaximum < maximumRate ) maximumRate = rangeList[i].mMaximum;
855 if ( haveValueRange ) {
856 for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
857 if ( SAMPLE_RATES[k] >= (unsigned int) minimumRate && SAMPLE_RATES[k] <= (unsigned int) maximumRate ) {
858 info.sampleRates.push_back( SAMPLE_RATES[k] );
860 if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
861 info.preferredSampleRate = SAMPLE_RATES[k];
866 // Sort and remove any redundant values
867 std::sort( info.sampleRates.begin(), info.sampleRates.end() );
868 info.sampleRates.erase( unique( info.sampleRates.begin(), info.sampleRates.end() ), info.sampleRates.end() );
870 if ( info.sampleRates.size() == 0 ) {
871 errorStream_ << "RtApiCore::probeDeviceInfo: No supported sample rates found for device (" << device << ").";
872 errorText_ = errorStream_.str();
873 error( RtAudioError::WARNING );
877 // CoreAudio always uses 32-bit floating point data for PCM streams.
878 // Thus, any other "physical" formats supported by the device are of
879 // no interest to the client.
880 info.nativeFormats = RTAUDIO_FLOAT32;
882 if ( info.outputChannels > 0 )
883 if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true;
884 if ( info.inputChannels > 0 )
885 if ( getDefaultInputDevice() == device ) info.isDefaultInput = true;
891 static OSStatus callbackHandler( AudioDeviceID inDevice,
892 const AudioTimeStamp* /*inNow*/,
893 const AudioBufferList* inInputData,
894 const AudioTimeStamp* /*inInputTime*/,
895 AudioBufferList* outOutputData,
896 const AudioTimeStamp* /*inOutputTime*/,
899 CallbackInfo *info = (CallbackInfo *) infoPointer;
901 RtApiCore *object = (RtApiCore *) info->object;
902 if ( object->callbackEvent( inDevice, inInputData, outOutputData ) == false )
903 return kAudioHardwareUnspecifiedError;
905 return kAudioHardwareNoError;
908 static OSStatus xrunListener( AudioObjectID /*inDevice*/,
910 const AudioObjectPropertyAddress properties[],
911 void* handlePointer )
913 CoreHandle *handle = (CoreHandle *) handlePointer;
914 for ( UInt32 i=0; i<nAddresses; i++ ) {
915 if ( properties[i].mSelector == kAudioDeviceProcessorOverload ) {
916 if ( properties[i].mScope == kAudioDevicePropertyScopeInput )
917 handle->xrun[1] = true;
919 handle->xrun[0] = true;
923 return kAudioHardwareNoError;
926 static OSStatus rateListener( AudioObjectID inDevice,
927 UInt32 /*nAddresses*/,
928 const AudioObjectPropertyAddress /*properties*/[],
931 Float64 *rate = (Float64 *) ratePointer;
932 UInt32 dataSize = sizeof( Float64 );
933 AudioObjectPropertyAddress property = { kAudioDevicePropertyNominalSampleRate,
934 kAudioObjectPropertyScopeGlobal,
935 kAudioObjectPropertyElementMaster };
936 AudioObjectGetPropertyData( inDevice, &property, 0, NULL, &dataSize, rate );
937 return kAudioHardwareNoError;
940 bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
941 unsigned int firstChannel, unsigned int sampleRate,
942 RtAudioFormat format, unsigned int *bufferSize,
943 RtAudio::StreamOptions *options )
946 unsigned int nDevices = getDeviceCount();
947 if ( nDevices == 0 ) {
948 // This should not happen because a check is made before this function is called.
949 errorText_ = "RtApiCore::probeDeviceOpen: no devices found!";
953 if ( device >= nDevices ) {
954 // This should not happen because a check is made before this function is called.
955 errorText_ = "RtApiCore::probeDeviceOpen: device ID is invalid!";
959 AudioDeviceID deviceList[ nDevices ];
960 UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;
961 AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
962 kAudioObjectPropertyScopeGlobal,
963 kAudioObjectPropertyElementMaster };
964 OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property,
965 0, NULL, &dataSize, (void *) &deviceList );
966 if ( result != noErr ) {
967 errorText_ = "RtApiCore::probeDeviceOpen: OS-X system error getting device IDs.";
971 AudioDeviceID id = deviceList[ device ];
973 // Setup for stream mode.
974 bool isInput = false;
975 if ( mode == INPUT ) {
977 property.mScope = kAudioDevicePropertyScopeInput;
980 property.mScope = kAudioDevicePropertyScopeOutput;
982 // Get the stream "configuration".
983 AudioBufferList *bufferList = nil;
985 property.mSelector = kAudioDevicePropertyStreamConfiguration;
986 result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
987 if ( result != noErr || dataSize == 0 ) {
988 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration info for device (" << device << ").";
989 errorText_ = errorStream_.str();
993 // Allocate the AudioBufferList.
994 bufferList = (AudioBufferList *) malloc( dataSize );
995 if ( bufferList == NULL ) {
996 errorText_ = "RtApiCore::probeDeviceOpen: memory error allocating AudioBufferList.";
1000 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
1001 if (result != noErr || dataSize == 0) {
1003 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration for device (" << device << ").";
1004 errorText_ = errorStream_.str();
1008 // Search for one or more streams that contain the desired number of
1009 // channels. CoreAudio devices can have an arbitrary number of
1010 // streams and each stream can have an arbitrary number of channels.
1011 // For each stream, a single buffer of interleaved samples is
1012 // provided. RtAudio prefers the use of one stream of interleaved
1013 // data or multiple consecutive single-channel streams. However, we
1014 // now support multiple consecutive multi-channel streams of
1015 // interleaved data as well.
1016 UInt32 iStream, offsetCounter = firstChannel;
1017 UInt32 nStreams = bufferList->mNumberBuffers;
1018 bool monoMode = false;
1019 bool foundStream = false;
1021 // First check that the device supports the requested number of
1023 UInt32 deviceChannels = 0;
1024 for ( iStream=0; iStream<nStreams; iStream++ )
1025 deviceChannels += bufferList->mBuffers[iStream].mNumberChannels;
1027 if ( deviceChannels < ( channels + firstChannel ) ) {
1029 errorStream_ << "RtApiCore::probeDeviceOpen: the device (" << device << ") does not support the requested channel count.";
1030 errorText_ = errorStream_.str();
1034 // Look for a single stream meeting our needs.
1035 UInt32 firstStream, streamCount = 1, streamChannels = 0, channelOffset = 0;
1036 for ( iStream=0; iStream<nStreams; iStream++ ) {
1037 streamChannels = bufferList->mBuffers[iStream].mNumberChannels;
1038 if ( streamChannels >= channels + offsetCounter ) {
1039 firstStream = iStream;
1040 channelOffset = offsetCounter;
1044 if ( streamChannels > offsetCounter ) break;
1045 offsetCounter -= streamChannels;
1048 // If we didn't find a single stream above, then we should be able
1049 // to meet the channel specification with multiple streams.
1050 if ( foundStream == false ) {
1052 offsetCounter = firstChannel;
1053 for ( iStream=0; iStream<nStreams; iStream++ ) {
1054 streamChannels = bufferList->mBuffers[iStream].mNumberChannels;
1055 if ( streamChannels > offsetCounter ) break;
1056 offsetCounter -= streamChannels;
1059 firstStream = iStream;
1060 channelOffset = offsetCounter;
1061 Int32 channelCounter = channels + offsetCounter - streamChannels;
1063 if ( streamChannels > 1 ) monoMode = false;
1064 while ( channelCounter > 0 ) {
1065 streamChannels = bufferList->mBuffers[++iStream].mNumberChannels;
1066 if ( streamChannels > 1 ) monoMode = false;
1067 channelCounter -= streamChannels;
1074 // Determine the buffer size.
1075 AudioValueRange bufferRange;
1076 dataSize = sizeof( AudioValueRange );
1077 property.mSelector = kAudioDevicePropertyBufferFrameSizeRange;
1078 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &bufferRange );
1080 if ( result != noErr ) {
1081 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting buffer size range for device (" << device << ").";
1082 errorText_ = errorStream_.str();
1086 if ( bufferRange.mMinimum > *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMinimum;
1087 else if ( bufferRange.mMaximum < *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMaximum;
1088 if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) *bufferSize = (unsigned long) bufferRange.mMinimum;
1090 // Set the buffer size. For multiple streams, I'm assuming we only
1091 // need to make this setting for the master channel.
1092 UInt32 theSize = (UInt32) *bufferSize;
1093 dataSize = sizeof( UInt32 );
1094 property.mSelector = kAudioDevicePropertyBufferFrameSize;
1095 result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &theSize );
1097 if ( result != noErr ) {
1098 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting the buffer size for device (" << device << ").";
1099 errorText_ = errorStream_.str();
1103 // If attempting to setup a duplex stream, the bufferSize parameter
1104 // MUST be the same in both directions!
1105 *bufferSize = theSize;
1106 if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {
1107 errorStream_ << "RtApiCore::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << device << ").";
1108 errorText_ = errorStream_.str();
1112 stream_.bufferSize = *bufferSize;
1113 stream_.nBuffers = 1;
1115 // Try to set "hog" mode ... it's not clear to me this is working.
1116 if ( options && options->flags & RTAUDIO_HOG_DEVICE ) {
1118 dataSize = sizeof( hog_pid );
1119 property.mSelector = kAudioDevicePropertyHogMode;
1120 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &hog_pid );
1121 if ( result != noErr ) {
1122 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting 'hog' state!";
1123 errorText_ = errorStream_.str();
1127 if ( hog_pid != getpid() ) {
1129 result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &hog_pid );
1130 if ( result != noErr ) {
1131 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting 'hog' state!";
1132 errorText_ = errorStream_.str();
1138 // Check and if necessary, change the sample rate for the device.
1139 Float64 nominalRate;
1140 dataSize = sizeof( Float64 );
1141 property.mSelector = kAudioDevicePropertyNominalSampleRate;
1142 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &nominalRate );
1143 if ( result != noErr ) {
1144 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting current sample rate.";
1145 errorText_ = errorStream_.str();
1149 // Only change the sample rate if off by more than 1 Hz.
1150 if ( fabs( nominalRate - (double)sampleRate ) > 1.0 ) {
1152 // Set a property listener for the sample rate change
1153 Float64 reportedRate = 0.0;
1154 AudioObjectPropertyAddress tmp = { kAudioDevicePropertyNominalSampleRate, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
1155 result = AudioObjectAddPropertyListener( id, &tmp, rateListener, (void *) &reportedRate );
1156 if ( result != noErr ) {
1157 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate property listener for device (" << device << ").";
1158 errorText_ = errorStream_.str();
1162 nominalRate = (Float64) sampleRate;
1163 result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &nominalRate );
1164 if ( result != noErr ) {
1165 AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate );
1166 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate for device (" << device << ").";
1167 errorText_ = errorStream_.str();
1171 // Now wait until the reported nominal rate is what we just set.
1172 UInt32 microCounter = 0;
1173 while ( reportedRate != nominalRate ) {
1174 microCounter += 5000;
1175 if ( microCounter > 5000000 ) break;
1179 // Remove the property listener.
1180 AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate );
1182 if ( microCounter > 5000000 ) {
1183 errorStream_ << "RtApiCore::probeDeviceOpen: timeout waiting for sample rate update for device (" << device << ").";
1184 errorText_ = errorStream_.str();
1189 // Now set the stream format for all streams. Also, check the
1190 // physical format of the device and change that if necessary.
1191 AudioStreamBasicDescription description;
1192 dataSize = sizeof( AudioStreamBasicDescription );
1193 property.mSelector = kAudioStreamPropertyVirtualFormat;
1194 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &description );
1195 if ( result != noErr ) {
1196 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream format for device (" << device << ").";
1197 errorText_ = errorStream_.str();
1201 // Set the sample rate and data format id. However, only make the
1202 // change if the sample rate is not within 1.0 of the desired
1203 // rate and the format is not linear pcm.
1204 bool updateFormat = false;
1205 if ( fabs( description.mSampleRate - (Float64)sampleRate ) > 1.0 ) {
1206 description.mSampleRate = (Float64) sampleRate;
1207 updateFormat = true;
1210 if ( description.mFormatID != kAudioFormatLinearPCM ) {
1211 description.mFormatID = kAudioFormatLinearPCM;
1212 updateFormat = true;
1215 if ( updateFormat ) {
1216 result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &description );
1217 if ( result != noErr ) {
1218 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate or data format for device (" << device << ").";
1219 errorText_ = errorStream_.str();
1224 // Now check the physical format.
1225 property.mSelector = kAudioStreamPropertyPhysicalFormat;
1226 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &description );
1227 if ( result != noErr ) {
1228 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream physical format for device (" << device << ").";
1229 errorText_ = errorStream_.str();
1233 //std::cout << "Current physical stream format:" << std::endl;
1234 //std::cout << " mBitsPerChan = " << description.mBitsPerChannel << std::endl;
1235 //std::cout << " aligned high = " << (description.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (description.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl;
1236 //std::cout << " bytesPerFrame = " << description.mBytesPerFrame << std::endl;
1237 //std::cout << " sample rate = " << description.mSampleRate << std::endl;
1239 if ( description.mFormatID != kAudioFormatLinearPCM || description.mBitsPerChannel < 16 ) {
1240 description.mFormatID = kAudioFormatLinearPCM;
1241 //description.mSampleRate = (Float64) sampleRate;
1242 AudioStreamBasicDescription testDescription = description;
1245 // We'll try higher bit rates first and then work our way down.
1246 std::vector< std::pair<UInt32, UInt32> > physicalFormats;
1247 formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsFloat) & ~kLinearPCMFormatFlagIsSignedInteger;
1248 physicalFormats.push_back( std::pair<Float32, UInt32>( 32, formatFlags ) );
1249 formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat;
1250 physicalFormats.push_back( std::pair<Float32, UInt32>( 32, formatFlags ) );
1251 physicalFormats.push_back( std::pair<Float32, UInt32>( 24, formatFlags ) ); // 24-bit packed
1252 formatFlags &= ~( kAudioFormatFlagIsPacked | kAudioFormatFlagIsAlignedHigh );
1253 physicalFormats.push_back( std::pair<Float32, UInt32>( 24.2, formatFlags ) ); // 24-bit in 4 bytes, aligned low
1254 formatFlags |= kAudioFormatFlagIsAlignedHigh;
1255 physicalFormats.push_back( std::pair<Float32, UInt32>( 24.4, formatFlags ) ); // 24-bit in 4 bytes, aligned high
1256 formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat;
1257 physicalFormats.push_back( std::pair<Float32, UInt32>( 16, formatFlags ) );
1258 physicalFormats.push_back( std::pair<Float32, UInt32>( 8, formatFlags ) );
1260 bool setPhysicalFormat = false;
1261 for( unsigned int i=0; i<physicalFormats.size(); i++ ) {
1262 testDescription = description;
1263 testDescription.mBitsPerChannel = (UInt32) physicalFormats[i].first;
1264 testDescription.mFormatFlags = physicalFormats[i].second;
1265 if ( (24 == (UInt32)physicalFormats[i].first) && ~( physicalFormats[i].second & kAudioFormatFlagIsPacked ) )
1266 testDescription.mBytesPerFrame = 4 * testDescription.mChannelsPerFrame;
1268 testDescription.mBytesPerFrame = testDescription.mBitsPerChannel/8 * testDescription.mChannelsPerFrame;
1269 testDescription.mBytesPerPacket = testDescription.mBytesPerFrame * testDescription.mFramesPerPacket;
1270 result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &testDescription );
1271 if ( result == noErr ) {
1272 setPhysicalFormat = true;
1273 //std::cout << "Updated physical stream format:" << std::endl;
1274 //std::cout << " mBitsPerChan = " << testDescription.mBitsPerChannel << std::endl;
1275 //std::cout << " aligned high = " << (testDescription.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (testDescription.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl;
1276 //std::cout << " bytesPerFrame = " << testDescription.mBytesPerFrame << std::endl;
1277 //std::cout << " sample rate = " << testDescription.mSampleRate << std::endl;
1282 if ( !setPhysicalFormat ) {
1283 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting physical data format for device (" << device << ").";
1284 errorText_ = errorStream_.str();
1287 } // done setting virtual/physical formats.
1289 // Get the stream / device latency.
1291 dataSize = sizeof( UInt32 );
1292 property.mSelector = kAudioDevicePropertyLatency;
1293 if ( AudioObjectHasProperty( id, &property ) == true ) {
1294 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &latency );
1295 if ( result == kAudioHardwareNoError ) stream_.latency[ mode ] = latency;
1297 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting device latency for device (" << device << ").";
1298 errorText_ = errorStream_.str();
1299 error( RtAudioError::WARNING );
1303 // Byte-swapping: According to AudioHardware.h, the stream data will
1304 // always be presented in native-endian format, so we should never
1305 // need to byte swap.
1306 stream_.doByteSwap[mode] = false;
1308 // From the CoreAudio documentation, PCM data must be supplied as
1310 stream_.userFormat = format;
1311 stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
1313 if ( streamCount == 1 )
1314 stream_.nDeviceChannels[mode] = description.mChannelsPerFrame;
1315 else // multiple streams
1316 stream_.nDeviceChannels[mode] = channels;
1317 stream_.nUserChannels[mode] = channels;
1318 stream_.channelOffset[mode] = channelOffset; // offset within a CoreAudio stream
1319 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
1320 else stream_.userInterleaved = true;
1321 stream_.deviceInterleaved[mode] = true;
1322 if ( monoMode == true ) stream_.deviceInterleaved[mode] = false;
1324 // Set flags for buffer conversion.
1325 stream_.doConvertBuffer[mode] = false;
1326 if ( stream_.userFormat != stream_.deviceFormat[mode] )
1327 stream_.doConvertBuffer[mode] = true;
1328 if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
1329 stream_.doConvertBuffer[mode] = true;
1330 if ( streamCount == 1 ) {
1331 if ( stream_.nUserChannels[mode] > 1 &&
1332 stream_.userInterleaved != stream_.deviceInterleaved[mode] )
1333 stream_.doConvertBuffer[mode] = true;
1335 else if ( monoMode && stream_.userInterleaved )
1336 stream_.doConvertBuffer[mode] = true;
1338 // Allocate our CoreHandle structure for the stream.
1339 CoreHandle *handle = 0;
1340 if ( stream_.apiHandle == 0 ) {
1342 handle = new CoreHandle;
1344 catch ( std::bad_alloc& ) {
1345 errorText_ = "RtApiCore::probeDeviceOpen: error allocating CoreHandle memory.";
1349 if ( pthread_cond_init( &handle->condition, NULL ) ) {
1350 errorText_ = "RtApiCore::probeDeviceOpen: error initializing pthread condition variable.";
1353 stream_.apiHandle = (void *) handle;
1356 handle = (CoreHandle *) stream_.apiHandle;
1357 handle->iStream[mode] = firstStream;
1358 handle->nStreams[mode] = streamCount;
1359 handle->id[mode] = id;
1361 // Allocate necessary internal buffers.
1362 unsigned long bufferBytes;
1363 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
1364 // stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
1365 stream_.userBuffer[mode] = (char *) malloc( bufferBytes * sizeof(char) );
1366 memset( stream_.userBuffer[mode], 0, bufferBytes * sizeof(char) );
1367 if ( stream_.userBuffer[mode] == NULL ) {
1368 errorText_ = "RtApiCore::probeDeviceOpen: error allocating user buffer memory.";
1372 // If possible, we will make use of the CoreAudio stream buffers as
1373 // "device buffers". However, we can't do this if using multiple
1375 if ( stream_.doConvertBuffer[mode] && handle->nStreams[mode] > 1 ) {
1377 bool makeBuffer = true;
1378 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
1379 if ( mode == INPUT ) {
1380 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
1381 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
1382 if ( bufferBytes <= bytesOut ) makeBuffer = false;
1387 bufferBytes *= *bufferSize;
1388 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
1389 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
1390 if ( stream_.deviceBuffer == NULL ) {
1391 errorText_ = "RtApiCore::probeDeviceOpen: error allocating device buffer memory.";
1397 stream_.sampleRate = sampleRate;
1398 stream_.device[mode] = device;
1399 stream_.state = STREAM_STOPPED;
1400 stream_.callbackInfo.object = (void *) this;
1402 // Setup the buffer conversion information structure.
1403 if ( stream_.doConvertBuffer[mode] ) {
1404 if ( streamCount > 1 ) setConvertInfo( mode, 0 );
1405 else setConvertInfo( mode, channelOffset );
1408 if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device )
1409 // Only one callback procedure per device.
1410 stream_.mode = DUPLEX;
1412 #if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
1413 result = AudioDeviceCreateIOProcID( id, callbackHandler, (void *) &stream_.callbackInfo, &handle->procId[mode] );
1415 // deprecated in favor of AudioDeviceCreateIOProcID()
1416 result = AudioDeviceAddIOProc( id, callbackHandler, (void *) &stream_.callbackInfo );
1418 if ( result != noErr ) {
1419 errorStream_ << "RtApiCore::probeDeviceOpen: system error setting callback for device (" << device << ").";
1420 errorText_ = errorStream_.str();
1423 if ( stream_.mode == OUTPUT && mode == INPUT )
1424 stream_.mode = DUPLEX;
1426 stream_.mode = mode;
1429 // Setup the device property listener for over/underload.
1430 property.mSelector = kAudioDeviceProcessorOverload;
1431 property.mScope = kAudioObjectPropertyScopeGlobal;
1432 result = AudioObjectAddPropertyListener( id, &property, xrunListener, (void *) handle );
1438 pthread_cond_destroy( &handle->condition );
1440 stream_.apiHandle = 0;
1443 for ( int i=0; i<2; i++ ) {
1444 if ( stream_.userBuffer[i] ) {
1445 free( stream_.userBuffer[i] );
1446 stream_.userBuffer[i] = 0;
1450 if ( stream_.deviceBuffer ) {
1451 free( stream_.deviceBuffer );
1452 stream_.deviceBuffer = 0;
1455 stream_.state = STREAM_CLOSED;
1459 void RtApiCore :: closeStream( void )
1461 if ( stream_.state == STREAM_CLOSED ) {
1462 errorText_ = "RtApiCore::closeStream(): no open stream to close!";
1463 error( RtAudioError::WARNING );
1467 CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
1468 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
1470 AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
1471 kAudioObjectPropertyScopeGlobal,
1472 kAudioObjectPropertyElementMaster };
1474 property.mSelector = kAudioDeviceProcessorOverload;
1475 property.mScope = kAudioObjectPropertyScopeGlobal;
1476 if (AudioObjectRemovePropertyListener( handle->id[0], &property, xrunListener, (void *) handle ) != noErr) {
1477 errorText_ = "RtApiCore::closeStream(): error removing property listener!";
1478 error( RtAudioError::WARNING );
1481 if ( stream_.state == STREAM_RUNNING )
1482 AudioDeviceStop( handle->id[0], callbackHandler );
1483 #if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
1484 AudioDeviceDestroyIOProcID( handle->id[0], handle->procId[0] );
1486 // deprecated in favor of AudioDeviceDestroyIOProcID()
1487 AudioDeviceRemoveIOProc( handle->id[0], callbackHandler );
1491 if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
1493 AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
1494 kAudioObjectPropertyScopeGlobal,
1495 kAudioObjectPropertyElementMaster };
1497 property.mSelector = kAudioDeviceProcessorOverload;
1498 property.mScope = kAudioObjectPropertyScopeGlobal;
1499 if (AudioObjectRemovePropertyListener( handle->id[1], &property, xrunListener, (void *) handle ) != noErr) {
1500 errorText_ = "RtApiCore::closeStream(): error removing property listener!";
1501 error( RtAudioError::WARNING );
1504 if ( stream_.state == STREAM_RUNNING )
1505 AudioDeviceStop( handle->id[1], callbackHandler );
1506 #if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
1507 AudioDeviceDestroyIOProcID( handle->id[1], handle->procId[1] );
1509 // deprecated in favor of AudioDeviceDestroyIOProcID()
1510 AudioDeviceRemoveIOProc( handle->id[1], callbackHandler );
1514 for ( int i=0; i<2; i++ ) {
1515 if ( stream_.userBuffer[i] ) {
1516 free( stream_.userBuffer[i] );
1517 stream_.userBuffer[i] = 0;
1521 if ( stream_.deviceBuffer ) {
1522 free( stream_.deviceBuffer );
1523 stream_.deviceBuffer = 0;
1526 // Destroy pthread condition variable.
1527 pthread_cond_destroy( &handle->condition );
1529 stream_.apiHandle = 0;
1531 stream_.mode = UNINITIALIZED;
1532 stream_.state = STREAM_CLOSED;
1535 void RtApiCore :: startStream( void )
1538 if ( stream_.state == STREAM_RUNNING ) {
1539 errorText_ = "RtApiCore::startStream(): the stream is already running!";
1540 error( RtAudioError::WARNING );
1544 OSStatus result = noErr;
1545 CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
1546 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
1548 result = AudioDeviceStart( handle->id[0], callbackHandler );
1549 if ( result != noErr ) {
1550 errorStream_ << "RtApiCore::startStream: system error (" << getErrorCode( result ) << ") starting callback procedure on device (" << stream_.device[0] << ").";
1551 errorText_ = errorStream_.str();
1556 if ( stream_.mode == INPUT ||
1557 ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
1559 result = AudioDeviceStart( handle->id[1], callbackHandler );
1560 if ( result != noErr ) {
1561 errorStream_ << "RtApiCore::startStream: system error starting input callback procedure on device (" << stream_.device[1] << ").";
1562 errorText_ = errorStream_.str();
1567 handle->drainCounter = 0;
1568 handle->internalDrain = false;
1569 stream_.state = STREAM_RUNNING;
1572 if ( result == noErr ) return;
1573 error( RtAudioError::SYSTEM_ERROR );
1576 void RtApiCore :: stopStream( void )
1579 if ( stream_.state == STREAM_STOPPED ) {
1580 errorText_ = "RtApiCore::stopStream(): the stream is already stopped!";
1581 error( RtAudioError::WARNING );
1585 OSStatus result = noErr;
1586 CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
1587 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
1589 if ( handle->drainCounter == 0 ) {
1590 handle->drainCounter = 2;
1591 pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
1594 result = AudioDeviceStop( handle->id[0], callbackHandler );
1595 if ( result != noErr ) {
1596 errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping callback procedure on device (" << stream_.device[0] << ").";
1597 errorText_ = errorStream_.str();
1602 if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
1604 result = AudioDeviceStop( handle->id[1], callbackHandler );
1605 if ( result != noErr ) {
1606 errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping input callback procedure on device (" << stream_.device[1] << ").";
1607 errorText_ = errorStream_.str();
1612 stream_.state = STREAM_STOPPED;
1615 if ( result == noErr ) return;
1616 error( RtAudioError::SYSTEM_ERROR );
1619 void RtApiCore :: abortStream( void )
1622 if ( stream_.state == STREAM_STOPPED ) {
1623 errorText_ = "RtApiCore::abortStream(): the stream is already stopped!";
1624 error( RtAudioError::WARNING );
1628 CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
1629 handle->drainCounter = 2;
1634 // This function will be called by a spawned thread when the user
1635 // callback function signals that the stream should be stopped or
1636 // aborted. It is better to handle it this way because the
1637 // callbackEvent() function probably should return before the AudioDeviceStop()
1638 // function is called.
1639 static void *coreStopStream( void *ptr )
1641 CallbackInfo *info = (CallbackInfo *) ptr;
1642 RtApiCore *object = (RtApiCore *) info->object;
1644 object->stopStream();
1645 pthread_exit( NULL );
1648 bool RtApiCore :: callbackEvent( AudioDeviceID deviceId,
1649 const AudioBufferList *inBufferList,
1650 const AudioBufferList *outBufferList )
1652 if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;
1653 if ( stream_.state == STREAM_CLOSED ) {
1654 errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";
1655 error( RtAudioError::WARNING );
1659 CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
1660 CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
1662 // Check if we were draining the stream and signal is finished.
1663 if ( handle->drainCounter > 3 ) {
1664 ThreadHandle threadId;
1666 stream_.state = STREAM_STOPPING;
1667 if ( handle->internalDrain == true )
1668 pthread_create( &threadId, NULL, coreStopStream, info );
1669 else // external call to stopStream()
1670 pthread_cond_signal( &handle->condition );
1674 AudioDeviceID outputDevice = handle->id[0];
1676 // Invoke user callback to get fresh output data UNLESS we are
1677 // draining stream or duplex mode AND the input/output devices are
1678 // different AND this function is called for the input device.
1679 if ( handle->drainCounter == 0 && ( stream_.mode != DUPLEX || deviceId == outputDevice ) ) {
1680 RtAudioCallback callback = (RtAudioCallback) info->callback;
1681 double streamTime = getStreamTime();
1682 RtAudioStreamStatus status = 0;
1683 if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
1684 status |= RTAUDIO_OUTPUT_UNDERFLOW;
1685 handle->xrun[0] = false;
1687 if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
1688 status |= RTAUDIO_INPUT_OVERFLOW;
1689 handle->xrun[1] = false;
1692 int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
1693 stream_.bufferSize, streamTime, status, info->userData );
1694 if ( cbReturnValue == 2 ) {
1695 stream_.state = STREAM_STOPPING;
1696 handle->drainCounter = 2;
1700 else if ( cbReturnValue == 1 ) {
1701 handle->drainCounter = 1;
1702 handle->internalDrain = true;
1706 if ( stream_.mode == OUTPUT || ( stream_.mode == DUPLEX && deviceId == outputDevice ) ) {
1708 if ( handle->drainCounter > 1 ) { // write zeros to the output stream
1710 if ( handle->nStreams[0] == 1 ) {
1711 memset( outBufferList->mBuffers[handle->iStream[0]].mData,
1713 outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );
1715 else { // fill multiple streams with zeros
1716 for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) {
1717 memset( outBufferList->mBuffers[handle->iStream[0]+i].mData,
1719 outBufferList->mBuffers[handle->iStream[0]+i].mDataByteSize );
1723 else if ( handle->nStreams[0] == 1 ) {
1724 if ( stream_.doConvertBuffer[0] ) { // convert directly to CoreAudio stream buffer
1725 convertBuffer( (char *) outBufferList->mBuffers[handle->iStream[0]].mData,
1726 stream_.userBuffer[0], stream_.convertInfo[0] );
1728 else { // copy from user buffer
1729 memcpy( outBufferList->mBuffers[handle->iStream[0]].mData,
1730 stream_.userBuffer[0],
1731 outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );
1734 else { // fill multiple streams
1735 Float32 *inBuffer = (Float32 *) stream_.userBuffer[0];
1736 if ( stream_.doConvertBuffer[0] ) {
1737 convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
1738 inBuffer = (Float32 *) stream_.deviceBuffer;
1741 if ( stream_.deviceInterleaved[0] == false ) { // mono mode
1742 UInt32 bufferBytes = outBufferList->mBuffers[handle->iStream[0]].mDataByteSize;
1743 for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
1744 memcpy( outBufferList->mBuffers[handle->iStream[0]+i].mData,
1745 (void *)&inBuffer[i*stream_.bufferSize], bufferBytes );
1748 else { // fill multiple multi-channel streams with interleaved data
1749 UInt32 streamChannels, channelsLeft, inJump, outJump, inOffset;
1752 bool inInterleaved = ( stream_.userInterleaved ) ? true : false;
1753 UInt32 inChannels = stream_.nUserChannels[0];
1754 if ( stream_.doConvertBuffer[0] ) {
1755 inInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode
1756 inChannels = stream_.nDeviceChannels[0];
1759 if ( inInterleaved ) inOffset = 1;
1760 else inOffset = stream_.bufferSize;
1762 channelsLeft = inChannels;
1763 for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) {
1765 out = (Float32 *) outBufferList->mBuffers[handle->iStream[0]+i].mData;
1766 streamChannels = outBufferList->mBuffers[handle->iStream[0]+i].mNumberChannels;
1769 // Account for possible channel offset in first stream
1770 if ( i == 0 && stream_.channelOffset[0] > 0 ) {
1771 streamChannels -= stream_.channelOffset[0];
1772 outJump = stream_.channelOffset[0];
1776 // Account for possible unfilled channels at end of the last stream
1777 if ( streamChannels > channelsLeft ) {
1778 outJump = streamChannels - channelsLeft;
1779 streamChannels = channelsLeft;
1782 // Determine input buffer offsets and skips
1783 if ( inInterleaved ) {
1784 inJump = inChannels;
1785 in += inChannels - channelsLeft;
1789 in += (inChannels - channelsLeft) * inOffset;
1792 for ( unsigned int i=0; i<stream_.bufferSize; i++ ) {
1793 for ( unsigned int j=0; j<streamChannels; j++ ) {
1794 *out++ = in[j*inOffset];
1799 channelsLeft -= streamChannels;
1805 // Don't bother draining input
1806 if ( handle->drainCounter ) {
1807 handle->drainCounter++;
1811 AudioDeviceID inputDevice;
1812 inputDevice = handle->id[1];
1813 if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && deviceId == inputDevice ) ) {
1815 if ( handle->nStreams[1] == 1 ) {
1816 if ( stream_.doConvertBuffer[1] ) { // convert directly from CoreAudio stream buffer
1817 convertBuffer( stream_.userBuffer[1],
1818 (char *) inBufferList->mBuffers[handle->iStream[1]].mData,
1819 stream_.convertInfo[1] );
1821 else { // copy to user buffer
1822 memcpy( stream_.userBuffer[1],
1823 inBufferList->mBuffers[handle->iStream[1]].mData,
1824 inBufferList->mBuffers[handle->iStream[1]].mDataByteSize );
1827 else { // read from multiple streams
1828 Float32 *outBuffer = (Float32 *) stream_.userBuffer[1];
1829 if ( stream_.doConvertBuffer[1] ) outBuffer = (Float32 *) stream_.deviceBuffer;
1831 if ( stream_.deviceInterleaved[1] == false ) { // mono mode
1832 UInt32 bufferBytes = inBufferList->mBuffers[handle->iStream[1]].mDataByteSize;
1833 for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
1834 memcpy( (void *)&outBuffer[i*stream_.bufferSize],
1835 inBufferList->mBuffers[handle->iStream[1]+i].mData, bufferBytes );
1838 else { // read from multiple multi-channel streams
1839 UInt32 streamChannels, channelsLeft, inJump, outJump, outOffset;
1842 bool outInterleaved = ( stream_.userInterleaved ) ? true : false;
1843 UInt32 outChannels = stream_.nUserChannels[1];
1844 if ( stream_.doConvertBuffer[1] ) {
1845 outInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode
1846 outChannels = stream_.nDeviceChannels[1];
1849 if ( outInterleaved ) outOffset = 1;
1850 else outOffset = stream_.bufferSize;
1852 channelsLeft = outChannels;
1853 for ( unsigned int i=0; i<handle->nStreams[1]; i++ ) {
1855 in = (Float32 *) inBufferList->mBuffers[handle->iStream[1]+i].mData;
1856 streamChannels = inBufferList->mBuffers[handle->iStream[1]+i].mNumberChannels;
1859 // Account for possible channel offset in first stream
1860 if ( i == 0 && stream_.channelOffset[1] > 0 ) {
1861 streamChannels -= stream_.channelOffset[1];
1862 inJump = stream_.channelOffset[1];
1866 // Account for possible unread channels at end of the last stream
1867 if ( streamChannels > channelsLeft ) {
1868 inJump = streamChannels - channelsLeft;
1869 streamChannels = channelsLeft;
1872 // Determine output buffer offsets and skips
1873 if ( outInterleaved ) {
1874 outJump = outChannels;
1875 out += outChannels - channelsLeft;
1879 out += (outChannels - channelsLeft) * outOffset;
1882 for ( unsigned int i=0; i<stream_.bufferSize; i++ ) {
1883 for ( unsigned int j=0; j<streamChannels; j++ ) {
1884 out[j*outOffset] = *in++;
1889 channelsLeft -= streamChannels;
1893 if ( stream_.doConvertBuffer[1] ) { // convert from our internal "device" buffer
1894 convertBuffer( stream_.userBuffer[1],
1895 stream_.deviceBuffer,
1896 stream_.convertInfo[1] );
1902 //MUTEX_UNLOCK( &stream_.mutex );
1904 RtApi::tickStreamTime();
1908 const char* RtApiCore :: getErrorCode( OSStatus code )
1912 case kAudioHardwareNotRunningError:
1913 return "kAudioHardwareNotRunningError";
1915 case kAudioHardwareUnspecifiedError:
1916 return "kAudioHardwareUnspecifiedError";
1918 case kAudioHardwareUnknownPropertyError:
1919 return "kAudioHardwareUnknownPropertyError";
1921 case kAudioHardwareBadPropertySizeError:
1922 return "kAudioHardwareBadPropertySizeError";
1924 case kAudioHardwareIllegalOperationError:
1925 return "kAudioHardwareIllegalOperationError";
1927 case kAudioHardwareBadObjectError:
1928 return "kAudioHardwareBadObjectError";
1930 case kAudioHardwareBadDeviceError:
1931 return "kAudioHardwareBadDeviceError";
1933 case kAudioHardwareBadStreamError:
1934 return "kAudioHardwareBadStreamError";
1936 case kAudioHardwareUnsupportedOperationError:
1937 return "kAudioHardwareUnsupportedOperationError";
1939 case kAudioDeviceUnsupportedFormatError:
1940 return "kAudioDeviceUnsupportedFormatError";
1942 case kAudioDevicePermissionsError:
1943 return "kAudioDevicePermissionsError";
1946 return "CoreAudio unknown error";
1950 //******************** End of __MACOSX_CORE__ *********************//
1953 #if defined(__UNIX_JACK__)
1955 // JACK is a low-latency audio server, originally written for the
1956 // GNU/Linux operating system and now also ported to OS-X. It can
1957 // connect a number of different applications to an audio device, as
1958 // well as allowing them to share audio between themselves.
1960 // When using JACK with RtAudio, "devices" refer to JACK clients that
1961 // have ports connected to the server. The JACK server is typically
1962 // started in a terminal as follows:
1964 // .jackd -d alsa -d hw:0
1966 // or through an interface program such as qjackctl. Many of the
1967 // parameters normally set for a stream are fixed by the JACK server
1968 // and can be specified when the JACK server is started. In
1971 // .jackd -d alsa -d hw:0 -r 44100 -p 512 -n 4
1973 // specifies a sample rate of 44100 Hz, a buffer size of 512 sample
1974 // frames, and number of buffers = 4. Once the server is running, it
1975 // is not possible to override these values. If the values are not
1976 // specified in the command-line, the JACK server uses default values.
1978 // The JACK server does not have to be running when an instance of
1979 // RtApiJack is created, though the function getDeviceCount() will
1980 // report 0 devices found until JACK has been started. When no
1981 // devices are available (i.e., the JACK server is not running), a
1982 // stream cannot be opened.
1984 #include <jack/jack.h>
1988 // A structure to hold various information related to the Jack API
1991 jack_client_t *client;
1992 jack_port_t **ports[2];
1993 std::string deviceName[2];
1995 pthread_cond_t condition;
1996 int drainCounter; // Tracks callback counts when draining
1997 bool internalDrain; // Indicates if stop is initiated from callback or not.
2000 :client(0), drainCounter(0), internalDrain(false) { ports[0] = 0; ports[1] = 0; xrun[0] = false; xrun[1] = false; }
2003 #if !defined(__RTAUDIO_DEBUG__)
2004 static void jackSilentError( const char * ) {};
2007 RtApiJack :: RtApiJack()
2008 :shouldAutoconnect_(true) {
2009 // Nothing to do here.
2010 #if !defined(__RTAUDIO_DEBUG__)
2011 // Turn off Jack's internal error reporting.
2012 jack_set_error_function( &jackSilentError );
2016 RtApiJack :: ~RtApiJack()
2018 if ( stream_.state != STREAM_CLOSED ) closeStream();
2021 unsigned int RtApiJack :: getDeviceCount( void )
2023 // See if we can become a jack client.
2024 jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption;
2025 jack_status_t *status = NULL;
2026 jack_client_t *client = jack_client_open( "RtApiJackCount", options, status );
2027 if ( client == 0 ) return 0;
2030 std::string port, previousPort;
2031 unsigned int nChannels = 0, nDevices = 0;
2032 ports = jack_get_ports( client, NULL, JACK_DEFAULT_AUDIO_TYPE, 0 );
2034 // Parse the port names up to the first colon (:).
2037 port = (char *) ports[ nChannels ];
2038 iColon = port.find(":");
2039 if ( iColon != std::string::npos ) {
2040 port = port.substr( 0, iColon + 1 );
2041 if ( port != previousPort ) {
2043 previousPort = port;
2046 } while ( ports[++nChannels] );
2050 jack_client_close( client );
2054 RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device )
2056 RtAudio::DeviceInfo info;
2057 info.probed = false;
2059 jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption
2060 jack_status_t *status = NULL;
2061 jack_client_t *client = jack_client_open( "RtApiJackInfo", options, status );
2062 if ( client == 0 ) {
2063 errorText_ = "RtApiJack::getDeviceInfo: Jack server not found or connection error!";
2064 error( RtAudioError::WARNING );
2069 std::string port, previousPort;
2070 unsigned int nPorts = 0, nDevices = 0;
2071 ports = jack_get_ports( client, NULL, JACK_DEFAULT_AUDIO_TYPE, 0 );
2073 // Parse the port names up to the first colon (:).
2076 port = (char *) ports[ nPorts ];
2077 iColon = port.find(":");
2078 if ( iColon != std::string::npos ) {
2079 port = port.substr( 0, iColon );
2080 if ( port != previousPort ) {
2081 if ( nDevices == device ) info.name = port;
2083 previousPort = port;
2086 } while ( ports[++nPorts] );
2090 if ( device >= nDevices ) {
2091 jack_client_close( client );
2092 errorText_ = "RtApiJack::getDeviceInfo: device ID is invalid!";
2093 error( RtAudioError::INVALID_USE );
2097 // Get the current jack server sample rate.
2098 info.sampleRates.clear();
2100 info.preferredSampleRate = jack_get_sample_rate( client );
2101 info.sampleRates.push_back( info.preferredSampleRate );
2103 // Count the available ports containing the client name as device
2104 // channels. Jack "input ports" equal RtAudio output channels.
2105 unsigned int nChannels = 0;
2106 ports = jack_get_ports( client, info.name.c_str(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput );
2108 while ( ports[ nChannels ] ) nChannels++;
2110 info.outputChannels = nChannels;
2113 // Jack "output ports" equal RtAudio input channels.
2115 ports = jack_get_ports( client, info.name.c_str(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput );
2117 while ( ports[ nChannels ] ) nChannels++;
2119 info.inputChannels = nChannels;
2122 if ( info.outputChannels == 0 && info.inputChannels == 0 ) {
2123 jack_client_close(client);
2124 errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!";
2125 error( RtAudioError::WARNING );
2129 // If device opens for both playback and capture, we determine the channels.
2130 if ( info.outputChannels > 0 && info.inputChannels > 0 )
2131 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
2133 // Jack always uses 32-bit floats.
2134 info.nativeFormats = RTAUDIO_FLOAT32;
2136 // Jack doesn't provide default devices so we'll use the first available one.
2137 if ( device == 0 && info.outputChannels > 0 )
2138 info.isDefaultOutput = true;
2139 if ( device == 0 && info.inputChannels > 0 )
2140 info.isDefaultInput = true;
2142 jack_client_close(client);
2147 static int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer )
2149 CallbackInfo *info = (CallbackInfo *) infoPointer;
2151 RtApiJack *object = (RtApiJack *) info->object;
2152 if ( object->callbackEvent( (unsigned long) nframes ) == false ) return 1;
2157 // This function will be called by a spawned thread when the Jack
2158 // server signals that it is shutting down. It is necessary to handle
2159 // it this way because the jackShutdown() function must return before
2160 // the jack_deactivate() function (in closeStream()) will return.
2161 static void *jackCloseStream( void *ptr )
2163 CallbackInfo *info = (CallbackInfo *) ptr;
2164 RtApiJack *object = (RtApiJack *) info->object;
2166 object->closeStream();
2168 pthread_exit( NULL );
2170 static void jackShutdown( void *infoPointer )
2172 CallbackInfo *info = (CallbackInfo *) infoPointer;
2173 RtApiJack *object = (RtApiJack *) info->object;
2175 // Check current stream state. If stopped, then we'll assume this
2176 // was called as a result of a call to RtApiJack::stopStream (the
2177 // deactivation of a client handle causes this function to be called).
2178 // If not, we'll assume the Jack server is shutting down or some
2179 // other problem occurred and we should close the stream.
2180 if ( object->isStreamRunning() == false ) return;
2182 ThreadHandle threadId;
2183 pthread_create( &threadId, NULL, jackCloseStream, info );
2184 std::cerr << "\nRtApiJack: the Jack server is shutting down this client ... stream stopped and closed!!\n" << std::endl;
2187 static int jackXrun( void *infoPointer )
2189 JackHandle *handle = *((JackHandle **) infoPointer);
2191 if ( handle->ports[0] ) handle->xrun[0] = true;
2192 if ( handle->ports[1] ) handle->xrun[1] = true;
2197 bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
2198 unsigned int firstChannel, unsigned int sampleRate,
2199 RtAudioFormat format, unsigned int *bufferSize,
2200 RtAudio::StreamOptions *options )
2202 JackHandle *handle = (JackHandle *) stream_.apiHandle;
2204 // Look for jack server and try to become a client (only do once per stream).
2205 jack_client_t *client = 0;
2206 if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) {
2207 jack_options_t jackoptions = (jack_options_t) ( JackNoStartServer ); //JackNullOption;
2208 jack_status_t *status = NULL;
2209 if ( options && !options->streamName.empty() )
2210 client = jack_client_open( options->streamName.c_str(), jackoptions, status );
2212 client = jack_client_open( "RtApiJack", jackoptions, status );
2213 if ( client == 0 ) {
2214 errorText_ = "RtApiJack::probeDeviceOpen: Jack server not found or connection error!";
2215 error( RtAudioError::WARNING );
2220 // The handle must have been created on an earlier pass.
2221 client = handle->client;
2225 std::string port, previousPort, deviceName;
2226 unsigned int nPorts = 0, nDevices = 0;
2227 ports = jack_get_ports( client, NULL, JACK_DEFAULT_AUDIO_TYPE, 0 );
2229 // Parse the port names up to the first colon (:).
2232 port = (char *) ports[ nPorts ];
2233 iColon = port.find(":");
2234 if ( iColon != std::string::npos ) {
2235 port = port.substr( 0, iColon );
2236 if ( port != previousPort ) {
2237 if ( nDevices == device ) deviceName = port;
2239 previousPort = port;
2242 } while ( ports[++nPorts] );
2246 if ( device >= nDevices ) {
2247 errorText_ = "RtApiJack::probeDeviceOpen: device ID is invalid!";
2251 unsigned long flag = JackPortIsInput;
2252 if ( mode == INPUT ) flag = JackPortIsOutput;
2254 if ( ! (options && (options->flags & RTAUDIO_JACK_DONT_CONNECT)) ) {
2255 // Count the available ports containing the client name as device
2256 // channels. Jack "input ports" equal RtAudio output channels.
2257 unsigned int nChannels = 0;
2258 ports = jack_get_ports( client, deviceName.c_str(), JACK_DEFAULT_AUDIO_TYPE, flag );
2260 while ( ports[ nChannels ] ) nChannels++;
2263 // Compare the jack ports for specified client to the requested number of channels.
2264 if ( nChannels < (channels + firstChannel) ) {
2265 errorStream_ << "RtApiJack::probeDeviceOpen: requested number of channels (" << channels << ") + offset (" << firstChannel << ") not found for specified device (" << device << ":" << deviceName << ").";
2266 errorText_ = errorStream_.str();
2271 // Check the jack server sample rate.
2272 unsigned int jackRate = jack_get_sample_rate( client );
2273 if ( sampleRate != jackRate ) {
2274 jack_client_close( client );
2275 errorStream_ << "RtApiJack::probeDeviceOpen: the requested sample rate (" << sampleRate << ") is different than the JACK server rate (" << jackRate << ").";
2276 errorText_ = errorStream_.str();
2279 stream_.sampleRate = jackRate;
2281 // Get the latency of the JACK port.
2282 ports = jack_get_ports( client, deviceName.c_str(), JACK_DEFAULT_AUDIO_TYPE, flag );
2283 if ( ports[ firstChannel ] ) {
2285 jack_latency_callback_mode_t cbmode = (mode == INPUT ? JackCaptureLatency : JackPlaybackLatency);
2286 // the range (usually the min and max are equal)
2287 jack_latency_range_t latrange; latrange.min = latrange.max = 0;
2288 // get the latency range
2289 jack_port_get_latency_range( jack_port_by_name( client, ports[firstChannel] ), cbmode, &latrange );
2290 // be optimistic, use the min!
2291 stream_.latency[mode] = latrange.min;
2292 //stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) );
2296 // The jack server always uses 32-bit floating-point data.
2297 stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
2298 stream_.userFormat = format;
2300 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
2301 else stream_.userInterleaved = true;
2303 // Jack always uses non-interleaved buffers.
2304 stream_.deviceInterleaved[mode] = false;
2306 // Jack always provides host byte-ordered data.
2307 stream_.doByteSwap[mode] = false;
2309 // Get the buffer size. The buffer size and number of buffers
2310 // (periods) is set when the jack server is started.
2311 stream_.bufferSize = (int) jack_get_buffer_size( client );
2312 *bufferSize = stream_.bufferSize;
2314 stream_.nDeviceChannels[mode] = channels;
2315 stream_.nUserChannels[mode] = channels;
2317 // Set flags for buffer conversion.
2318 stream_.doConvertBuffer[mode] = false;
2319 if ( stream_.userFormat != stream_.deviceFormat[mode] )
2320 stream_.doConvertBuffer[mode] = true;
2321 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
2322 stream_.nUserChannels[mode] > 1 )
2323 stream_.doConvertBuffer[mode] = true;
2325 // Allocate our JackHandle structure for the stream.
2326 if ( handle == 0 ) {
2328 handle = new JackHandle;
2330 catch ( std::bad_alloc& ) {
2331 errorText_ = "RtApiJack::probeDeviceOpen: error allocating JackHandle memory.";
2335 if ( pthread_cond_init(&handle->condition, NULL) ) {
2336 errorText_ = "RtApiJack::probeDeviceOpen: error initializing pthread condition variable.";
2339 stream_.apiHandle = (void *) handle;
2340 handle->client = client;
2342 handle->deviceName[mode] = deviceName;
2344 // Allocate necessary internal buffers.
2345 unsigned long bufferBytes;
2346 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
2347 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
2348 if ( stream_.userBuffer[mode] == NULL ) {
2349 errorText_ = "RtApiJack::probeDeviceOpen: error allocating user buffer memory.";
2353 if ( stream_.doConvertBuffer[mode] ) {
2355 bool makeBuffer = true;
2356 if ( mode == OUTPUT )
2357 bufferBytes = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
2358 else { // mode == INPUT
2359 bufferBytes = stream_.nDeviceChannels[1] * formatBytes( stream_.deviceFormat[1] );
2360 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
2361 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]);
2362 if ( bufferBytes < bytesOut ) makeBuffer = false;
2367 bufferBytes *= *bufferSize;
2368 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
2369 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
2370 if ( stream_.deviceBuffer == NULL ) {
2371 errorText_ = "RtApiJack::probeDeviceOpen: error allocating device buffer memory.";
2377 // Allocate memory for the Jack ports (channels) identifiers.
2378 handle->ports[mode] = (jack_port_t **) malloc ( sizeof (jack_port_t *) * channels );
2379 if ( handle->ports[mode] == NULL ) {
2380 errorText_ = "RtApiJack::probeDeviceOpen: error allocating port memory.";
2384 stream_.device[mode] = device;
2385 stream_.channelOffset[mode] = firstChannel;
2386 stream_.state = STREAM_STOPPED;
2387 stream_.callbackInfo.object = (void *) this;
2389 if ( stream_.mode == OUTPUT && mode == INPUT )
2390 // We had already set up the stream for output.
2391 stream_.mode = DUPLEX;
2393 stream_.mode = mode;
2394 jack_set_process_callback( handle->client, jackCallbackHandler, (void *) &stream_.callbackInfo );
2395 jack_set_xrun_callback( handle->client, jackXrun, (void *) &stream_.apiHandle );
2396 jack_on_shutdown( handle->client, jackShutdown, (void *) &stream_.callbackInfo );
2399 // Register our ports.
2401 if ( mode == OUTPUT ) {
2402 for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
2403 snprintf( label, 64, "outport %d", i );
2404 handle->ports[0][i] = jack_port_register( handle->client, (const char *)label,
2405 JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0 );
2409 for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
2410 snprintf( label, 64, "inport %d", i );
2411 handle->ports[1][i] = jack_port_register( handle->client, (const char *)label,
2412 JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0 );
2416 // Setup the buffer conversion information structure. We don't use
2417 // buffers to do channel offsets, so we override that parameter
2419 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );
2421 if ( options && options->flags & RTAUDIO_JACK_DONT_CONNECT ) shouldAutoconnect_ = false;
2427 pthread_cond_destroy( &handle->condition );
2428 jack_client_close( handle->client );
2430 if ( handle->ports[0] ) free( handle->ports[0] );
2431 if ( handle->ports[1] ) free( handle->ports[1] );
2434 stream_.apiHandle = 0;
2437 for ( int i=0; i<2; i++ ) {
2438 if ( stream_.userBuffer[i] ) {
2439 free( stream_.userBuffer[i] );
2440 stream_.userBuffer[i] = 0;
2444 if ( stream_.deviceBuffer ) {
2445 free( stream_.deviceBuffer );
2446 stream_.deviceBuffer = 0;
2452 void RtApiJack :: closeStream( void )
2454 if ( stream_.state == STREAM_CLOSED ) {
2455 errorText_ = "RtApiJack::closeStream(): no open stream to close!";
2456 error( RtAudioError::WARNING );
2460 JackHandle *handle = (JackHandle *) stream_.apiHandle;
2463 if ( stream_.state == STREAM_RUNNING )
2464 jack_deactivate( handle->client );
2466 jack_client_close( handle->client );
2470 if ( handle->ports[0] ) free( handle->ports[0] );
2471 if ( handle->ports[1] ) free( handle->ports[1] );
2472 pthread_cond_destroy( &handle->condition );
2474 stream_.apiHandle = 0;
2477 for ( int i=0; i<2; i++ ) {
2478 if ( stream_.userBuffer[i] ) {
2479 free( stream_.userBuffer[i] );
2480 stream_.userBuffer[i] = 0;
2484 if ( stream_.deviceBuffer ) {
2485 free( stream_.deviceBuffer );
2486 stream_.deviceBuffer = 0;
2489 stream_.mode = UNINITIALIZED;
2490 stream_.state = STREAM_CLOSED;
2493 void RtApiJack :: startStream( void )
2496 if ( stream_.state == STREAM_RUNNING ) {
2497 errorText_ = "RtApiJack::startStream(): the stream is already running!";
2498 error( RtAudioError::WARNING );
2502 JackHandle *handle = (JackHandle *) stream_.apiHandle;
2503 int result = jack_activate( handle->client );
2505 errorText_ = "RtApiJack::startStream(): unable to activate JACK client!";
2511 // Get the list of available ports.
2512 if ( shouldAutoconnect_ && (stream_.mode == OUTPUT || stream_.mode == DUPLEX) ) {
2514 ports = jack_get_ports( handle->client, handle->deviceName[0].c_str(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput);
2515 if ( ports == NULL) {
2516 errorText_ = "RtApiJack::startStream(): error determining available JACK input ports!";
2520 // Now make the port connections. Since RtAudio wasn't designed to
2521 // allow the user to select particular channels of a device, we'll
2522 // just open the first "nChannels" ports with offset.
2523 for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
2525 if ( ports[ stream_.channelOffset[0] + i ] )
2526 result = jack_connect( handle->client, jack_port_name( handle->ports[0][i] ), ports[ stream_.channelOffset[0] + i ] );
2529 errorText_ = "RtApiJack::startStream(): error connecting output ports!";
2536 if ( shouldAutoconnect_ && (stream_.mode == INPUT || stream_.mode == DUPLEX) ) {
2538 ports = jack_get_ports( handle->client, handle->deviceName[1].c_str(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput );
2539 if ( ports == NULL) {
2540 errorText_ = "RtApiJack::startStream(): error determining available JACK output ports!";
2544 // Now make the port connections. See note above.
2545 for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
2547 if ( ports[ stream_.channelOffset[1] + i ] )
2548 result = jack_connect( handle->client, ports[ stream_.channelOffset[1] + i ], jack_port_name( handle->ports[1][i] ) );
2551 errorText_ = "RtApiJack::startStream(): error connecting input ports!";
2558 handle->drainCounter = 0;
2559 handle->internalDrain = false;
2560 stream_.state = STREAM_RUNNING;
2563 if ( result == 0 ) return;
2564 error( RtAudioError::SYSTEM_ERROR );
2567 void RtApiJack :: stopStream( void )
2570 if ( stream_.state == STREAM_STOPPED ) {
2571 errorText_ = "RtApiJack::stopStream(): the stream is already stopped!";
2572 error( RtAudioError::WARNING );
2576 JackHandle *handle = (JackHandle *) stream_.apiHandle;
2577 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
2579 if ( handle->drainCounter == 0 ) {
2580 handle->drainCounter = 2;
2581 pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
2585 jack_deactivate( handle->client );
2586 stream_.state = STREAM_STOPPED;
2589 void RtApiJack :: abortStream( void )
2592 if ( stream_.state == STREAM_STOPPED ) {
2593 errorText_ = "RtApiJack::abortStream(): the stream is already stopped!";
2594 error( RtAudioError::WARNING );
2598 JackHandle *handle = (JackHandle *) stream_.apiHandle;
2599 handle->drainCounter = 2;
2604 // This function will be called by a spawned thread when the user
2605 // callback function signals that the stream should be stopped or
2606 // aborted. It is necessary to handle it this way because the
2607 // callbackEvent() function must return before the jack_deactivate()
2608 // function will return.
2609 static void *jackStopStream( void *ptr )
2611 CallbackInfo *info = (CallbackInfo *) ptr;
2612 RtApiJack *object = (RtApiJack *) info->object;
2614 object->stopStream();
2615 pthread_exit( NULL );
2618 bool RtApiJack :: callbackEvent( unsigned long nframes )
2620 if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;
2621 if ( stream_.state == STREAM_CLOSED ) {
2622 errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";
2623 error( RtAudioError::WARNING );
2626 if ( stream_.bufferSize != nframes ) {
2627 errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!";
2628 error( RtAudioError::WARNING );
2632 CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
2633 JackHandle *handle = (JackHandle *) stream_.apiHandle;
2635 // Check if we were draining the stream and signal is finished.
2636 if ( handle->drainCounter > 3 ) {
2637 ThreadHandle threadId;
2639 stream_.state = STREAM_STOPPING;
2640 if ( handle->internalDrain == true )
2641 pthread_create( &threadId, NULL, jackStopStream, info );
2643 pthread_cond_signal( &handle->condition );
2647 // Invoke user callback first, to get fresh output data.
2648 if ( handle->drainCounter == 0 ) {
2649 RtAudioCallback callback = (RtAudioCallback) info->callback;
2650 double streamTime = getStreamTime();
2651 RtAudioStreamStatus status = 0;
2652 if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
2653 status |= RTAUDIO_OUTPUT_UNDERFLOW;
2654 handle->xrun[0] = false;
2656 if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
2657 status |= RTAUDIO_INPUT_OVERFLOW;
2658 handle->xrun[1] = false;
2660 int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
2661 stream_.bufferSize, streamTime, status, info->userData );
2662 if ( cbReturnValue == 2 ) {
2663 stream_.state = STREAM_STOPPING;
2664 handle->drainCounter = 2;
2666 pthread_create( &id, NULL, jackStopStream, info );
2669 else if ( cbReturnValue == 1 ) {
2670 handle->drainCounter = 1;
2671 handle->internalDrain = true;
2675 jack_default_audio_sample_t *jackbuffer;
2676 unsigned long bufferBytes = nframes * sizeof( jack_default_audio_sample_t );
2677 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
2679 if ( handle->drainCounter > 1 ) { // write zeros to the output stream
2681 for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
2682 jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
2683 memset( jackbuffer, 0, bufferBytes );
2687 else if ( stream_.doConvertBuffer[0] ) {
2689 convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
2691 for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
2692 jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
2693 memcpy( jackbuffer, &stream_.deviceBuffer[i*bufferBytes], bufferBytes );
2696 else { // no buffer conversion
2697 for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
2698 jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
2699 memcpy( jackbuffer, &stream_.userBuffer[0][i*bufferBytes], bufferBytes );
2704 // Don't bother draining input
2705 if ( handle->drainCounter ) {
2706 handle->drainCounter++;
2710 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
2712 if ( stream_.doConvertBuffer[1] ) {
2713 for ( unsigned int i=0; i<stream_.nDeviceChannels[1]; i++ ) {
2714 jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );
2715 memcpy( &stream_.deviceBuffer[i*bufferBytes], jackbuffer, bufferBytes );
2717 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
2719 else { // no buffer conversion
2720 for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
2721 jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );
2722 memcpy( &stream_.userBuffer[1][i*bufferBytes], jackbuffer, bufferBytes );
2728 RtApi::tickStreamTime();
2731 //******************** End of __UNIX_JACK__ *********************//
2734 #if defined(__WINDOWS_ASIO__) // ASIO API on Windows
2736 // The ASIO API is designed around a callback scheme, so this
2737 // implementation is similar to that used for OS-X CoreAudio and Linux
2738 // Jack. The primary constraint with ASIO is that it only allows
2739 // access to a single driver at a time. Thus, it is not possible to
2740 // have more than one simultaneous RtAudio stream.
2742 // This implementation also requires a number of external ASIO files
2743 // and a few global variables. The ASIO callback scheme does not
2744 // allow for the passing of user data, so we must create a global
2745 // pointer to our callbackInfo structure.
2747 // On unix systems, we make use of a pthread condition variable.
2748 // Since there is no equivalent in Windows, I hacked something based
2749 // on information found in
2750 // http://www.cs.wustl.edu/~schmidt/win32-cv-1.html.
2752 #include "asiosys.h"
2754 #include "iasiothiscallresolver.h"
2755 #include "asiodrivers.h"
2758 static AsioDrivers drivers;
2759 static ASIOCallbacks asioCallbacks;
2760 static ASIODriverInfo driverInfo;
2761 static CallbackInfo *asioCallbackInfo;
2762 static bool asioXRun;
2765 int drainCounter; // Tracks callback counts when draining
2766 bool internalDrain; // Indicates if stop is initiated from callback or not.
2767 ASIOBufferInfo *bufferInfos;
2771 :drainCounter(0), internalDrain(false), bufferInfos(0) {}
2774 // Function declarations (definitions at end of section)
2775 static const char* getAsioErrorString( ASIOError result );
2776 static void sampleRateChanged( ASIOSampleRate sRate );
2777 static long asioMessages( long selector, long value, void* message, double* opt );
2779 RtApiAsio :: RtApiAsio()
2781 // ASIO cannot run on a multi-threaded appartment. You can call
2782 // CoInitialize beforehand, but it must be for appartment threading
2783 // (in which case, CoInitilialize will return S_FALSE here).
2784 coInitialized_ = false;
2785 HRESULT hr = CoInitialize( NULL );
2787 errorText_ = "RtApiAsio::ASIO requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)";
2788 error( RtAudioError::WARNING );
2790 coInitialized_ = true;
2792 drivers.removeCurrentDriver();
2793 driverInfo.asioVersion = 2;
2795 // See note in DirectSound implementation about GetDesktopWindow().
2796 driverInfo.sysRef = GetForegroundWindow();
2799 RtApiAsio :: ~RtApiAsio()
2801 if ( stream_.state != STREAM_CLOSED ) closeStream();
2802 if ( coInitialized_ ) CoUninitialize();
2805 unsigned int RtApiAsio :: getDeviceCount( void )
2807 return (unsigned int) drivers.asioGetNumDev();
2810 RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device )
2812 RtAudio::DeviceInfo info;
2813 info.probed = false;
2816 unsigned int nDevices = getDeviceCount();
2817 if ( nDevices == 0 ) {
2818 errorText_ = "RtApiAsio::getDeviceInfo: no devices found!";
2819 error( RtAudioError::INVALID_USE );
2823 if ( device >= nDevices ) {
2824 errorText_ = "RtApiAsio::getDeviceInfo: device ID is invalid!";
2825 error( RtAudioError::INVALID_USE );
2829 // If a stream is already open, we cannot probe other devices. Thus, use the saved results.
2830 if ( stream_.state != STREAM_CLOSED ) {
2831 if ( device >= devices_.size() ) {
2832 errorText_ = "RtApiAsio::getDeviceInfo: device ID was not present before stream was opened.";
2833 error( RtAudioError::WARNING );
2836 return devices_[ device ];
2839 char driverName[32];
2840 ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 );
2841 if ( result != ASE_OK ) {
2842 errorStream_ << "RtApiAsio::getDeviceInfo: unable to get driver name (" << getAsioErrorString( result ) << ").";
2843 errorText_ = errorStream_.str();
2844 error( RtAudioError::WARNING );
2848 info.name = driverName;
2850 if ( !drivers.loadDriver( driverName ) ) {
2851 errorStream_ << "RtApiAsio::getDeviceInfo: unable to load driver (" << driverName << ").";
2852 errorText_ = errorStream_.str();
2853 error( RtAudioError::WARNING );
2857 result = ASIOInit( &driverInfo );
2858 if ( result != ASE_OK ) {
2859 errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
2860 errorText_ = errorStream_.str();
2861 error( RtAudioError::WARNING );
2865 // Determine the device channel information.
2866 long inputChannels, outputChannels;
2867 result = ASIOGetChannels( &inputChannels, &outputChannels );
2868 if ( result != ASE_OK ) {
2869 drivers.removeCurrentDriver();
2870 errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
2871 errorText_ = errorStream_.str();
2872 error( RtAudioError::WARNING );
2876 info.outputChannels = outputChannels;
2877 info.inputChannels = inputChannels;
2878 if ( info.outputChannels > 0 && info.inputChannels > 0 )
2879 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
2881 // Determine the supported sample rates.
2882 info.sampleRates.clear();
2883 for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {
2884 result = ASIOCanSampleRate( (ASIOSampleRate) SAMPLE_RATES[i] );
2885 if ( result == ASE_OK ) {
2886 info.sampleRates.push_back( SAMPLE_RATES[i] );
2888 if ( !info.preferredSampleRate || ( SAMPLE_RATES[i] <= 48000 && SAMPLE_RATES[i] > info.preferredSampleRate ) )
2889 info.preferredSampleRate = SAMPLE_RATES[i];
2893 // Determine supported data types ... just check first channel and assume rest are the same.
2894 ASIOChannelInfo channelInfo;
2895 channelInfo.channel = 0;
2896 channelInfo.isInput = true;
2897 if ( info.inputChannels <= 0 ) channelInfo.isInput = false;
2898 result = ASIOGetChannelInfo( &channelInfo );
2899 if ( result != ASE_OK ) {
2900 drivers.removeCurrentDriver();
2901 errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting driver channel info (" << driverName << ").";
2902 errorText_ = errorStream_.str();
2903 error( RtAudioError::WARNING );
2907 info.nativeFormats = 0;
2908 if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB )
2909 info.nativeFormats |= RTAUDIO_SINT16;
2910 else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB )
2911 info.nativeFormats |= RTAUDIO_SINT32;
2912 else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB )
2913 info.nativeFormats |= RTAUDIO_FLOAT32;
2914 else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB )
2915 info.nativeFormats |= RTAUDIO_FLOAT64;
2916 else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB )
2917 info.nativeFormats |= RTAUDIO_SINT24;
2919 if ( info.outputChannels > 0 )
2920 if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true;
2921 if ( info.inputChannels > 0 )
2922 if ( getDefaultInputDevice() == device ) info.isDefaultInput = true;
2925 drivers.removeCurrentDriver();
2929 static void bufferSwitch( long index, ASIOBool /*processNow*/ )
2931 RtApiAsio *object = (RtApiAsio *) asioCallbackInfo->object;
2932 object->callbackEvent( index );
2935 void RtApiAsio :: saveDeviceInfo( void )
2939 unsigned int nDevices = getDeviceCount();
2940 devices_.resize( nDevices );
2941 for ( unsigned int i=0; i<nDevices; i++ )
2942 devices_[i] = getDeviceInfo( i );
2945 bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
2946 unsigned int firstChannel, unsigned int sampleRate,
2947 RtAudioFormat format, unsigned int *bufferSize,
2948 RtAudio::StreamOptions *options )
2949 {////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
2951 bool isDuplexInput = mode == INPUT && stream_.mode == OUTPUT;
2953 // For ASIO, a duplex stream MUST use the same driver.
2954 if ( isDuplexInput && stream_.device[0] != device ) {
2955 errorText_ = "RtApiAsio::probeDeviceOpen: an ASIO duplex stream must use the same device for input and output!";
2959 char driverName[32];
2960 ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 );
2961 if ( result != ASE_OK ) {
2962 errorStream_ << "RtApiAsio::probeDeviceOpen: unable to get driver name (" << getAsioErrorString( result ) << ").";
2963 errorText_ = errorStream_.str();
2967 // Only load the driver once for duplex stream.
2968 if ( !isDuplexInput ) {
2969 // The getDeviceInfo() function will not work when a stream is open
2970 // because ASIO does not allow multiple devices to run at the same
2971 // time. Thus, we'll probe the system before opening a stream and
2972 // save the results for use by getDeviceInfo().
2973 this->saveDeviceInfo();
2975 if ( !drivers.loadDriver( driverName ) ) {
2976 errorStream_ << "RtApiAsio::probeDeviceOpen: unable to load driver (" << driverName << ").";
2977 errorText_ = errorStream_.str();
2981 result = ASIOInit( &driverInfo );
2982 if ( result != ASE_OK ) {
2983 errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
2984 errorText_ = errorStream_.str();
2989 // keep them before any "goto error", they are used for error cleanup + goto device boundary checks
2990 bool buffersAllocated = false;
2991 AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
2992 unsigned int nChannels;
2995 // Check the device channel count.
2996 long inputChannels, outputChannels;
2997 result = ASIOGetChannels( &inputChannels, &outputChannels );
2998 if ( result != ASE_OK ) {
2999 errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
3000 errorText_ = errorStream_.str();
3004 if ( ( mode == OUTPUT && (channels+firstChannel) > (unsigned int) outputChannels) ||
3005 ( mode == INPUT && (channels+firstChannel) > (unsigned int) inputChannels) ) {
3006 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested channel count (" << channels << ") + offset (" << firstChannel << ").";
3007 errorText_ = errorStream_.str();
3010 stream_.nDeviceChannels[mode] = channels;
3011 stream_.nUserChannels[mode] = channels;
3012 stream_.channelOffset[mode] = firstChannel;
3014 // Verify the sample rate is supported.
3015 result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate );
3016 if ( result != ASE_OK ) {
3017 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested sample rate (" << sampleRate << ").";
3018 errorText_ = errorStream_.str();
3022 // Get the current sample rate
3023 ASIOSampleRate currentRate;
3024 result = ASIOGetSampleRate( ¤tRate );
3025 if ( result != ASE_OK ) {
3026 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error getting sample rate.";
3027 errorText_ = errorStream_.str();
3031 // Set the sample rate only if necessary
3032 if ( currentRate != sampleRate ) {
3033 result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate );
3034 if ( result != ASE_OK ) {
3035 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ").";
3036 errorText_ = errorStream_.str();
3041 // Determine the driver data type.
3042 ASIOChannelInfo channelInfo;
3043 channelInfo.channel = 0;
3044 if ( mode == OUTPUT ) channelInfo.isInput = false;
3045 else channelInfo.isInput = true;
3046 result = ASIOGetChannelInfo( &channelInfo );
3047 if ( result != ASE_OK ) {
3048 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting data format.";
3049 errorText_ = errorStream_.str();
3053 // Assuming WINDOWS host is always little-endian.
3054 stream_.doByteSwap[mode] = false;
3055 stream_.userFormat = format;
3056 stream_.deviceFormat[mode] = 0;
3057 if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) {
3058 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
3059 if ( channelInfo.type == ASIOSTInt16MSB ) stream_.doByteSwap[mode] = true;
3061 else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) {
3062 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
3063 if ( channelInfo.type == ASIOSTInt32MSB ) stream_.doByteSwap[mode] = true;
3065 else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) {
3066 stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
3067 if ( channelInfo.type == ASIOSTFloat32MSB ) stream_.doByteSwap[mode] = true;
3069 else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) {
3070 stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
3071 if ( channelInfo.type == ASIOSTFloat64MSB ) stream_.doByteSwap[mode] = true;
3073 else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB ) {
3074 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
3075 if ( channelInfo.type == ASIOSTInt24MSB ) stream_.doByteSwap[mode] = true;
3078 if ( stream_.deviceFormat[mode] == 0 ) {
3079 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") data format not supported by RtAudio.";
3080 errorText_ = errorStream_.str();
3084 // Set the buffer size. For a duplex stream, this will end up
3085 // setting the buffer size based on the input constraints, which
3087 long minSize, maxSize, preferSize, granularity;
3088 result = ASIOGetBufferSize( &minSize, &maxSize, &preferSize, &granularity );
3089 if ( result != ASE_OK ) {
3090 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting buffer size.";
3091 errorText_ = errorStream_.str();
3095 if ( isDuplexInput ) {
3096 // When this is the duplex input (output was opened before), then we have to use the same
3097 // buffersize as the output, because it might use the preferred buffer size, which most
3098 // likely wasn't passed as input to this. The buffer sizes have to be identically anyway,
3099 // So instead of throwing an error, make them equal. The caller uses the reference
3100 // to the "bufferSize" param as usual to set up processing buffers.
3102 *bufferSize = stream_.bufferSize;
3105 if ( *bufferSize == 0 ) *bufferSize = preferSize;
3106 else if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;
3107 else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;
3108 else if ( granularity == -1 ) {
3109 // Make sure bufferSize is a power of two.
3110 int log2_of_min_size = 0;
3111 int log2_of_max_size = 0;
3113 for ( unsigned int i = 0; i < sizeof(long) * 8; i++ ) {
3114 if ( minSize & ((long)1 << i) ) log2_of_min_size = i;
3115 if ( maxSize & ((long)1 << i) ) log2_of_max_size = i;
3118 long min_delta = std::abs( (long)*bufferSize - ((long)1 << log2_of_min_size) );
3119 int min_delta_num = log2_of_min_size;
3121 for (int i = log2_of_min_size + 1; i <= log2_of_max_size; i++) {
3122 long current_delta = std::abs( (long)*bufferSize - ((long)1 << i) );
3123 if (current_delta < min_delta) {
3124 min_delta = current_delta;
3129 *bufferSize = ( (unsigned int)1 << min_delta_num );
3130 if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;
3131 else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;
3133 else if ( granularity != 0 ) {
3134 // Set to an even multiple of granularity, rounding up.
3135 *bufferSize = (*bufferSize + granularity-1) / granularity * granularity;
3140 // we don't use it anymore, see above!
3141 // Just left it here for the case...
3142 if ( isDuplexInput && stream_.bufferSize != *bufferSize ) {
3143 errorText_ = "RtApiAsio::probeDeviceOpen: input/output buffersize discrepancy!";
3148 stream_.bufferSize = *bufferSize;
3149 stream_.nBuffers = 2;
3151 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
3152 else stream_.userInterleaved = true;
3154 // ASIO always uses non-interleaved buffers.
3155 stream_.deviceInterleaved[mode] = false;
3157 // Allocate, if necessary, our AsioHandle structure for the stream.
3158 if ( handle == 0 ) {
3160 handle = new AsioHandle;
3162 catch ( std::bad_alloc& ) {
3163 errorText_ = "RtApiAsio::probeDeviceOpen: error allocating AsioHandle memory.";
3166 handle->bufferInfos = 0;
3168 // Create a manual-reset event.
3169 handle->condition = CreateEvent( NULL, // no security
3170 TRUE, // manual-reset
3171 FALSE, // non-signaled initially
3173 stream_.apiHandle = (void *) handle;
3176 // Create the ASIO internal buffers. Since RtAudio sets up input
3177 // and output separately, we'll have to dispose of previously
3178 // created output buffers for a duplex stream.
3179 if ( mode == INPUT && stream_.mode == OUTPUT ) {
3180 ASIODisposeBuffers();
3181 if ( handle->bufferInfos ) free( handle->bufferInfos );
3184 // Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure.
3186 nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
3187 handle->bufferInfos = (ASIOBufferInfo *) malloc( nChannels * sizeof(ASIOBufferInfo) );
3188 if ( handle->bufferInfos == NULL ) {
3189 errorStream_ << "RtApiAsio::probeDeviceOpen: error allocating bufferInfo memory for driver (" << driverName << ").";
3190 errorText_ = errorStream_.str();
3194 ASIOBufferInfo *infos;
3195 infos = handle->bufferInfos;
3196 for ( i=0; i<stream_.nDeviceChannels[0]; i++, infos++ ) {
3197 infos->isInput = ASIOFalse;
3198 infos->channelNum = i + stream_.channelOffset[0];
3199 infos->buffers[0] = infos->buffers[1] = 0;
3201 for ( i=0; i<stream_.nDeviceChannels[1]; i++, infos++ ) {
3202 infos->isInput = ASIOTrue;
3203 infos->channelNum = i + stream_.channelOffset[1];
3204 infos->buffers[0] = infos->buffers[1] = 0;
3207 // prepare for callbacks
3208 stream_.sampleRate = sampleRate;
3209 stream_.device[mode] = device;
3210 stream_.mode = isDuplexInput ? DUPLEX : mode;
3212 // store this class instance before registering callbacks, that are going to use it
3213 asioCallbackInfo = &stream_.callbackInfo;
3214 stream_.callbackInfo.object = (void *) this;
3216 // Set up the ASIO callback structure and create the ASIO data buffers.
3217 asioCallbacks.bufferSwitch = &bufferSwitch;
3218 asioCallbacks.sampleRateDidChange = &sampleRateChanged;
3219 asioCallbacks.asioMessage = &asioMessages;
3220 asioCallbacks.bufferSwitchTimeInfo = NULL;
3221 result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks );
3222 if ( result != ASE_OK ) {
3223 // Standard method failed. This can happen with strict/misbehaving drivers that return valid buffer size ranges
3224 // but only accept the preferred buffer size as parameter for ASIOCreateBuffers (e.g. Creative's ASIO driver).
3225 // In that case, let's be naïve and try that instead.
3226 *bufferSize = preferSize;
3227 stream_.bufferSize = *bufferSize;
3228 result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks );
3231 if ( result != ASE_OK ) {
3232 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") creating buffers.";
3233 errorText_ = errorStream_.str();
3236 buffersAllocated = true;
3237 stream_.state = STREAM_STOPPED;
3239 // Set flags for buffer conversion.
3240 stream_.doConvertBuffer[mode] = false;
3241 if ( stream_.userFormat != stream_.deviceFormat[mode] )
3242 stream_.doConvertBuffer[mode] = true;
3243 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
3244 stream_.nUserChannels[mode] > 1 )
3245 stream_.doConvertBuffer[mode] = true;
3247 // Allocate necessary internal buffers
3248 unsigned long bufferBytes;
3249 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
3250 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
3251 if ( stream_.userBuffer[mode] == NULL ) {
3252 errorText_ = "RtApiAsio::probeDeviceOpen: error allocating user buffer memory.";
3256 if ( stream_.doConvertBuffer[mode] ) {
3258 bool makeBuffer = true;
3259 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
3260 if ( isDuplexInput && stream_.deviceBuffer ) {
3261 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
3262 if ( bufferBytes <= bytesOut ) makeBuffer = false;
3266 bufferBytes *= *bufferSize;
3267 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
3268 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
3269 if ( stream_.deviceBuffer == NULL ) {
3270 errorText_ = "RtApiAsio::probeDeviceOpen: error allocating device buffer memory.";
3276 // Determine device latencies
3277 long inputLatency, outputLatency;
3278 result = ASIOGetLatencies( &inputLatency, &outputLatency );
3279 if ( result != ASE_OK ) {
3280 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting latency.";
3281 errorText_ = errorStream_.str();
3282 error( RtAudioError::WARNING); // warn but don't fail
3285 stream_.latency[0] = outputLatency;
3286 stream_.latency[1] = inputLatency;
3289 // Setup the buffer conversion information structure. We don't use
3290 // buffers to do channel offsets, so we override that parameter
3292 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );
3297 if ( !isDuplexInput ) {
3298 // the cleanup for error in the duplex input, is done by RtApi::openStream
3299 // So we clean up for single channel only
3301 if ( buffersAllocated )
3302 ASIODisposeBuffers();
3304 drivers.removeCurrentDriver();
3307 CloseHandle( handle->condition );
3308 if ( handle->bufferInfos )
3309 free( handle->bufferInfos );
3312 stream_.apiHandle = 0;
3316 if ( stream_.userBuffer[mode] ) {
3317 free( stream_.userBuffer[mode] );
3318 stream_.userBuffer[mode] = 0;
3321 if ( stream_.deviceBuffer ) {
3322 free( stream_.deviceBuffer );
3323 stream_.deviceBuffer = 0;
3328 }////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
3330 void RtApiAsio :: closeStream()
3332 if ( stream_.state == STREAM_CLOSED ) {
3333 errorText_ = "RtApiAsio::closeStream(): no open stream to close!";
3334 error( RtAudioError::WARNING );
3338 if ( stream_.state == STREAM_RUNNING ) {
3339 stream_.state = STREAM_STOPPED;
3342 ASIODisposeBuffers();
3343 drivers.removeCurrentDriver();
3345 AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
3347 CloseHandle( handle->condition );
3348 if ( handle->bufferInfos )
3349 free( handle->bufferInfos );
3351 stream_.apiHandle = 0;
3354 for ( int i=0; i<2; i++ ) {
3355 if ( stream_.userBuffer[i] ) {
3356 free( stream_.userBuffer[i] );
3357 stream_.userBuffer[i] = 0;
3361 if ( stream_.deviceBuffer ) {
3362 free( stream_.deviceBuffer );
3363 stream_.deviceBuffer = 0;
3366 stream_.mode = UNINITIALIZED;
3367 stream_.state = STREAM_CLOSED;
3370 bool stopThreadCalled = false;
3372 void RtApiAsio :: startStream()
3375 if ( stream_.state == STREAM_RUNNING ) {
3376 errorText_ = "RtApiAsio::startStream(): the stream is already running!";
3377 error( RtAudioError::WARNING );
3381 AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
3382 ASIOError result = ASIOStart();
3383 if ( result != ASE_OK ) {
3384 errorStream_ << "RtApiAsio::startStream: error (" << getAsioErrorString( result ) << ") starting device.";
3385 errorText_ = errorStream_.str();
3389 handle->drainCounter = 0;
3390 handle->internalDrain = false;
3391 ResetEvent( handle->condition );
3392 stream_.state = STREAM_RUNNING;
3396 stopThreadCalled = false;
3398 if ( result == ASE_OK ) return;
3399 error( RtAudioError::SYSTEM_ERROR );
3402 void RtApiAsio :: stopStream()
3405 if ( stream_.state == STREAM_STOPPED ) {
3406 errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!";
3407 error( RtAudioError::WARNING );
3411 AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
3412 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
3413 if ( handle->drainCounter == 0 ) {
3414 handle->drainCounter = 2;
3415 WaitForSingleObject( handle->condition, INFINITE ); // block until signaled
3419 stream_.state = STREAM_STOPPED;
3421 ASIOError result = ASIOStop();
3422 if ( result != ASE_OK ) {
3423 errorStream_ << "RtApiAsio::stopStream: error (" << getAsioErrorString( result ) << ") stopping device.";
3424 errorText_ = errorStream_.str();
3427 if ( result == ASE_OK ) return;
3428 error( RtAudioError::SYSTEM_ERROR );
3431 void RtApiAsio :: abortStream()
3434 if ( stream_.state == STREAM_STOPPED ) {
3435 errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!";
3436 error( RtAudioError::WARNING );
3440 // The following lines were commented-out because some behavior was
3441 // noted where the device buffers need to be zeroed to avoid
3442 // continuing sound, even when the device buffers are completely
3443 // disposed. So now, calling abort is the same as calling stop.
3444 // AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
3445 // handle->drainCounter = 2;
3449 // This function will be called by a spawned thread when the user
3450 // callback function signals that the stream should be stopped or
3451 // aborted. It is necessary to handle it this way because the
3452 // callbackEvent() function must return before the ASIOStop()
3453 // function will return.
3454 static unsigned __stdcall asioStopStream( void *ptr )
3456 CallbackInfo *info = (CallbackInfo *) ptr;
3457 RtApiAsio *object = (RtApiAsio *) info->object;
3459 object->stopStream();
3464 bool RtApiAsio :: callbackEvent( long bufferIndex )
3466 if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;
3467 if ( stream_.state == STREAM_CLOSED ) {
3468 errorText_ = "RtApiAsio::callbackEvent(): the stream is closed ... this shouldn't happen!";
3469 error( RtAudioError::WARNING );
3473 CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
3474 AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
3476 // Check if we were draining the stream and signal if finished.
3477 if ( handle->drainCounter > 3 ) {
3479 stream_.state = STREAM_STOPPING;
3480 if ( handle->internalDrain == false )
3481 SetEvent( handle->condition );
3482 else { // spawn a thread to stop the stream
3484 stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream,
3485 &stream_.callbackInfo, 0, &threadId );
3490 // Invoke user callback to get fresh output data UNLESS we are
3492 if ( handle->drainCounter == 0 ) {
3493 RtAudioCallback callback = (RtAudioCallback) info->callback;
3494 double streamTime = getStreamTime();
3495 RtAudioStreamStatus status = 0;
3496 if ( stream_.mode != INPUT && asioXRun == true ) {
3497 status |= RTAUDIO_OUTPUT_UNDERFLOW;
3500 if ( stream_.mode != OUTPUT && asioXRun == true ) {
3501 status |= RTAUDIO_INPUT_OVERFLOW;
3504 int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
3505 stream_.bufferSize, streamTime, status, info->userData );
3506 if ( cbReturnValue == 2 ) {
3507 stream_.state = STREAM_STOPPING;
3508 handle->drainCounter = 2;
3510 stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream,
3511 &stream_.callbackInfo, 0, &threadId );
3514 else if ( cbReturnValue == 1 ) {
3515 handle->drainCounter = 1;
3516 handle->internalDrain = true;
3520 unsigned int nChannels, bufferBytes, i, j;
3521 nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
3522 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
3524 bufferBytes = stream_.bufferSize * formatBytes( stream_.deviceFormat[0] );
3526 if ( handle->drainCounter > 1 ) { // write zeros to the output stream
3528 for ( i=0, j=0; i<nChannels; i++ ) {
3529 if ( handle->bufferInfos[i].isInput != ASIOTrue )
3530 memset( handle->bufferInfos[i].buffers[bufferIndex], 0, bufferBytes );
3534 else if ( stream_.doConvertBuffer[0] ) {
3536 convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
3537 if ( stream_.doByteSwap[0] )
3538 byteSwapBuffer( stream_.deviceBuffer,
3539 stream_.bufferSize * stream_.nDeviceChannels[0],
3540 stream_.deviceFormat[0] );
3542 for ( i=0, j=0; i<nChannels; i++ ) {
3543 if ( handle->bufferInfos[i].isInput != ASIOTrue )
3544 memcpy( handle->bufferInfos[i].buffers[bufferIndex],
3545 &stream_.deviceBuffer[j++*bufferBytes], bufferBytes );
3551 if ( stream_.doByteSwap[0] )
3552 byteSwapBuffer( stream_.userBuffer[0],
3553 stream_.bufferSize * stream_.nUserChannels[0],
3554 stream_.userFormat );
3556 for ( i=0, j=0; i<nChannels; i++ ) {
3557 if ( handle->bufferInfos[i].isInput != ASIOTrue )
3558 memcpy( handle->bufferInfos[i].buffers[bufferIndex],
3559 &stream_.userBuffer[0][bufferBytes*j++], bufferBytes );
3565 // Don't bother draining input
3566 if ( handle->drainCounter ) {
3567 handle->drainCounter++;
3571 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
3573 bufferBytes = stream_.bufferSize * formatBytes(stream_.deviceFormat[1]);
3575 if (stream_.doConvertBuffer[1]) {
3577 // Always interleave ASIO input data.
3578 for ( i=0, j=0; i<nChannels; i++ ) {
3579 if ( handle->bufferInfos[i].isInput == ASIOTrue )
3580 memcpy( &stream_.deviceBuffer[j++*bufferBytes],
3581 handle->bufferInfos[i].buffers[bufferIndex],
3585 if ( stream_.doByteSwap[1] )
3586 byteSwapBuffer( stream_.deviceBuffer,
3587 stream_.bufferSize * stream_.nDeviceChannels[1],
3588 stream_.deviceFormat[1] );
3589 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
3593 for ( i=0, j=0; i<nChannels; i++ ) {
3594 if ( handle->bufferInfos[i].isInput == ASIOTrue ) {
3595 memcpy( &stream_.userBuffer[1][bufferBytes*j++],
3596 handle->bufferInfos[i].buffers[bufferIndex],
3601 if ( stream_.doByteSwap[1] )
3602 byteSwapBuffer( stream_.userBuffer[1],
3603 stream_.bufferSize * stream_.nUserChannels[1],
3604 stream_.userFormat );
3609 // The following call was suggested by Malte Clasen. While the API
3610 // documentation indicates it should not be required, some device
3611 // drivers apparently do not function correctly without it.
3614 RtApi::tickStreamTime();
3618 static void sampleRateChanged( ASIOSampleRate sRate )
3620 // The ASIO documentation says that this usually only happens during
3621 // external sync. Audio processing is not stopped by the driver,
3622 // actual sample rate might not have even changed, maybe only the
3623 // sample rate status of an AES/EBU or S/PDIF digital input at the
3626 RtApi *object = (RtApi *) asioCallbackInfo->object;
3628 object->stopStream();
3630 catch ( RtAudioError &exception ) {
3631 std::cerr << "\nRtApiAsio: sampleRateChanged() error (" << exception.getMessage() << ")!\n" << std::endl;
3635 std::cerr << "\nRtApiAsio: driver reports sample rate changed to " << sRate << " ... stream stopped!!!\n" << std::endl;
3638 static long asioMessages( long selector, long value, void* /*message*/, double* /*opt*/ )
3642 switch( selector ) {
3643 case kAsioSelectorSupported:
3644 if ( value == kAsioResetRequest
3645 || value == kAsioEngineVersion
3646 || value == kAsioResyncRequest
3647 || value == kAsioLatenciesChanged
3648 // The following three were added for ASIO 2.0, you don't
3649 // necessarily have to support them.
3650 || value == kAsioSupportsTimeInfo
3651 || value == kAsioSupportsTimeCode
3652 || value == kAsioSupportsInputMonitor)
3655 case kAsioResetRequest:
3656 // Defer the task and perform the reset of the driver during the
3657 // next "safe" situation. You cannot reset the driver right now,
3658 // as this code is called from the driver. Reset the driver is
3659 // done by completely destruct is. I.e. ASIOStop(),
3660 // ASIODisposeBuffers(), Destruction Afterwards you initialize the
3662 std::cerr << "\nRtApiAsio: driver reset requested!!!" << std::endl;
3665 case kAsioResyncRequest:
3666 // This informs the application that the driver encountered some
3667 // non-fatal data loss. It is used for synchronization purposes
3668 // of different media. Added mainly to work around the Win16Mutex
3669 // problems in Windows 95/98 with the Windows Multimedia system,
3670 // which could lose data because the Mutex was held too long by
3671 // another thread. However a driver can issue it in other
3673 // std::cerr << "\nRtApiAsio: driver resync requested!!!" << std::endl;
3677 case kAsioLatenciesChanged:
3678 // This will inform the host application that the drivers were
3679 // latencies changed. Beware, it this does not mean that the
3680 // buffer sizes have changed! You might need to update internal
3682 std::cerr << "\nRtApiAsio: driver latency may have changed!!!" << std::endl;
3685 case kAsioEngineVersion:
3686 // Return the supported ASIO version of the host application. If
3687 // a host application does not implement this selector, ASIO 1.0
3688 // is assumed by the driver.
3691 case kAsioSupportsTimeInfo:
3692 // Informs the driver whether the
3693 // asioCallbacks.bufferSwitchTimeInfo() callback is supported.
3694 // For compatibility with ASIO 1.0 drivers the host application
3695 // should always support the "old" bufferSwitch method, too.
3698 case kAsioSupportsTimeCode:
3699 // Informs the driver whether application is interested in time
3700 // code info. If an application does not need to know about time
3701 // code, the driver has less work to do.
3708 static const char* getAsioErrorString( ASIOError result )
3716 static const Messages m[] =
3718 { ASE_NotPresent, "Hardware input or output is not present or available." },
3719 { ASE_HWMalfunction, "Hardware is malfunctioning." },
3720 { ASE_InvalidParameter, "Invalid input parameter." },
3721 { ASE_InvalidMode, "Invalid mode." },
3722 { ASE_SPNotAdvancing, "Sample position not advancing." },
3723 { ASE_NoClock, "Sample clock or rate cannot be determined or is not present." },
3724 { ASE_NoMemory, "Not enough memory to complete the request." }
3727 for ( unsigned int i = 0; i < sizeof(m)/sizeof(m[0]); ++i )
3728 if ( m[i].value == result ) return m[i].message;
3730 return "Unknown error.";
3733 //******************** End of __WINDOWS_ASIO__ *********************//
3737 #if defined(__WINDOWS_WASAPI__) // Windows WASAPI API
3739 // Authored by Marcus Tomlinson <themarcustomlinson@gmail.com>, April 2014
3740 // - Introduces support for the Windows WASAPI API
3741 // - Aims to deliver bit streams to and from hardware at the lowest possible latency, via the absolute minimum buffer sizes required
3742 // - Provides flexible stream configuration to an otherwise strict and inflexible WASAPI interface
3743 // - Includes automatic internal conversion of sample rate and buffer size between hardware and the user
3750 #include <mferror.h>
3752 #include <mftransform.h>
3753 #include <wmcodecdsp.h>
3755 #include <audioclient.h>
3757 #include <mmdeviceapi.h>
3758 #include <functiondiscoverykeys_devpkey.h>
3760 #ifndef MF_E_TRANSFORM_NEED_MORE_INPUT
3761 #define MF_E_TRANSFORM_NEED_MORE_INPUT _HRESULT_TYPEDEF_(0xc00d6d72)
3764 #ifndef MFSTARTUP_NOSOCKET
3765 #define MFSTARTUP_NOSOCKET 0x1
3769 #pragma comment( lib, "ksuser" )
3770 #pragma comment( lib, "mfplat.lib" )
3771 #pragma comment( lib, "mfuuid.lib" )
3772 #pragma comment( lib, "wmcodecdspuuid" )
3775 //=============================================================================
3777 #define SAFE_RELEASE( objectPtr )\
3780 objectPtr->Release();\
3784 typedef HANDLE ( __stdcall *TAvSetMmThreadCharacteristicsPtr )( LPCWSTR TaskName, LPDWORD TaskIndex );
3786 //-----------------------------------------------------------------------------
3788 // WASAPI dictates stream sample rate, format, channel count, and in some cases, buffer size.
3789 // Therefore we must perform all necessary conversions to user buffers in order to satisfy these
3790 // requirements. WasapiBuffer ring buffers are used between HwIn->UserIn and UserOut->HwOut to
3791 // provide intermediate storage for read / write synchronization.
3805 // sets the length of the internal ring buffer
3806 void setBufferSize( unsigned int bufferSize, unsigned int formatBytes ) {
3809 buffer_ = ( char* ) calloc( bufferSize, formatBytes );
3811 bufferSize_ = bufferSize;
3816 // attempt to push a buffer into the ring buffer at the current "in" index
3817 bool pushBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format )
3819 if ( !buffer || // incoming buffer is NULL
3820 bufferSize == 0 || // incoming buffer has no data
3821 bufferSize > bufferSize_ ) // incoming buffer too large
3826 unsigned int relOutIndex = outIndex_;
3827 unsigned int inIndexEnd = inIndex_ + bufferSize;
3828 if ( relOutIndex < inIndex_ && inIndexEnd >= bufferSize_ ) {
3829 relOutIndex += bufferSize_;
3832 // "in" index can end on the "out" index but cannot begin at it
3833 if ( inIndex_ <= relOutIndex && inIndexEnd > relOutIndex ) {
3834 return false; // not enough space between "in" index and "out" index
3837 // copy buffer from external to internal
3838 int fromZeroSize = inIndex_ + bufferSize - bufferSize_;
3839 fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize;
3840 int fromInSize = bufferSize - fromZeroSize;
3845 memcpy( &( ( char* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( char ) );
3846 memcpy( buffer_, &( ( char* ) buffer )[fromInSize], fromZeroSize * sizeof( char ) );
3848 case RTAUDIO_SINT16:
3849 memcpy( &( ( short* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( short ) );
3850 memcpy( buffer_, &( ( short* ) buffer )[fromInSize], fromZeroSize * sizeof( short ) );
3852 case RTAUDIO_SINT24:
3853 memcpy( &( ( S24* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( S24 ) );
3854 memcpy( buffer_, &( ( S24* ) buffer )[fromInSize], fromZeroSize * sizeof( S24 ) );
3856 case RTAUDIO_SINT32:
3857 memcpy( &( ( int* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( int ) );
3858 memcpy( buffer_, &( ( int* ) buffer )[fromInSize], fromZeroSize * sizeof( int ) );
3860 case RTAUDIO_FLOAT32:
3861 memcpy( &( ( float* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( float ) );
3862 memcpy( buffer_, &( ( float* ) buffer )[fromInSize], fromZeroSize * sizeof( float ) );
3864 case RTAUDIO_FLOAT64:
3865 memcpy( &( ( double* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( double ) );
3866 memcpy( buffer_, &( ( double* ) buffer )[fromInSize], fromZeroSize * sizeof( double ) );
3870 // update "in" index
3871 inIndex_ += bufferSize;
3872 inIndex_ %= bufferSize_;
3877 // attempt to pull a buffer from the ring buffer from the current "out" index
3878 bool pullBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format )
3880 if ( !buffer || // incoming buffer is NULL
3881 bufferSize == 0 || // incoming buffer has no data
3882 bufferSize > bufferSize_ ) // incoming buffer too large
3887 unsigned int relInIndex = inIndex_;
3888 unsigned int outIndexEnd = outIndex_ + bufferSize;
3889 if ( relInIndex < outIndex_ && outIndexEnd >= bufferSize_ ) {
3890 relInIndex += bufferSize_;
3893 // "out" index can begin at and end on the "in" index
3894 if ( outIndex_ < relInIndex && outIndexEnd > relInIndex ) {
3895 return false; // not enough space between "out" index and "in" index
3898 // copy buffer from internal to external
3899 int fromZeroSize = outIndex_ + bufferSize - bufferSize_;
3900 fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize;
3901 int fromOutSize = bufferSize - fromZeroSize;
3906 memcpy( buffer, &( ( char* ) buffer_ )[outIndex_], fromOutSize * sizeof( char ) );
3907 memcpy( &( ( char* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( char ) );
3909 case RTAUDIO_SINT16:
3910 memcpy( buffer, &( ( short* ) buffer_ )[outIndex_], fromOutSize * sizeof( short ) );
3911 memcpy( &( ( short* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( short ) );
3913 case RTAUDIO_SINT24:
3914 memcpy( buffer, &( ( S24* ) buffer_ )[outIndex_], fromOutSize * sizeof( S24 ) );
3915 memcpy( &( ( S24* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( S24 ) );
3917 case RTAUDIO_SINT32:
3918 memcpy( buffer, &( ( int* ) buffer_ )[outIndex_], fromOutSize * sizeof( int ) );
3919 memcpy( &( ( int* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( int ) );
3921 case RTAUDIO_FLOAT32:
3922 memcpy( buffer, &( ( float* ) buffer_ )[outIndex_], fromOutSize * sizeof( float ) );
3923 memcpy( &( ( float* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( float ) );
3925 case RTAUDIO_FLOAT64:
3926 memcpy( buffer, &( ( double* ) buffer_ )[outIndex_], fromOutSize * sizeof( double ) );
3927 memcpy( &( ( double* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( double ) );
3931 // update "out" index
3932 outIndex_ += bufferSize;
3933 outIndex_ %= bufferSize_;
3940 unsigned int bufferSize_;
3941 unsigned int inIndex_;
3942 unsigned int outIndex_;
3945 //-----------------------------------------------------------------------------
3947 // In order to satisfy WASAPI's buffer requirements, we need a means of converting sample rate
3948 // between HW and the user. The WasapiResampler class is used to perform this conversion between
3949 // HwIn->UserIn and UserOut->HwOut during the stream callback loop.
3950 class WasapiResampler
3953 WasapiResampler( bool isFloat, unsigned int bitsPerSample, unsigned int channelCount,
3954 unsigned int inSampleRate, unsigned int outSampleRate )
3955 : _bytesPerSample( bitsPerSample / 8 )
3956 , _channelCount( channelCount )
3957 , _sampleRatio( ( float ) outSampleRate / inSampleRate )
3958 , _transformUnk( NULL )
3959 , _transform( NULL )
3960 , _mediaType( NULL )
3961 , _inputMediaType( NULL )
3962 , _outputMediaType( NULL )
3964 #ifdef __IWMResamplerProps_FWD_DEFINED__
3965 , _resamplerProps( NULL )
3968 // 1. Initialization
3970 MFStartup( MF_VERSION, MFSTARTUP_NOSOCKET );
3972 // 2. Create Resampler Transform Object
3974 CoCreateInstance( CLSID_CResamplerMediaObject, NULL, CLSCTX_INPROC_SERVER,
3975 IID_IUnknown, ( void** ) &_transformUnk );
3977 _transformUnk->QueryInterface( IID_PPV_ARGS( &_transform ) );
3979 #ifdef __IWMResamplerProps_FWD_DEFINED__
3980 _transformUnk->QueryInterface( IID_PPV_ARGS( &_resamplerProps ) );
3981 _resamplerProps->SetHalfFilterLength( 60 ); // best conversion quality
3984 // 3. Specify input / output format
3986 MFCreateMediaType( &_mediaType );
3987 _mediaType->SetGUID( MF_MT_MAJOR_TYPE, MFMediaType_Audio );
3988 _mediaType->SetGUID( MF_MT_SUBTYPE, isFloat ? MFAudioFormat_Float : MFAudioFormat_PCM );
3989 _mediaType->SetUINT32( MF_MT_AUDIO_NUM_CHANNELS, channelCount );
3990 _mediaType->SetUINT32( MF_MT_AUDIO_SAMPLES_PER_SECOND, inSampleRate );
3991 _mediaType->SetUINT32( MF_MT_AUDIO_BLOCK_ALIGNMENT, _bytesPerSample * channelCount );
3992 _mediaType->SetUINT32( MF_MT_AUDIO_AVG_BYTES_PER_SECOND, _bytesPerSample * channelCount * inSampleRate );
3993 _mediaType->SetUINT32( MF_MT_AUDIO_BITS_PER_SAMPLE, bitsPerSample );
3994 _mediaType->SetUINT32( MF_MT_ALL_SAMPLES_INDEPENDENT, TRUE );
3996 MFCreateMediaType( &_inputMediaType );
3997 _mediaType->CopyAllItems( _inputMediaType );
3999 _transform->SetInputType( 0, _inputMediaType, 0 );
4001 MFCreateMediaType( &_outputMediaType );
4002 _mediaType->CopyAllItems( _outputMediaType );
4004 _outputMediaType->SetUINT32( MF_MT_AUDIO_SAMPLES_PER_SECOND, outSampleRate );
4005 _outputMediaType->SetUINT32( MF_MT_AUDIO_AVG_BYTES_PER_SECOND, _bytesPerSample * channelCount * outSampleRate );
4007 _transform->SetOutputType( 0, _outputMediaType, 0 );
4009 // 4. Send stream start messages to Resampler
4011 _transform->ProcessMessage( MFT_MESSAGE_COMMAND_FLUSH, 0 );
4012 _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_BEGIN_STREAMING, 0 );
4013 _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_START_OF_STREAM, 0 );
4018 // 8. Send stream stop messages to Resampler
4020 _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_END_OF_STREAM, 0 );
4021 _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_END_STREAMING, 0 );
4027 SAFE_RELEASE( _transformUnk );
4028 SAFE_RELEASE( _transform );
4029 SAFE_RELEASE( _mediaType );
4030 SAFE_RELEASE( _inputMediaType );
4031 SAFE_RELEASE( _outputMediaType );
4033 #ifdef __IWMResamplerProps_FWD_DEFINED__
4034 SAFE_RELEASE( _resamplerProps );
4038 void Convert( char* outBuffer, const char* inBuffer, unsigned int inSampleCount, unsigned int& outSampleCount )
4040 unsigned int inputBufferSize = _bytesPerSample * _channelCount * inSampleCount;
4041 if ( _sampleRatio == 1 )
4043 // no sample rate conversion required
4044 memcpy( outBuffer, inBuffer, inputBufferSize );
4045 outSampleCount = inSampleCount;
4049 unsigned int outputBufferSize = ( unsigned int ) ceilf( inputBufferSize * _sampleRatio ) + ( _bytesPerSample * _channelCount );
4051 IMFMediaBuffer* rInBuffer;
4052 IMFSample* rInSample;
4053 BYTE* rInByteBuffer = NULL;
4055 // 5. Create Sample object from input data
4057 MFCreateMemoryBuffer( inputBufferSize, &rInBuffer );
4059 rInBuffer->Lock( &rInByteBuffer, NULL, NULL );
4060 memcpy( rInByteBuffer, inBuffer, inputBufferSize );
4061 rInBuffer->Unlock();
4062 rInByteBuffer = NULL;
4064 rInBuffer->SetCurrentLength( inputBufferSize );
4066 MFCreateSample( &rInSample );
4067 rInSample->AddBuffer( rInBuffer );
4069 // 6. Pass input data to Resampler
4071 _transform->ProcessInput( 0, rInSample, 0 );
4073 SAFE_RELEASE( rInBuffer );
4074 SAFE_RELEASE( rInSample );
4076 // 7. Perform sample rate conversion
4078 IMFMediaBuffer* rOutBuffer = NULL;
4079 BYTE* rOutByteBuffer = NULL;
4081 MFT_OUTPUT_DATA_BUFFER rOutDataBuffer;
4083 DWORD rBytes = outputBufferSize; // maximum bytes accepted per ProcessOutput
4085 // 7.1 Create Sample object for output data
4087 memset( &rOutDataBuffer, 0, sizeof rOutDataBuffer );
4088 MFCreateSample( &( rOutDataBuffer.pSample ) );
4089 MFCreateMemoryBuffer( rBytes, &rOutBuffer );
4090 rOutDataBuffer.pSample->AddBuffer( rOutBuffer );
4091 rOutDataBuffer.dwStreamID = 0;
4092 rOutDataBuffer.dwStatus = 0;
4093 rOutDataBuffer.pEvents = NULL;
4095 // 7.2 Get output data from Resampler
4097 if ( _transform->ProcessOutput( 0, 1, &rOutDataBuffer, &rStatus ) == MF_E_TRANSFORM_NEED_MORE_INPUT )
4100 SAFE_RELEASE( rOutBuffer );
4101 SAFE_RELEASE( rOutDataBuffer.pSample );
4105 // 7.3 Write output data to outBuffer
4107 SAFE_RELEASE( rOutBuffer );
4108 rOutDataBuffer.pSample->ConvertToContiguousBuffer( &rOutBuffer );
4109 rOutBuffer->GetCurrentLength( &rBytes );
4111 rOutBuffer->Lock( &rOutByteBuffer, NULL, NULL );
4112 memcpy( outBuffer, rOutByteBuffer, rBytes );
4113 rOutBuffer->Unlock();
4114 rOutByteBuffer = NULL;
4116 outSampleCount = rBytes / _bytesPerSample / _channelCount;
4117 SAFE_RELEASE( rOutBuffer );
4118 SAFE_RELEASE( rOutDataBuffer.pSample );
4122 unsigned int _bytesPerSample;
4123 unsigned int _channelCount;
4126 IUnknown* _transformUnk;
4127 IMFTransform* _transform;
4128 IMFMediaType* _mediaType;
4129 IMFMediaType* _inputMediaType;
4130 IMFMediaType* _outputMediaType;
4132 #ifdef __IWMResamplerProps_FWD_DEFINED__
4133 IWMResamplerProps* _resamplerProps;
4137 //-----------------------------------------------------------------------------
4139 // A structure to hold various information related to the WASAPI implementation.
4142 IAudioClient* captureAudioClient;
4143 IAudioClient* renderAudioClient;
4144 IAudioCaptureClient* captureClient;
4145 IAudioRenderClient* renderClient;
4146 HANDLE captureEvent;
4150 : captureAudioClient( NULL ),
4151 renderAudioClient( NULL ),
4152 captureClient( NULL ),
4153 renderClient( NULL ),
4154 captureEvent( NULL ),
4155 renderEvent( NULL ) {}
4158 //=============================================================================
4160 RtApiWasapi::RtApiWasapi()
4161 : coInitialized_( false ), deviceEnumerator_( NULL )
4163 // WASAPI can run either apartment or multi-threaded
4164 HRESULT hr = CoInitialize( NULL );
4165 if ( !FAILED( hr ) )
4166 coInitialized_ = true;
4168 // Instantiate device enumerator
4169 hr = CoCreateInstance( __uuidof( MMDeviceEnumerator ), NULL,
4170 CLSCTX_ALL, __uuidof( IMMDeviceEnumerator ),
4171 ( void** ) &deviceEnumerator_ );
4173 // If this runs on an old Windows, it will fail. Ignore and proceed.
4175 deviceEnumerator_ = NULL;
4178 //-----------------------------------------------------------------------------
4180 RtApiWasapi::~RtApiWasapi()
4182 if ( stream_.state != STREAM_CLOSED )
4185 SAFE_RELEASE( deviceEnumerator_ );
4187 // If this object previously called CoInitialize()
4188 if ( coInitialized_ )
4192 //=============================================================================
4194 unsigned int RtApiWasapi::getDeviceCount( void )
4196 unsigned int captureDeviceCount = 0;
4197 unsigned int renderDeviceCount = 0;
4199 IMMDeviceCollection* captureDevices = NULL;
4200 IMMDeviceCollection* renderDevices = NULL;
4202 if ( !deviceEnumerator_ )
4205 // Count capture devices
4207 HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
4208 if ( FAILED( hr ) ) {
4209 errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device collection.";
4213 hr = captureDevices->GetCount( &captureDeviceCount );
4214 if ( FAILED( hr ) ) {
4215 errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device count.";
4219 // Count render devices
4220 hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
4221 if ( FAILED( hr ) ) {
4222 errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device collection.";
4226 hr = renderDevices->GetCount( &renderDeviceCount );
4227 if ( FAILED( hr ) ) {
4228 errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device count.";
4233 // release all references
4234 SAFE_RELEASE( captureDevices );
4235 SAFE_RELEASE( renderDevices );
4237 if ( errorText_.empty() )
4238 return captureDeviceCount + renderDeviceCount;
4240 error( RtAudioError::DRIVER_ERROR );
4244 //-----------------------------------------------------------------------------
4246 RtAudio::DeviceInfo RtApiWasapi::getDeviceInfo( unsigned int device )
4248 RtAudio::DeviceInfo info;
4249 unsigned int captureDeviceCount = 0;
4250 unsigned int renderDeviceCount = 0;
4251 std::string defaultDeviceName;
4252 bool isCaptureDevice = false;
4254 PROPVARIANT deviceNameProp;
4255 PROPVARIANT defaultDeviceNameProp;
4257 IMMDeviceCollection* captureDevices = NULL;
4258 IMMDeviceCollection* renderDevices = NULL;
4259 IMMDevice* devicePtr = NULL;
4260 IMMDevice* defaultDevicePtr = NULL;
4261 IAudioClient* audioClient = NULL;
4262 IPropertyStore* devicePropStore = NULL;
4263 IPropertyStore* defaultDevicePropStore = NULL;
4265 WAVEFORMATEX* deviceFormat = NULL;
4266 WAVEFORMATEX* closestMatchFormat = NULL;
4269 info.probed = false;
4271 // Count capture devices
4273 RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
4274 HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
4275 if ( FAILED( hr ) ) {
4276 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device collection.";
4280 hr = captureDevices->GetCount( &captureDeviceCount );
4281 if ( FAILED( hr ) ) {
4282 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device count.";
4286 // Count render devices
4287 hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
4288 if ( FAILED( hr ) ) {
4289 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device collection.";
4293 hr = renderDevices->GetCount( &renderDeviceCount );
4294 if ( FAILED( hr ) ) {
4295 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device count.";
4299 // validate device index
4300 if ( device >= captureDeviceCount + renderDeviceCount ) {
4301 errorText_ = "RtApiWasapi::getDeviceInfo: Invalid device index.";
4302 errorType = RtAudioError::INVALID_USE;
4306 // determine whether index falls within capture or render devices
4307 if ( device >= renderDeviceCount ) {
4308 hr = captureDevices->Item( device - renderDeviceCount, &devicePtr );
4309 if ( FAILED( hr ) ) {
4310 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device handle.";
4313 isCaptureDevice = true;
4316 hr = renderDevices->Item( device, &devicePtr );
4317 if ( FAILED( hr ) ) {
4318 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device handle.";
4321 isCaptureDevice = false;
4324 // get default device name
4325 if ( isCaptureDevice ) {
4326 hr = deviceEnumerator_->GetDefaultAudioEndpoint( eCapture, eConsole, &defaultDevicePtr );
4327 if ( FAILED( hr ) ) {
4328 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default capture device handle.";
4333 hr = deviceEnumerator_->GetDefaultAudioEndpoint( eRender, eConsole, &defaultDevicePtr );
4334 if ( FAILED( hr ) ) {
4335 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default render device handle.";
4340 hr = defaultDevicePtr->OpenPropertyStore( STGM_READ, &defaultDevicePropStore );
4341 if ( FAILED( hr ) ) {
4342 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open default device property store.";
4345 PropVariantInit( &defaultDeviceNameProp );
4347 hr = defaultDevicePropStore->GetValue( PKEY_Device_FriendlyName, &defaultDeviceNameProp );
4348 if ( FAILED( hr ) ) {
4349 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default device property: PKEY_Device_FriendlyName.";
4353 defaultDeviceName = convertCharPointerToStdString(defaultDeviceNameProp.pwszVal);
4356 hr = devicePtr->OpenPropertyStore( STGM_READ, &devicePropStore );
4357 if ( FAILED( hr ) ) {
4358 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open device property store.";
4362 PropVariantInit( &deviceNameProp );
4364 hr = devicePropStore->GetValue( PKEY_Device_FriendlyName, &deviceNameProp );
4365 if ( FAILED( hr ) ) {
4366 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device property: PKEY_Device_FriendlyName.";
4370 info.name =convertCharPointerToStdString(deviceNameProp.pwszVal);
4373 if ( isCaptureDevice ) {
4374 info.isDefaultInput = info.name == defaultDeviceName;
4375 info.isDefaultOutput = false;
4378 info.isDefaultInput = false;
4379 info.isDefaultOutput = info.name == defaultDeviceName;
4383 hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, NULL, ( void** ) &audioClient );
4384 if ( FAILED( hr ) ) {
4385 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device audio client.";
4389 hr = audioClient->GetMixFormat( &deviceFormat );
4390 if ( FAILED( hr ) ) {
4391 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device mix format.";
4395 if ( isCaptureDevice ) {
4396 info.inputChannels = deviceFormat->nChannels;
4397 info.outputChannels = 0;
4398 info.duplexChannels = 0;
4401 info.inputChannels = 0;
4402 info.outputChannels = deviceFormat->nChannels;
4403 info.duplexChannels = 0;
4407 info.sampleRates.clear();
4409 // allow support for all sample rates as we have a built-in sample rate converter
4410 for ( unsigned int i = 0; i < MAX_SAMPLE_RATES; i++ ) {
4411 info.sampleRates.push_back( SAMPLE_RATES[i] );
4413 info.preferredSampleRate = deviceFormat->nSamplesPerSec;
4416 info.nativeFormats = 0;
4418 if ( deviceFormat->wFormatTag == WAVE_FORMAT_IEEE_FLOAT ||
4419 ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
4420 ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_IEEE_FLOAT ) )
4422 if ( deviceFormat->wBitsPerSample == 32 ) {
4423 info.nativeFormats |= RTAUDIO_FLOAT32;
4425 else if ( deviceFormat->wBitsPerSample == 64 ) {
4426 info.nativeFormats |= RTAUDIO_FLOAT64;
4429 else if ( deviceFormat->wFormatTag == WAVE_FORMAT_PCM ||
4430 ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
4431 ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_PCM ) )
4433 if ( deviceFormat->wBitsPerSample == 8 ) {
4434 info.nativeFormats |= RTAUDIO_SINT8;
4436 else if ( deviceFormat->wBitsPerSample == 16 ) {
4437 info.nativeFormats |= RTAUDIO_SINT16;
4439 else if ( deviceFormat->wBitsPerSample == 24 ) {
4440 info.nativeFormats |= RTAUDIO_SINT24;
4442 else if ( deviceFormat->wBitsPerSample == 32 ) {
4443 info.nativeFormats |= RTAUDIO_SINT32;
4451 // release all references
4452 PropVariantClear( &deviceNameProp );
4453 PropVariantClear( &defaultDeviceNameProp );
4455 SAFE_RELEASE( captureDevices );
4456 SAFE_RELEASE( renderDevices );
4457 SAFE_RELEASE( devicePtr );
4458 SAFE_RELEASE( defaultDevicePtr );
4459 SAFE_RELEASE( audioClient );
4460 SAFE_RELEASE( devicePropStore );
4461 SAFE_RELEASE( defaultDevicePropStore );
4463 CoTaskMemFree( deviceFormat );
4464 CoTaskMemFree( closestMatchFormat );
4466 if ( !errorText_.empty() )
4471 //-----------------------------------------------------------------------------
4473 unsigned int RtApiWasapi::getDefaultOutputDevice( void )
4475 for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {
4476 if ( getDeviceInfo( i ).isDefaultOutput ) {
4484 //-----------------------------------------------------------------------------
4486 unsigned int RtApiWasapi::getDefaultInputDevice( void )
4488 for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {
4489 if ( getDeviceInfo( i ).isDefaultInput ) {
4497 //-----------------------------------------------------------------------------
4499 void RtApiWasapi::closeStream( void )
4501 if ( stream_.state == STREAM_CLOSED ) {
4502 errorText_ = "RtApiWasapi::closeStream: No open stream to close.";
4503 error( RtAudioError::WARNING );
4507 if ( stream_.state != STREAM_STOPPED )
4510 // clean up stream memory
4511 SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient )
4512 SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient )
4514 SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureClient )
4515 SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderClient )
4517 if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent )
4518 CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent );
4520 if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent )
4521 CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent );
4523 delete ( WasapiHandle* ) stream_.apiHandle;
4524 stream_.apiHandle = NULL;
4526 for ( int i = 0; i < 2; i++ ) {
4527 if ( stream_.userBuffer[i] ) {
4528 free( stream_.userBuffer[i] );
4529 stream_.userBuffer[i] = 0;
4533 if ( stream_.deviceBuffer ) {
4534 free( stream_.deviceBuffer );
4535 stream_.deviceBuffer = 0;
4538 // update stream state
4539 stream_.state = STREAM_CLOSED;
4542 //-----------------------------------------------------------------------------
4544 void RtApiWasapi::startStream( void )
4548 if ( stream_.state == STREAM_RUNNING ) {
4549 errorText_ = "RtApiWasapi::startStream: The stream is already running.";
4550 error( RtAudioError::WARNING );
4554 // update stream state
4555 stream_.state = STREAM_RUNNING;
4557 // create WASAPI stream thread
4558 stream_.callbackInfo.thread = ( ThreadHandle ) CreateThread( NULL, 0, runWasapiThread, this, CREATE_SUSPENDED, NULL );
4560 if ( !stream_.callbackInfo.thread ) {
4561 errorText_ = "RtApiWasapi::startStream: Unable to instantiate callback thread.";
4562 error( RtAudioError::THREAD_ERROR );
4565 SetThreadPriority( ( void* ) stream_.callbackInfo.thread, stream_.callbackInfo.priority );
4566 ResumeThread( ( void* ) stream_.callbackInfo.thread );
4570 //-----------------------------------------------------------------------------
4572 void RtApiWasapi::stopStream( void )
4576 if ( stream_.state == STREAM_STOPPED ) {
4577 errorText_ = "RtApiWasapi::stopStream: The stream is already stopped.";
4578 error( RtAudioError::WARNING );
4582 // inform stream thread by setting stream state to STREAM_STOPPING
4583 stream_.state = STREAM_STOPPING;
4585 // wait until stream thread is stopped
4586 while( stream_.state != STREAM_STOPPED ) {
4590 // Wait for the last buffer to play before stopping.
4591 Sleep( 1000 * stream_.bufferSize / stream_.sampleRate );
4593 // stop capture client if applicable
4594 if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) {
4595 HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop();
4596 if ( FAILED( hr ) ) {
4597 errorText_ = "RtApiWasapi::stopStream: Unable to stop capture stream.";
4598 error( RtAudioError::DRIVER_ERROR );
4603 // stop render client if applicable
4604 if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) {
4605 HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop();
4606 if ( FAILED( hr ) ) {
4607 errorText_ = "RtApiWasapi::stopStream: Unable to stop render stream.";
4608 error( RtAudioError::DRIVER_ERROR );
4613 // close thread handle
4614 if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
4615 errorText_ = "RtApiWasapi::stopStream: Unable to close callback thread.";
4616 error( RtAudioError::THREAD_ERROR );
4620 stream_.callbackInfo.thread = (ThreadHandle) NULL;
4623 //-----------------------------------------------------------------------------
4625 void RtApiWasapi::abortStream( void )
4629 if ( stream_.state == STREAM_STOPPED ) {
4630 errorText_ = "RtApiWasapi::abortStream: The stream is already stopped.";
4631 error( RtAudioError::WARNING );
4635 // inform stream thread by setting stream state to STREAM_STOPPING
4636 stream_.state = STREAM_STOPPING;
4638 // wait until stream thread is stopped
4639 while ( stream_.state != STREAM_STOPPED ) {
4643 // stop capture client if applicable
4644 if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) {
4645 HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop();
4646 if ( FAILED( hr ) ) {
4647 errorText_ = "RtApiWasapi::abortStream: Unable to stop capture stream.";
4648 error( RtAudioError::DRIVER_ERROR );
4653 // stop render client if applicable
4654 if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) {
4655 HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop();
4656 if ( FAILED( hr ) ) {
4657 errorText_ = "RtApiWasapi::abortStream: Unable to stop render stream.";
4658 error( RtAudioError::DRIVER_ERROR );
4663 // close thread handle
4664 if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
4665 errorText_ = "RtApiWasapi::abortStream: Unable to close callback thread.";
4666 error( RtAudioError::THREAD_ERROR );
4670 stream_.callbackInfo.thread = (ThreadHandle) NULL;
4673 //-----------------------------------------------------------------------------
4675 bool RtApiWasapi::probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
4676 unsigned int firstChannel, unsigned int sampleRate,
4677 RtAudioFormat format, unsigned int* bufferSize,
4678 RtAudio::StreamOptions* options )
4680 bool methodResult = FAILURE;
4681 unsigned int captureDeviceCount = 0;
4682 unsigned int renderDeviceCount = 0;
4684 IMMDeviceCollection* captureDevices = NULL;
4685 IMMDeviceCollection* renderDevices = NULL;
4686 IMMDevice* devicePtr = NULL;
4687 WAVEFORMATEX* deviceFormat = NULL;
4688 unsigned int bufferBytes;
4689 stream_.state = STREAM_STOPPED;
4691 // create API Handle if not already created
4692 if ( !stream_.apiHandle )
4693 stream_.apiHandle = ( void* ) new WasapiHandle();
4695 // Count capture devices
4697 RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
4698 HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
4699 if ( FAILED( hr ) ) {
4700 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device collection.";
4704 hr = captureDevices->GetCount( &captureDeviceCount );
4705 if ( FAILED( hr ) ) {
4706 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device count.";
4710 // Count render devices
4711 hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
4712 if ( FAILED( hr ) ) {
4713 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device collection.";
4717 hr = renderDevices->GetCount( &renderDeviceCount );
4718 if ( FAILED( hr ) ) {
4719 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device count.";
4723 // validate device index
4724 if ( device >= captureDeviceCount + renderDeviceCount ) {
4725 errorType = RtAudioError::INVALID_USE;
4726 errorText_ = "RtApiWasapi::probeDeviceOpen: Invalid device index.";
4730 // if device index falls within capture devices
4731 if ( device >= renderDeviceCount ) {
4732 if ( mode != INPUT ) {
4733 errorType = RtAudioError::INVALID_USE;
4734 errorText_ = "RtApiWasapi::probeDeviceOpen: Capture device selected as output device.";
4738 // retrieve captureAudioClient from devicePtr
4739 IAudioClient*& captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
4741 hr = captureDevices->Item( device - renderDeviceCount, &devicePtr );
4742 if ( FAILED( hr ) ) {
4743 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device handle.";
4747 hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
4748 NULL, ( void** ) &captureAudioClient );
4749 if ( FAILED( hr ) ) {
4750 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device audio client.";
4754 hr = captureAudioClient->GetMixFormat( &deviceFormat );
4755 if ( FAILED( hr ) ) {
4756 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device mix format.";
4760 stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
4761 captureAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
4764 // if device index falls within render devices and is configured for loopback
4765 if ( device < renderDeviceCount && mode == INPUT )
4767 // if renderAudioClient is not initialised, initialise it now
4768 IAudioClient*& renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
4769 if ( !renderAudioClient )
4771 probeDeviceOpen( device, OUTPUT, channels, firstChannel, sampleRate, format, bufferSize, options );
4774 // retrieve captureAudioClient from devicePtr
4775 IAudioClient*& captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
4777 hr = renderDevices->Item( device, &devicePtr );
4778 if ( FAILED( hr ) ) {
4779 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device handle.";
4783 hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
4784 NULL, ( void** ) &captureAudioClient );
4785 if ( FAILED( hr ) ) {
4786 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device audio client.";
4790 hr = captureAudioClient->GetMixFormat( &deviceFormat );
4791 if ( FAILED( hr ) ) {
4792 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device mix format.";
4796 stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
4797 captureAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
4800 // if device index falls within render devices and is configured for output
4801 if ( device < renderDeviceCount && mode == OUTPUT )
4803 // if renderAudioClient is already initialised, don't initialise it again
4804 IAudioClient*& renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
4805 if ( renderAudioClient )
4807 methodResult = SUCCESS;
4811 hr = renderDevices->Item( device, &devicePtr );
4812 if ( FAILED( hr ) ) {
4813 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device handle.";
4817 hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
4818 NULL, ( void** ) &renderAudioClient );
4819 if ( FAILED( hr ) ) {
4820 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device audio client.";
4824 hr = renderAudioClient->GetMixFormat( &deviceFormat );
4825 if ( FAILED( hr ) ) {
4826 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device mix format.";
4830 stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
4831 renderAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
4835 if ( ( stream_.mode == OUTPUT && mode == INPUT ) ||
4836 ( stream_.mode == INPUT && mode == OUTPUT ) ) {
4837 stream_.mode = DUPLEX;
4840 stream_.mode = mode;
4843 stream_.device[mode] = device;
4844 stream_.doByteSwap[mode] = false;
4845 stream_.sampleRate = sampleRate;
4846 stream_.bufferSize = *bufferSize;
4847 stream_.nBuffers = 1;
4848 stream_.nUserChannels[mode] = channels;
4849 stream_.channelOffset[mode] = firstChannel;
4850 stream_.userFormat = format;
4851 stream_.deviceFormat[mode] = getDeviceInfo( device ).nativeFormats;
4853 if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
4854 stream_.userInterleaved = false;
4856 stream_.userInterleaved = true;
4857 stream_.deviceInterleaved[mode] = true;
4859 // Set flags for buffer conversion.
4860 stream_.doConvertBuffer[mode] = false;
4861 if ( stream_.userFormat != stream_.deviceFormat[mode] ||
4862 stream_.nUserChannels[0] != stream_.nDeviceChannels[0] ||
4863 stream_.nUserChannels[1] != stream_.nDeviceChannels[1] )
4864 stream_.doConvertBuffer[mode] = true;
4865 else if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
4866 stream_.nUserChannels[mode] > 1 )
4867 stream_.doConvertBuffer[mode] = true;
4869 if ( stream_.doConvertBuffer[mode] )
4870 setConvertInfo( mode, 0 );
4872 // Allocate necessary internal buffers
4873 bufferBytes = stream_.nUserChannels[mode] * stream_.bufferSize * formatBytes( stream_.userFormat );
4875 stream_.userBuffer[mode] = ( char* ) calloc( bufferBytes, 1 );
4876 if ( !stream_.userBuffer[mode] ) {
4877 errorType = RtAudioError::MEMORY_ERROR;
4878 errorText_ = "RtApiWasapi::probeDeviceOpen: Error allocating user buffer memory.";
4882 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME )
4883 stream_.callbackInfo.priority = 15;
4885 stream_.callbackInfo.priority = 0;
4887 ///! TODO: RTAUDIO_MINIMIZE_LATENCY // Provide stream buffers directly to callback
4888 ///! TODO: RTAUDIO_HOG_DEVICE // Exclusive mode
4890 methodResult = SUCCESS;
4894 SAFE_RELEASE( captureDevices );
4895 SAFE_RELEASE( renderDevices );
4896 SAFE_RELEASE( devicePtr );
4897 CoTaskMemFree( deviceFormat );
4899 // if method failed, close the stream
4900 if ( methodResult == FAILURE )
4903 if ( !errorText_.empty() )
4905 return methodResult;
4908 //=============================================================================
4910 DWORD WINAPI RtApiWasapi::runWasapiThread( void* wasapiPtr )
4913 ( ( RtApiWasapi* ) wasapiPtr )->wasapiThread();
4918 DWORD WINAPI RtApiWasapi::stopWasapiThread( void* wasapiPtr )
4921 ( ( RtApiWasapi* ) wasapiPtr )->stopStream();
4926 DWORD WINAPI RtApiWasapi::abortWasapiThread( void* wasapiPtr )
4929 ( ( RtApiWasapi* ) wasapiPtr )->abortStream();
4934 //-----------------------------------------------------------------------------
4936 void RtApiWasapi::wasapiThread()
4938 // as this is a new thread, we must CoInitialize it
4939 CoInitialize( NULL );
4943 IAudioClient* captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
4944 IAudioClient* renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
4945 IAudioCaptureClient* captureClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureClient;
4946 IAudioRenderClient* renderClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderClient;
4947 HANDLE captureEvent = ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent;
4948 HANDLE renderEvent = ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent;
4950 WAVEFORMATEX* captureFormat = NULL;
4951 WAVEFORMATEX* renderFormat = NULL;
4952 float captureSrRatio = 0.0f;
4953 float renderSrRatio = 0.0f;
4954 WasapiBuffer captureBuffer;
4955 WasapiBuffer renderBuffer;
4956 WasapiResampler* captureResampler = NULL;
4957 WasapiResampler* renderResampler = NULL;
4959 // declare local stream variables
4960 RtAudioCallback callback = ( RtAudioCallback ) stream_.callbackInfo.callback;
4961 BYTE* streamBuffer = NULL;
4962 unsigned long captureFlags = 0;
4963 unsigned int bufferFrameCount = 0;
4964 unsigned int numFramesPadding = 0;
4965 unsigned int convBufferSize = 0;
4966 bool loopbackEnabled = stream_.device[INPUT] == stream_.device[OUTPUT];
4967 bool callbackPushed = true;
4968 bool callbackPulled = false;
4969 bool callbackStopped = false;
4970 int callbackResult = 0;
4972 // convBuffer is used to store converted buffers between WASAPI and the user
4973 char* convBuffer = NULL;
4974 unsigned int convBuffSize = 0;
4975 unsigned int deviceBuffSize = 0;
4978 RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
4980 // Attempt to assign "Pro Audio" characteristic to thread
4981 HMODULE AvrtDll = LoadLibrary( (LPCTSTR) "AVRT.dll" );
4983 DWORD taskIndex = 0;
4984 TAvSetMmThreadCharacteristicsPtr AvSetMmThreadCharacteristicsPtr = ( TAvSetMmThreadCharacteristicsPtr ) GetProcAddress( AvrtDll, "AvSetMmThreadCharacteristicsW" );
4985 AvSetMmThreadCharacteristicsPtr( L"Pro Audio", &taskIndex );
4986 FreeLibrary( AvrtDll );
4989 // start capture stream if applicable
4990 if ( captureAudioClient ) {
4991 hr = captureAudioClient->GetMixFormat( &captureFormat );
4992 if ( FAILED( hr ) ) {
4993 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
4997 // init captureResampler
4998 captureResampler = new WasapiResampler( stream_.deviceFormat[INPUT] == RTAUDIO_FLOAT32 || stream_.deviceFormat[INPUT] == RTAUDIO_FLOAT64,
4999 formatBytes( stream_.deviceFormat[INPUT] ) * 8, stream_.nDeviceChannels[INPUT],
5000 captureFormat->nSamplesPerSec, stream_.sampleRate );
5002 captureSrRatio = ( ( float ) captureFormat->nSamplesPerSec / stream_.sampleRate );
5004 if ( !captureClient ) {
5005 hr = captureAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
5006 loopbackEnabled ? AUDCLNT_STREAMFLAGS_LOOPBACK : AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
5011 if ( FAILED( hr ) ) {
5012 errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize capture audio client.";
5016 hr = captureAudioClient->GetService( __uuidof( IAudioCaptureClient ),
5017 ( void** ) &captureClient );
5018 if ( FAILED( hr ) ) {
5019 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture client handle.";
5023 // don't configure captureEvent if in loopback mode
5024 if ( !loopbackEnabled )
5026 // configure captureEvent to trigger on every available capture buffer
5027 captureEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
5028 if ( !captureEvent ) {
5029 errorType = RtAudioError::SYSTEM_ERROR;
5030 errorText_ = "RtApiWasapi::wasapiThread: Unable to create capture event.";
5034 hr = captureAudioClient->SetEventHandle( captureEvent );
5035 if ( FAILED( hr ) ) {
5036 errorText_ = "RtApiWasapi::wasapiThread: Unable to set capture event handle.";
5040 ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent = captureEvent;
5043 ( ( WasapiHandle* ) stream_.apiHandle )->captureClient = captureClient;
5046 unsigned int inBufferSize = 0;
5047 hr = captureAudioClient->GetBufferSize( &inBufferSize );
5048 if ( FAILED( hr ) ) {
5049 errorText_ = "RtApiWasapi::wasapiThread: Unable to get capture buffer size.";
5053 // scale outBufferSize according to stream->user sample rate ratio
5054 unsigned int outBufferSize = ( unsigned int ) ceilf( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT];
5055 inBufferSize *= stream_.nDeviceChannels[INPUT];
5057 // set captureBuffer size
5058 captureBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[INPUT] ) );
5060 // reset the capture stream
5061 hr = captureAudioClient->Reset();
5062 if ( FAILED( hr ) ) {
5063 errorText_ = "RtApiWasapi::wasapiThread: Unable to reset capture stream.";
5067 // start the capture stream
5068 hr = captureAudioClient->Start();
5069 if ( FAILED( hr ) ) {
5070 errorText_ = "RtApiWasapi::wasapiThread: Unable to start capture stream.";
5075 // start render stream if applicable
5076 if ( renderAudioClient ) {
5077 hr = renderAudioClient->GetMixFormat( &renderFormat );
5078 if ( FAILED( hr ) ) {
5079 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
5083 // init renderResampler
5084 renderResampler = new WasapiResampler( stream_.deviceFormat[OUTPUT] == RTAUDIO_FLOAT32 || stream_.deviceFormat[OUTPUT] == RTAUDIO_FLOAT64,
5085 formatBytes( stream_.deviceFormat[OUTPUT] ) * 8, stream_.nDeviceChannels[OUTPUT],
5086 stream_.sampleRate, renderFormat->nSamplesPerSec );
5088 renderSrRatio = ( ( float ) renderFormat->nSamplesPerSec / stream_.sampleRate );
5090 if ( !renderClient ) {
5091 hr = renderAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
5092 AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
5097 if ( FAILED( hr ) ) {
5098 errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize render audio client.";
5102 hr = renderAudioClient->GetService( __uuidof( IAudioRenderClient ),
5103 ( void** ) &renderClient );
5104 if ( FAILED( hr ) ) {
5105 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render client handle.";
5109 // configure renderEvent to trigger on every available render buffer
5110 renderEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
5111 if ( !renderEvent ) {
5112 errorType = RtAudioError::SYSTEM_ERROR;
5113 errorText_ = "RtApiWasapi::wasapiThread: Unable to create render event.";
5117 hr = renderAudioClient->SetEventHandle( renderEvent );
5118 if ( FAILED( hr ) ) {
5119 errorText_ = "RtApiWasapi::wasapiThread: Unable to set render event handle.";
5123 ( ( WasapiHandle* ) stream_.apiHandle )->renderClient = renderClient;
5124 ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent = renderEvent;
5127 unsigned int outBufferSize = 0;
5128 hr = renderAudioClient->GetBufferSize( &outBufferSize );
5129 if ( FAILED( hr ) ) {
5130 errorText_ = "RtApiWasapi::wasapiThread: Unable to get render buffer size.";
5134 // scale inBufferSize according to user->stream sample rate ratio
5135 unsigned int inBufferSize = ( unsigned int ) ceilf( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT];
5136 outBufferSize *= stream_.nDeviceChannels[OUTPUT];
5138 // set renderBuffer size
5139 renderBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[OUTPUT] ) );
5141 // reset the render stream
5142 hr = renderAudioClient->Reset();
5143 if ( FAILED( hr ) ) {
5144 errorText_ = "RtApiWasapi::wasapiThread: Unable to reset render stream.";
5148 // start the render stream
5149 hr = renderAudioClient->Start();
5150 if ( FAILED( hr ) ) {
5151 errorText_ = "RtApiWasapi::wasapiThread: Unable to start render stream.";
5156 // malloc buffer memory
5157 if ( stream_.mode == INPUT )
5159 using namespace std; // for ceilf
5160 convBuffSize = ( size_t ) ( ceilf( stream_.bufferSize * captureSrRatio ) ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
5161 deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
5163 else if ( stream_.mode == OUTPUT )
5165 convBuffSize = ( size_t ) ( ceilf( stream_.bufferSize * renderSrRatio ) ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
5166 deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
5168 else if ( stream_.mode == DUPLEX )
5170 convBuffSize = std::max( ( size_t ) ( ceilf( stream_.bufferSize * captureSrRatio ) ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
5171 ( size_t ) ( ceilf( stream_.bufferSize * renderSrRatio ) ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
5172 deviceBuffSize = std::max( stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
5173 stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
5176 convBuffSize *= 2; // allow overflow for *SrRatio remainders
5177 convBuffer = ( char* ) malloc( convBuffSize );
5178 stream_.deviceBuffer = ( char* ) malloc( deviceBuffSize );
5179 if ( !convBuffer || !stream_.deviceBuffer ) {
5180 errorType = RtAudioError::MEMORY_ERROR;
5181 errorText_ = "RtApiWasapi::wasapiThread: Error allocating device buffer memory.";
5185 // stream process loop
5186 while ( stream_.state != STREAM_STOPPING ) {
5187 if ( !callbackPulled ) {
5190 // 1. Pull callback buffer from inputBuffer
5191 // 2. If 1. was successful: Convert callback buffer to user sample rate and channel count
5192 // Convert callback buffer to user format
5194 if ( captureAudioClient )
5196 int samplesToPull = ( unsigned int ) floorf( stream_.bufferSize * captureSrRatio );
5197 if ( captureSrRatio != 1 )
5199 // account for remainders
5204 while ( convBufferSize < stream_.bufferSize )
5206 // Pull callback buffer from inputBuffer
5207 callbackPulled = captureBuffer.pullBuffer( convBuffer,
5208 samplesToPull * stream_.nDeviceChannels[INPUT],
5209 stream_.deviceFormat[INPUT] );
5211 if ( !callbackPulled )
5216 // Convert callback buffer to user sample rate
5217 unsigned int deviceBufferOffset = convBufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
5218 unsigned int convSamples = 0;
5220 captureResampler->Convert( stream_.deviceBuffer + deviceBufferOffset,
5225 convBufferSize += convSamples;
5226 samplesToPull = 1; // now pull one sample at a time until we have stream_.bufferSize samples
5229 if ( callbackPulled )
5231 if ( stream_.doConvertBuffer[INPUT] ) {
5232 // Convert callback buffer to user format
5233 convertBuffer( stream_.userBuffer[INPUT],
5234 stream_.deviceBuffer,
5235 stream_.convertInfo[INPUT] );
5238 // no further conversion, simple copy deviceBuffer to userBuffer
5239 memcpy( stream_.userBuffer[INPUT],
5240 stream_.deviceBuffer,
5241 stream_.bufferSize * stream_.nUserChannels[INPUT] * formatBytes( stream_.userFormat ) );
5246 // if there is no capture stream, set callbackPulled flag
5247 callbackPulled = true;
5252 // 1. Execute user callback method
5253 // 2. Handle return value from callback
5255 // if callback has not requested the stream to stop
5256 if ( callbackPulled && !callbackStopped ) {
5257 // Execute user callback method
5258 callbackResult = callback( stream_.userBuffer[OUTPUT],
5259 stream_.userBuffer[INPUT],
5262 captureFlags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY ? RTAUDIO_INPUT_OVERFLOW : 0,
5263 stream_.callbackInfo.userData );
5265 // Handle return value from callback
5266 if ( callbackResult == 1 ) {
5267 // instantiate a thread to stop this thread
5268 HANDLE threadHandle = CreateThread( NULL, 0, stopWasapiThread, this, 0, NULL );
5269 if ( !threadHandle ) {
5270 errorType = RtAudioError::THREAD_ERROR;
5271 errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream stop thread.";
5274 else if ( !CloseHandle( threadHandle ) ) {
5275 errorType = RtAudioError::THREAD_ERROR;
5276 errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream stop thread handle.";
5280 callbackStopped = true;
5282 else if ( callbackResult == 2 ) {
5283 // instantiate a thread to stop this thread
5284 HANDLE threadHandle = CreateThread( NULL, 0, abortWasapiThread, this, 0, NULL );
5285 if ( !threadHandle ) {
5286 errorType = RtAudioError::THREAD_ERROR;
5287 errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream abort thread.";
5290 else if ( !CloseHandle( threadHandle ) ) {
5291 errorType = RtAudioError::THREAD_ERROR;
5292 errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream abort thread handle.";
5296 callbackStopped = true;
5303 // 1. Convert callback buffer to stream format
5304 // 2. Convert callback buffer to stream sample rate and channel count
5305 // 3. Push callback buffer into outputBuffer
5307 if ( renderAudioClient && callbackPulled )
5309 // if the last call to renderBuffer.PushBuffer() was successful
5310 if ( callbackPushed || convBufferSize == 0 )
5312 if ( stream_.doConvertBuffer[OUTPUT] )
5314 // Convert callback buffer to stream format
5315 convertBuffer( stream_.deviceBuffer,
5316 stream_.userBuffer[OUTPUT],
5317 stream_.convertInfo[OUTPUT] );
5321 // Convert callback buffer to stream sample rate
5322 renderResampler->Convert( convBuffer,
5323 stream_.deviceBuffer,
5328 // Push callback buffer into outputBuffer
5329 callbackPushed = renderBuffer.pushBuffer( convBuffer,
5330 convBufferSize * stream_.nDeviceChannels[OUTPUT],
5331 stream_.deviceFormat[OUTPUT] );
5334 // if there is no render stream, set callbackPushed flag
5335 callbackPushed = true;
5340 // 1. Get capture buffer from stream
5341 // 2. Push capture buffer into inputBuffer
5342 // 3. If 2. was successful: Release capture buffer
5344 if ( captureAudioClient ) {
5345 // if the callback input buffer was not pulled from captureBuffer, wait for next capture event
5346 if ( !callbackPulled ) {
5347 WaitForSingleObject( loopbackEnabled ? renderEvent : captureEvent, INFINITE );
5350 // Get capture buffer from stream
5351 hr = captureClient->GetBuffer( &streamBuffer,
5353 &captureFlags, NULL, NULL );
5354 if ( FAILED( hr ) ) {
5355 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture buffer.";
5359 if ( bufferFrameCount != 0 ) {
5360 // Push capture buffer into inputBuffer
5361 if ( captureBuffer.pushBuffer( ( char* ) streamBuffer,
5362 bufferFrameCount * stream_.nDeviceChannels[INPUT],
5363 stream_.deviceFormat[INPUT] ) )
5365 // Release capture buffer
5366 hr = captureClient->ReleaseBuffer( bufferFrameCount );
5367 if ( FAILED( hr ) ) {
5368 errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
5374 // Inform WASAPI that capture was unsuccessful
5375 hr = captureClient->ReleaseBuffer( 0 );
5376 if ( FAILED( hr ) ) {
5377 errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
5384 // Inform WASAPI that capture was unsuccessful
5385 hr = captureClient->ReleaseBuffer( 0 );
5386 if ( FAILED( hr ) ) {
5387 errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
5395 // 1. Get render buffer from stream
5396 // 2. Pull next buffer from outputBuffer
5397 // 3. If 2. was successful: Fill render buffer with next buffer
5398 // Release render buffer
5400 if ( renderAudioClient ) {
5401 // if the callback output buffer was not pushed to renderBuffer, wait for next render event
5402 if ( callbackPulled && !callbackPushed ) {
5403 WaitForSingleObject( renderEvent, INFINITE );
5406 // Get render buffer from stream
5407 hr = renderAudioClient->GetBufferSize( &bufferFrameCount );
5408 if ( FAILED( hr ) ) {
5409 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer size.";
5413 hr = renderAudioClient->GetCurrentPadding( &numFramesPadding );
5414 if ( FAILED( hr ) ) {
5415 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer padding.";
5419 bufferFrameCount -= numFramesPadding;
5421 if ( bufferFrameCount != 0 ) {
5422 hr = renderClient->GetBuffer( bufferFrameCount, &streamBuffer );
5423 if ( FAILED( hr ) ) {
5424 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer.";
5428 // Pull next buffer from outputBuffer
5429 // Fill render buffer with next buffer
5430 if ( renderBuffer.pullBuffer( ( char* ) streamBuffer,
5431 bufferFrameCount * stream_.nDeviceChannels[OUTPUT],
5432 stream_.deviceFormat[OUTPUT] ) )
5434 // Release render buffer
5435 hr = renderClient->ReleaseBuffer( bufferFrameCount, 0 );
5436 if ( FAILED( hr ) ) {
5437 errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
5443 // Inform WASAPI that render was unsuccessful
5444 hr = renderClient->ReleaseBuffer( 0, 0 );
5445 if ( FAILED( hr ) ) {
5446 errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
5453 // Inform WASAPI that render was unsuccessful
5454 hr = renderClient->ReleaseBuffer( 0, 0 );
5455 if ( FAILED( hr ) ) {
5456 errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
5462 // if the callback buffer was pushed renderBuffer reset callbackPulled flag
5463 if ( callbackPushed ) {
5464 // unsetting the callbackPulled flag lets the stream know that
5465 // the audio device is ready for another callback output buffer.
5466 callbackPulled = false;
5469 RtApi::tickStreamTime();
5476 CoTaskMemFree( captureFormat );
5477 CoTaskMemFree( renderFormat );
5479 free ( convBuffer );
5480 delete renderResampler;
5481 delete captureResampler;
5485 if ( !errorText_.empty() )
5488 // update stream state
5489 stream_.state = STREAM_STOPPED;
5492 //******************** End of __WINDOWS_WASAPI__ *********************//
5496 #if defined(__WINDOWS_DS__) // Windows DirectSound API
5498 // Modified by Robin Davies, October 2005
5499 // - Improvements to DirectX pointer chasing.
5500 // - Bug fix for non-power-of-two Asio granularity used by Edirol PCR-A30.
5501 // - Auto-call CoInitialize for DSOUND and ASIO platforms.
5502 // Various revisions for RtAudio 4.0 by Gary Scavone, April 2007
5503 // Changed device query structure for RtAudio 4.0.7, January 2010
5505 #include <windows.h>
5506 #include <process.h>
5507 #include <mmsystem.h>
5511 #include <algorithm>
5513 #if defined(__MINGW32__)
5514 // missing from latest mingw winapi
5515 #define WAVE_FORMAT_96M08 0x00010000 /* 96 kHz, Mono, 8-bit */
5516 #define WAVE_FORMAT_96S08 0x00020000 /* 96 kHz, Stereo, 8-bit */
5517 #define WAVE_FORMAT_96M16 0x00040000 /* 96 kHz, Mono, 16-bit */
5518 #define WAVE_FORMAT_96S16 0x00080000 /* 96 kHz, Stereo, 16-bit */
5521 #define MINIMUM_DEVICE_BUFFER_SIZE 32768
5523 #ifdef _MSC_VER // if Microsoft Visual C++
5524 #pragma comment( lib, "winmm.lib" ) // then, auto-link winmm.lib. Otherwise, it has to be added manually.
5527 static inline DWORD dsPointerBetween( DWORD pointer, DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize )
5529 if ( pointer > bufferSize ) pointer -= bufferSize;
5530 if ( laterPointer < earlierPointer ) laterPointer += bufferSize;
5531 if ( pointer < earlierPointer ) pointer += bufferSize;
5532 return pointer >= earlierPointer && pointer < laterPointer;
5535 // A structure to hold various information related to the DirectSound
5536 // API implementation.
5538 unsigned int drainCounter; // Tracks callback counts when draining
5539 bool internalDrain; // Indicates if stop is initiated from callback or not.
5543 UINT bufferPointer[2];
5544 DWORD dsBufferSize[2];
5545 DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by.
5549 :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; }
5552 // Declarations for utility functions, callbacks, and structures
5553 // specific to the DirectSound implementation.
5554 static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
5555 LPCTSTR description,
5559 static const char* getErrorString( int code );
5561 static unsigned __stdcall callbackHandler( void *ptr );
5570 : found(false) { validId[0] = false; validId[1] = false; }
5573 struct DsProbeData {
5575 std::vector<struct DsDevice>* dsDevices;
5578 RtApiDs :: RtApiDs()
5580 // Dsound will run both-threaded. If CoInitialize fails, then just
5581 // accept whatever the mainline chose for a threading model.
5582 coInitialized_ = false;
5583 HRESULT hr = CoInitialize( NULL );
5584 if ( !FAILED( hr ) ) coInitialized_ = true;
5587 RtApiDs :: ~RtApiDs()
5589 if ( stream_.state != STREAM_CLOSED ) closeStream();
5590 if ( coInitialized_ ) CoUninitialize(); // balanced call.
5593 // The DirectSound default output is always the first device.
5594 unsigned int RtApiDs :: getDefaultOutputDevice( void )
5599 // The DirectSound default input is always the first input device,
5600 // which is the first capture device enumerated.
5601 unsigned int RtApiDs :: getDefaultInputDevice( void )
5606 unsigned int RtApiDs :: getDeviceCount( void )
5608 // Set query flag for previously found devices to false, so that we
5609 // can check for any devices that have disappeared.
5610 for ( unsigned int i=0; i<dsDevices.size(); i++ )
5611 dsDevices[i].found = false;
5613 // Query DirectSound devices.
5614 struct DsProbeData probeInfo;
5615 probeInfo.isInput = false;
5616 probeInfo.dsDevices = &dsDevices;
5617 HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );
5618 if ( FAILED( result ) ) {
5619 errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!";
5620 errorText_ = errorStream_.str();
5621 error( RtAudioError::WARNING );
5624 // Query DirectSoundCapture devices.
5625 probeInfo.isInput = true;
5626 result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );
5627 if ( FAILED( result ) ) {
5628 errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!";
5629 errorText_ = errorStream_.str();
5630 error( RtAudioError::WARNING );
5633 // Clean out any devices that may have disappeared (code update submitted by Eli Zehngut).
5634 for ( unsigned int i=0; i<dsDevices.size(); ) {
5635 if ( dsDevices[i].found == false ) dsDevices.erase( dsDevices.begin() + i );
5639 return static_cast<unsigned int>(dsDevices.size());
5642 RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device )
5644 RtAudio::DeviceInfo info;
5645 info.probed = false;
5647 if ( dsDevices.size() == 0 ) {
5648 // Force a query of all devices
5650 if ( dsDevices.size() == 0 ) {
5651 errorText_ = "RtApiDs::getDeviceInfo: no devices found!";
5652 error( RtAudioError::INVALID_USE );
5657 if ( device >= dsDevices.size() ) {
5658 errorText_ = "RtApiDs::getDeviceInfo: device ID is invalid!";
5659 error( RtAudioError::INVALID_USE );
5664 if ( dsDevices[ device ].validId[0] == false ) goto probeInput;
5666 LPDIRECTSOUND output;
5668 result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );
5669 if ( FAILED( result ) ) {
5670 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!";
5671 errorText_ = errorStream_.str();
5672 error( RtAudioError::WARNING );
5676 outCaps.dwSize = sizeof( outCaps );
5677 result = output->GetCaps( &outCaps );
5678 if ( FAILED( result ) ) {
5680 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting capabilities!";
5681 errorText_ = errorStream_.str();
5682 error( RtAudioError::WARNING );
5686 // Get output channel information.
5687 info.outputChannels = ( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1;
5689 // Get sample rate information.
5690 info.sampleRates.clear();
5691 for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
5692 if ( SAMPLE_RATES[k] >= (unsigned int) outCaps.dwMinSecondarySampleRate &&
5693 SAMPLE_RATES[k] <= (unsigned int) outCaps.dwMaxSecondarySampleRate ) {
5694 info.sampleRates.push_back( SAMPLE_RATES[k] );
5696 if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
5697 info.preferredSampleRate = SAMPLE_RATES[k];
5701 // Get format information.
5702 if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT ) info.nativeFormats |= RTAUDIO_SINT16;
5703 if ( outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) info.nativeFormats |= RTAUDIO_SINT8;
5707 if ( getDefaultOutputDevice() == device )
5708 info.isDefaultOutput = true;
5710 if ( dsDevices[ device ].validId[1] == false ) {
5711 info.name = dsDevices[ device ].name;
5718 LPDIRECTSOUNDCAPTURE input;
5719 result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );
5720 if ( FAILED( result ) ) {
5721 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";
5722 errorText_ = errorStream_.str();
5723 error( RtAudioError::WARNING );
5728 inCaps.dwSize = sizeof( inCaps );
5729 result = input->GetCaps( &inCaps );
5730 if ( FAILED( result ) ) {
5732 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsDevices[ device ].name << ")!";
5733 errorText_ = errorStream_.str();
5734 error( RtAudioError::WARNING );
5738 // Get input channel information.
5739 info.inputChannels = inCaps.dwChannels;
5741 // Get sample rate and format information.
5742 std::vector<unsigned int> rates;
5743 if ( inCaps.dwChannels >= 2 ) {
5744 if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.nativeFormats |= RTAUDIO_SINT16;
5745 if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.nativeFormats |= RTAUDIO_SINT16;
5746 if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.nativeFormats |= RTAUDIO_SINT16;
5747 if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.nativeFormats |= RTAUDIO_SINT16;
5748 if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.nativeFormats |= RTAUDIO_SINT8;
5749 if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.nativeFormats |= RTAUDIO_SINT8;
5750 if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.nativeFormats |= RTAUDIO_SINT8;
5751 if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.nativeFormats |= RTAUDIO_SINT8;
5753 if ( info.nativeFormats & RTAUDIO_SINT16 ) {
5754 if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) rates.push_back( 11025 );
5755 if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) rates.push_back( 22050 );
5756 if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) rates.push_back( 44100 );
5757 if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) rates.push_back( 96000 );
5759 else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
5760 if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) rates.push_back( 11025 );
5761 if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) rates.push_back( 22050 );
5762 if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) rates.push_back( 44100 );
5763 if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) rates.push_back( 96000 );
5766 else if ( inCaps.dwChannels == 1 ) {
5767 if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.nativeFormats |= RTAUDIO_SINT16;
5768 if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.nativeFormats |= RTAUDIO_SINT16;
5769 if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.nativeFormats |= RTAUDIO_SINT16;
5770 if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.nativeFormats |= RTAUDIO_SINT16;
5771 if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.nativeFormats |= RTAUDIO_SINT8;
5772 if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.nativeFormats |= RTAUDIO_SINT8;
5773 if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.nativeFormats |= RTAUDIO_SINT8;
5774 if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.nativeFormats |= RTAUDIO_SINT8;
5776 if ( info.nativeFormats & RTAUDIO_SINT16 ) {
5777 if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) rates.push_back( 11025 );
5778 if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) rates.push_back( 22050 );
5779 if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) rates.push_back( 44100 );
5780 if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) rates.push_back( 96000 );
5782 else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
5783 if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) rates.push_back( 11025 );
5784 if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) rates.push_back( 22050 );
5785 if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) rates.push_back( 44100 );
5786 if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) rates.push_back( 96000 );
5789 else info.inputChannels = 0; // technically, this would be an error
5793 if ( info.inputChannels == 0 ) return info;
5795 // Copy the supported rates to the info structure but avoid duplication.
5797 for ( unsigned int i=0; i<rates.size(); i++ ) {
5799 for ( unsigned int j=0; j<info.sampleRates.size(); j++ ) {
5800 if ( rates[i] == info.sampleRates[j] ) {
5805 if ( found == false ) info.sampleRates.push_back( rates[i] );
5807 std::sort( info.sampleRates.begin(), info.sampleRates.end() );
5809 // If device opens for both playback and capture, we determine the channels.
5810 if ( info.outputChannels > 0 && info.inputChannels > 0 )
5811 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
5813 if ( device == 0 ) info.isDefaultInput = true;
5815 // Copy name and return.
5816 info.name = dsDevices[ device ].name;
5821 bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
5822 unsigned int firstChannel, unsigned int sampleRate,
5823 RtAudioFormat format, unsigned int *bufferSize,
5824 RtAudio::StreamOptions *options )
5826 if ( channels + firstChannel > 2 ) {
5827 errorText_ = "RtApiDs::probeDeviceOpen: DirectSound does not support more than 2 channels per device.";
5831 size_t nDevices = dsDevices.size();
5832 if ( nDevices == 0 ) {
5833 // This should not happen because a check is made before this function is called.
5834 errorText_ = "RtApiDs::probeDeviceOpen: no devices found!";
5838 if ( device >= nDevices ) {
5839 // This should not happen because a check is made before this function is called.
5840 errorText_ = "RtApiDs::probeDeviceOpen: device ID is invalid!";
5844 if ( mode == OUTPUT ) {
5845 if ( dsDevices[ device ].validId[0] == false ) {
5846 errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support output!";
5847 errorText_ = errorStream_.str();
5851 else { // mode == INPUT
5852 if ( dsDevices[ device ].validId[1] == false ) {
5853 errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support input!";
5854 errorText_ = errorStream_.str();
5859 // According to a note in PortAudio, using GetDesktopWindow()
5860 // instead of GetForegroundWindow() is supposed to avoid problems
5861 // that occur when the application's window is not the foreground
5862 // window. Also, if the application window closes before the
5863 // DirectSound buffer, DirectSound can crash. In the past, I had
5864 // problems when using GetDesktopWindow() but it seems fine now
5865 // (January 2010). I'll leave it commented here.
5866 // HWND hWnd = GetForegroundWindow();
5867 HWND hWnd = GetDesktopWindow();
5869 // Check the numberOfBuffers parameter and limit the lowest value to
5870 // two. This is a judgement call and a value of two is probably too
5871 // low for capture, but it should work for playback.
5873 if ( options ) nBuffers = options->numberOfBuffers;
5874 if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) nBuffers = 2;
5875 if ( nBuffers < 2 ) nBuffers = 3;
5877 // Check the lower range of the user-specified buffer size and set
5878 // (arbitrarily) to a lower bound of 32.
5879 if ( *bufferSize < 32 ) *bufferSize = 32;
5881 // Create the wave format structure. The data format setting will
5882 // be determined later.
5883 WAVEFORMATEX waveFormat;
5884 ZeroMemory( &waveFormat, sizeof(WAVEFORMATEX) );
5885 waveFormat.wFormatTag = WAVE_FORMAT_PCM;
5886 waveFormat.nChannels = channels + firstChannel;
5887 waveFormat.nSamplesPerSec = (unsigned long) sampleRate;
5889 // Determine the device buffer size. By default, we'll use the value
5890 // defined above (32K), but we will grow it to make allowances for
5891 // very large software buffer sizes.
5892 DWORD dsBufferSize = MINIMUM_DEVICE_BUFFER_SIZE;
5893 DWORD dsPointerLeadTime = 0;
5895 void *ohandle = 0, *bhandle = 0;
5897 if ( mode == OUTPUT ) {
5899 LPDIRECTSOUND output;
5900 result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );
5901 if ( FAILED( result ) ) {
5902 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!";
5903 errorText_ = errorStream_.str();
5908 outCaps.dwSize = sizeof( outCaps );
5909 result = output->GetCaps( &outCaps );
5910 if ( FAILED( result ) ) {
5912 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting capabilities (" << dsDevices[ device ].name << ")!";
5913 errorText_ = errorStream_.str();
5917 // Check channel information.
5918 if ( channels + firstChannel == 2 && !( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ) {
5919 errorStream_ << "RtApiDs::getDeviceInfo: the output device (" << dsDevices[ device ].name << ") does not support stereo playback.";
5920 errorText_ = errorStream_.str();
5924 // Check format information. Use 16-bit format unless not
5925 // supported or user requests 8-bit.
5926 if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT &&
5927 !( format == RTAUDIO_SINT8 && outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) ) {
5928 waveFormat.wBitsPerSample = 16;
5929 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
5932 waveFormat.wBitsPerSample = 8;
5933 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
5935 stream_.userFormat = format;
5937 // Update wave format structure and buffer information.
5938 waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
5939 waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
5940 dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
5942 // If the user wants an even bigger buffer, increase the device buffer size accordingly.
5943 while ( dsPointerLeadTime * 2U > dsBufferSize )
5946 // Set cooperative level to DSSCL_EXCLUSIVE ... sound stops when window focus changes.
5947 // result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE );
5948 // Set cooperative level to DSSCL_PRIORITY ... sound remains when window focus changes.
5949 result = output->SetCooperativeLevel( hWnd, DSSCL_PRIORITY );
5950 if ( FAILED( result ) ) {
5952 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting cooperative level (" << dsDevices[ device ].name << ")!";
5953 errorText_ = errorStream_.str();
5957 // Even though we will write to the secondary buffer, we need to
5958 // access the primary buffer to set the correct output format
5959 // (since the default is 8-bit, 22 kHz!). Setup the DS primary
5960 // buffer description.
5961 DSBUFFERDESC bufferDescription;
5962 ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
5963 bufferDescription.dwSize = sizeof( DSBUFFERDESC );
5964 bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER;
5966 // Obtain the primary buffer
5967 LPDIRECTSOUNDBUFFER buffer;
5968 result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
5969 if ( FAILED( result ) ) {
5971 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") accessing primary buffer (" << dsDevices[ device ].name << ")!";
5972 errorText_ = errorStream_.str();
5976 // Set the primary DS buffer sound format.
5977 result = buffer->SetFormat( &waveFormat );
5978 if ( FAILED( result ) ) {
5980 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting primary buffer format (" << dsDevices[ device ].name << ")!";
5981 errorText_ = errorStream_.str();
5985 // Setup the secondary DS buffer description.
5986 ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
5987 bufferDescription.dwSize = sizeof( DSBUFFERDESC );
5988 bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
5989 DSBCAPS_GLOBALFOCUS |
5990 DSBCAPS_GETCURRENTPOSITION2 |
5991 DSBCAPS_LOCHARDWARE ); // Force hardware mixing
5992 bufferDescription.dwBufferBytes = dsBufferSize;
5993 bufferDescription.lpwfxFormat = &waveFormat;
5995 // Try to create the secondary DS buffer. If that doesn't work,
5996 // try to use software mixing. Otherwise, there's a problem.
5997 result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
5998 if ( FAILED( result ) ) {
5999 bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
6000 DSBCAPS_GLOBALFOCUS |
6001 DSBCAPS_GETCURRENTPOSITION2 |
6002 DSBCAPS_LOCSOFTWARE ); // Force software mixing
6003 result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
6004 if ( FAILED( result ) ) {
6006 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating secondary buffer (" << dsDevices[ device ].name << ")!";
6007 errorText_ = errorStream_.str();
6012 // Get the buffer size ... might be different from what we specified.
6014 dsbcaps.dwSize = sizeof( DSBCAPS );
6015 result = buffer->GetCaps( &dsbcaps );
6016 if ( FAILED( result ) ) {
6019 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";
6020 errorText_ = errorStream_.str();
6024 dsBufferSize = dsbcaps.dwBufferBytes;
6026 // Lock the DS buffer
6029 result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );
6030 if ( FAILED( result ) ) {
6033 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking buffer (" << dsDevices[ device ].name << ")!";
6034 errorText_ = errorStream_.str();
6038 // Zero the DS buffer
6039 ZeroMemory( audioPtr, dataLen );
6041 // Unlock the DS buffer
6042 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
6043 if ( FAILED( result ) ) {
6046 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking buffer (" << dsDevices[ device ].name << ")!";
6047 errorText_ = errorStream_.str();
6051 ohandle = (void *) output;
6052 bhandle = (void *) buffer;
6055 if ( mode == INPUT ) {
6057 LPDIRECTSOUNDCAPTURE input;
6058 result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );
6059 if ( FAILED( result ) ) {
6060 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";
6061 errorText_ = errorStream_.str();
6066 inCaps.dwSize = sizeof( inCaps );
6067 result = input->GetCaps( &inCaps );
6068 if ( FAILED( result ) ) {
6070 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsDevices[ device ].name << ")!";
6071 errorText_ = errorStream_.str();
6075 // Check channel information.
6076 if ( inCaps.dwChannels < channels + firstChannel ) {
6077 errorText_ = "RtApiDs::getDeviceInfo: the input device does not support requested input channels.";
6081 // Check format information. Use 16-bit format unless user
6083 DWORD deviceFormats;
6084 if ( channels + firstChannel == 2 ) {
6085 deviceFormats = WAVE_FORMAT_1S08 | WAVE_FORMAT_2S08 | WAVE_FORMAT_4S08 | WAVE_FORMAT_96S08;
6086 if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
6087 waveFormat.wBitsPerSample = 8;
6088 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
6090 else { // assume 16-bit is supported
6091 waveFormat.wBitsPerSample = 16;
6092 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
6095 else { // channel == 1
6096 deviceFormats = WAVE_FORMAT_1M08 | WAVE_FORMAT_2M08 | WAVE_FORMAT_4M08 | WAVE_FORMAT_96M08;
6097 if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
6098 waveFormat.wBitsPerSample = 8;
6099 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
6101 else { // assume 16-bit is supported
6102 waveFormat.wBitsPerSample = 16;
6103 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
6106 stream_.userFormat = format;
6108 // Update wave format structure and buffer information.
6109 waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
6110 waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
6111 dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
6113 // If the user wants an even bigger buffer, increase the device buffer size accordingly.
6114 while ( dsPointerLeadTime * 2U > dsBufferSize )
6117 // Setup the secondary DS buffer description.
6118 DSCBUFFERDESC bufferDescription;
6119 ZeroMemory( &bufferDescription, sizeof( DSCBUFFERDESC ) );
6120 bufferDescription.dwSize = sizeof( DSCBUFFERDESC );
6121 bufferDescription.dwFlags = 0;
6122 bufferDescription.dwReserved = 0;
6123 bufferDescription.dwBufferBytes = dsBufferSize;
6124 bufferDescription.lpwfxFormat = &waveFormat;
6126 // Create the capture buffer.
6127 LPDIRECTSOUNDCAPTUREBUFFER buffer;
6128 result = input->CreateCaptureBuffer( &bufferDescription, &buffer, NULL );
6129 if ( FAILED( result ) ) {
6131 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating input buffer (" << dsDevices[ device ].name << ")!";
6132 errorText_ = errorStream_.str();
6136 // Get the buffer size ... might be different from what we specified.
6138 dscbcaps.dwSize = sizeof( DSCBCAPS );
6139 result = buffer->GetCaps( &dscbcaps );
6140 if ( FAILED( result ) ) {
6143 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";
6144 errorText_ = errorStream_.str();
6148 dsBufferSize = dscbcaps.dwBufferBytes;
6150 // NOTE: We could have a problem here if this is a duplex stream
6151 // and the play and capture hardware buffer sizes are different
6152 // (I'm actually not sure if that is a problem or not).
6153 // Currently, we are not verifying that.
6155 // Lock the capture buffer
6158 result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );
6159 if ( FAILED( result ) ) {
6162 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking input buffer (" << dsDevices[ device ].name << ")!";
6163 errorText_ = errorStream_.str();
6168 ZeroMemory( audioPtr, dataLen );
6170 // Unlock the buffer
6171 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
6172 if ( FAILED( result ) ) {
6175 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking input buffer (" << dsDevices[ device ].name << ")!";
6176 errorText_ = errorStream_.str();
6180 ohandle = (void *) input;
6181 bhandle = (void *) buffer;
6184 // Set various stream parameters
6185 DsHandle *handle = 0;
6186 stream_.nDeviceChannels[mode] = channels + firstChannel;
6187 stream_.nUserChannels[mode] = channels;
6188 stream_.bufferSize = *bufferSize;
6189 stream_.channelOffset[mode] = firstChannel;
6190 stream_.deviceInterleaved[mode] = true;
6191 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
6192 else stream_.userInterleaved = true;
6194 // Set flag for buffer conversion
6195 stream_.doConvertBuffer[mode] = false;
6196 if (stream_.nUserChannels[mode] != stream_.nDeviceChannels[mode])
6197 stream_.doConvertBuffer[mode] = true;
6198 if (stream_.userFormat != stream_.deviceFormat[mode])
6199 stream_.doConvertBuffer[mode] = true;
6200 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
6201 stream_.nUserChannels[mode] > 1 )
6202 stream_.doConvertBuffer[mode] = true;
6204 // Allocate necessary internal buffers
6205 long bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
6206 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
6207 if ( stream_.userBuffer[mode] == NULL ) {
6208 errorText_ = "RtApiDs::probeDeviceOpen: error allocating user buffer memory.";
6212 if ( stream_.doConvertBuffer[mode] ) {
6214 bool makeBuffer = true;
6215 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
6216 if ( mode == INPUT ) {
6217 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
6218 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
6219 if ( bufferBytes <= (long) bytesOut ) makeBuffer = false;
6224 bufferBytes *= *bufferSize;
6225 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
6226 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
6227 if ( stream_.deviceBuffer == NULL ) {
6228 errorText_ = "RtApiDs::probeDeviceOpen: error allocating device buffer memory.";
6234 // Allocate our DsHandle structures for the stream.
6235 if ( stream_.apiHandle == 0 ) {
6237 handle = new DsHandle;
6239 catch ( std::bad_alloc& ) {
6240 errorText_ = "RtApiDs::probeDeviceOpen: error allocating AsioHandle memory.";
6244 // Create a manual-reset event.
6245 handle->condition = CreateEvent( NULL, // no security
6246 TRUE, // manual-reset
6247 FALSE, // non-signaled initially
6249 stream_.apiHandle = (void *) handle;
6252 handle = (DsHandle *) stream_.apiHandle;
6253 handle->id[mode] = ohandle;
6254 handle->buffer[mode] = bhandle;
6255 handle->dsBufferSize[mode] = dsBufferSize;
6256 handle->dsPointerLeadTime[mode] = dsPointerLeadTime;
6258 stream_.device[mode] = device;
6259 stream_.state = STREAM_STOPPED;
6260 if ( stream_.mode == OUTPUT && mode == INPUT )
6261 // We had already set up an output stream.
6262 stream_.mode = DUPLEX;
6264 stream_.mode = mode;
6265 stream_.nBuffers = nBuffers;
6266 stream_.sampleRate = sampleRate;
6268 // Setup the buffer conversion information structure.
6269 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
6271 // Setup the callback thread.
6272 if ( stream_.callbackInfo.isRunning == false ) {
6274 stream_.callbackInfo.isRunning = true;
6275 stream_.callbackInfo.object = (void *) this;
6276 stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &callbackHandler,
6277 &stream_.callbackInfo, 0, &threadId );
6278 if ( stream_.callbackInfo.thread == 0 ) {
6279 errorText_ = "RtApiDs::probeDeviceOpen: error creating callback thread!";
6283 // Boost DS thread priority
6284 SetThreadPriority( (HANDLE) stream_.callbackInfo.thread, THREAD_PRIORITY_HIGHEST );
6290 if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
6291 LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
6292 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6293 if ( buffer ) buffer->Release();
6296 if ( handle->buffer[1] ) {
6297 LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
6298 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6299 if ( buffer ) buffer->Release();
6302 CloseHandle( handle->condition );
6304 stream_.apiHandle = 0;
6307 for ( int i=0; i<2; i++ ) {
6308 if ( stream_.userBuffer[i] ) {
6309 free( stream_.userBuffer[i] );
6310 stream_.userBuffer[i] = 0;
6314 if ( stream_.deviceBuffer ) {
6315 free( stream_.deviceBuffer );
6316 stream_.deviceBuffer = 0;
6319 stream_.state = STREAM_CLOSED;
6323 void RtApiDs :: closeStream()
6325 if ( stream_.state == STREAM_CLOSED ) {
6326 errorText_ = "RtApiDs::closeStream(): no open stream to close!";
6327 error( RtAudioError::WARNING );
6331 // Stop the callback thread.
6332 stream_.callbackInfo.isRunning = false;
6333 WaitForSingleObject( (HANDLE) stream_.callbackInfo.thread, INFINITE );
6334 CloseHandle( (HANDLE) stream_.callbackInfo.thread );
6336 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6338 if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
6339 LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
6340 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6347 if ( handle->buffer[1] ) {
6348 LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
6349 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6356 CloseHandle( handle->condition );
6358 stream_.apiHandle = 0;
6361 for ( int i=0; i<2; i++ ) {
6362 if ( stream_.userBuffer[i] ) {
6363 free( stream_.userBuffer[i] );
6364 stream_.userBuffer[i] = 0;
6368 if ( stream_.deviceBuffer ) {
6369 free( stream_.deviceBuffer );
6370 stream_.deviceBuffer = 0;
6373 stream_.mode = UNINITIALIZED;
6374 stream_.state = STREAM_CLOSED;
6377 void RtApiDs :: startStream()
6380 if ( stream_.state == STREAM_RUNNING ) {
6381 errorText_ = "RtApiDs::startStream(): the stream is already running!";
6382 error( RtAudioError::WARNING );
6386 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6388 // Increase scheduler frequency on lesser windows (a side-effect of
6389 // increasing timer accuracy). On greater windows (Win2K or later),
6390 // this is already in effect.
6391 timeBeginPeriod( 1 );
6393 buffersRolling = false;
6394 duplexPrerollBytes = 0;
6396 if ( stream_.mode == DUPLEX ) {
6397 // 0.5 seconds of silence in DUPLEX mode while the devices spin up and synchronize.
6398 duplexPrerollBytes = (int) ( 0.5 * stream_.sampleRate * formatBytes( stream_.deviceFormat[1] ) * stream_.nDeviceChannels[1] );
6402 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
6404 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6405 result = buffer->Play( 0, 0, DSBPLAY_LOOPING );
6406 if ( FAILED( result ) ) {
6407 errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting output buffer!";
6408 errorText_ = errorStream_.str();
6413 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
6415 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6416 result = buffer->Start( DSCBSTART_LOOPING );
6417 if ( FAILED( result ) ) {
6418 errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting input buffer!";
6419 errorText_ = errorStream_.str();
6424 handle->drainCounter = 0;
6425 handle->internalDrain = false;
6426 ResetEvent( handle->condition );
6427 stream_.state = STREAM_RUNNING;
6430 if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );
6433 void RtApiDs :: stopStream()
6436 if ( stream_.state == STREAM_STOPPED ) {
6437 errorText_ = "RtApiDs::stopStream(): the stream is already stopped!";
6438 error( RtAudioError::WARNING );
6445 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6446 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
6447 if ( handle->drainCounter == 0 ) {
6448 handle->drainCounter = 2;
6449 WaitForSingleObject( handle->condition, INFINITE ); // block until signaled
6452 stream_.state = STREAM_STOPPED;
6454 MUTEX_LOCK( &stream_.mutex );
6456 // Stop the buffer and clear memory
6457 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6458 result = buffer->Stop();
6459 if ( FAILED( result ) ) {
6460 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping output buffer!";
6461 errorText_ = errorStream_.str();
6465 // Lock the buffer and clear it so that if we start to play again,
6466 // we won't have old data playing.
6467 result = buffer->Lock( 0, handle->dsBufferSize[0], &audioPtr, &dataLen, NULL, NULL, 0 );
6468 if ( FAILED( result ) ) {
6469 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking output buffer!";
6470 errorText_ = errorStream_.str();
6474 // Zero the DS buffer
6475 ZeroMemory( audioPtr, dataLen );
6477 // Unlock the DS buffer
6478 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
6479 if ( FAILED( result ) ) {
6480 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking output buffer!";
6481 errorText_ = errorStream_.str();
6485 // If we start playing again, we must begin at beginning of buffer.
6486 handle->bufferPointer[0] = 0;
6489 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
6490 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6494 stream_.state = STREAM_STOPPED;
6496 if ( stream_.mode != DUPLEX )
6497 MUTEX_LOCK( &stream_.mutex );
6499 result = buffer->Stop();
6500 if ( FAILED( result ) ) {
6501 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping input buffer!";
6502 errorText_ = errorStream_.str();
6506 // Lock the buffer and clear it so that if we start to play again,
6507 // we won't have old data playing.
6508 result = buffer->Lock( 0, handle->dsBufferSize[1], &audioPtr, &dataLen, NULL, NULL, 0 );
6509 if ( FAILED( result ) ) {
6510 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking input buffer!";
6511 errorText_ = errorStream_.str();
6515 // Zero the DS buffer
6516 ZeroMemory( audioPtr, dataLen );
6518 // Unlock the DS buffer
6519 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
6520 if ( FAILED( result ) ) {
6521 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking input buffer!";
6522 errorText_ = errorStream_.str();
6526 // If we start recording again, we must begin at beginning of buffer.
6527 handle->bufferPointer[1] = 0;
6531 timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows.
6532 MUTEX_UNLOCK( &stream_.mutex );
6534 if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );
6537 void RtApiDs :: abortStream()
6540 if ( stream_.state == STREAM_STOPPED ) {
6541 errorText_ = "RtApiDs::abortStream(): the stream is already stopped!";
6542 error( RtAudioError::WARNING );
6546 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6547 handle->drainCounter = 2;
6552 void RtApiDs :: callbackEvent()
6554 if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) {
6555 Sleep( 50 ); // sleep 50 milliseconds
6559 if ( stream_.state == STREAM_CLOSED ) {
6560 errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!";
6561 error( RtAudioError::WARNING );
6565 CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
6566 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6568 // Check if we were draining the stream and signal is finished.
6569 if ( handle->drainCounter > stream_.nBuffers + 2 ) {
6571 stream_.state = STREAM_STOPPING;
6572 if ( handle->internalDrain == false )
6573 SetEvent( handle->condition );
6579 // Invoke user callback to get fresh output data UNLESS we are
6581 if ( handle->drainCounter == 0 ) {
6582 RtAudioCallback callback = (RtAudioCallback) info->callback;
6583 double streamTime = getStreamTime();
6584 RtAudioStreamStatus status = 0;
6585 if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
6586 status |= RTAUDIO_OUTPUT_UNDERFLOW;
6587 handle->xrun[0] = false;
6589 if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
6590 status |= RTAUDIO_INPUT_OVERFLOW;
6591 handle->xrun[1] = false;
6593 int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
6594 stream_.bufferSize, streamTime, status, info->userData );
6595 if ( cbReturnValue == 2 ) {
6596 stream_.state = STREAM_STOPPING;
6597 handle->drainCounter = 2;
6601 else if ( cbReturnValue == 1 ) {
6602 handle->drainCounter = 1;
6603 handle->internalDrain = true;
6608 DWORD currentWritePointer, safeWritePointer;
6609 DWORD currentReadPointer, safeReadPointer;
6610 UINT nextWritePointer;
6612 LPVOID buffer1 = NULL;
6613 LPVOID buffer2 = NULL;
6614 DWORD bufferSize1 = 0;
6615 DWORD bufferSize2 = 0;
6620 MUTEX_LOCK( &stream_.mutex );
6621 if ( stream_.state == STREAM_STOPPED ) {
6622 MUTEX_UNLOCK( &stream_.mutex );
6626 if ( buffersRolling == false ) {
6627 if ( stream_.mode == DUPLEX ) {
6628 //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
6630 // It takes a while for the devices to get rolling. As a result,
6631 // there's no guarantee that the capture and write device pointers
6632 // will move in lockstep. Wait here for both devices to start
6633 // rolling, and then set our buffer pointers accordingly.
6634 // e.g. Crystal Drivers: the capture buffer starts up 5700 to 9600
6635 // bytes later than the write buffer.
6637 // Stub: a serious risk of having a pre-emptive scheduling round
6638 // take place between the two GetCurrentPosition calls... but I'm
6639 // really not sure how to solve the problem. Temporarily boost to
6640 // Realtime priority, maybe; but I'm not sure what priority the
6641 // DirectSound service threads run at. We *should* be roughly
6642 // within a ms or so of correct.
6644 LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6645 LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6647 DWORD startSafeWritePointer, startSafeReadPointer;
6649 result = dsWriteBuffer->GetCurrentPosition( NULL, &startSafeWritePointer );
6650 if ( FAILED( result ) ) {
6651 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
6652 errorText_ = errorStream_.str();
6653 MUTEX_UNLOCK( &stream_.mutex );
6654 error( RtAudioError::SYSTEM_ERROR );
6657 result = dsCaptureBuffer->GetCurrentPosition( NULL, &startSafeReadPointer );
6658 if ( FAILED( result ) ) {
6659 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
6660 errorText_ = errorStream_.str();
6661 MUTEX_UNLOCK( &stream_.mutex );
6662 error( RtAudioError::SYSTEM_ERROR );
6666 result = dsWriteBuffer->GetCurrentPosition( NULL, &safeWritePointer );
6667 if ( FAILED( result ) ) {
6668 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
6669 errorText_ = errorStream_.str();
6670 MUTEX_UNLOCK( &stream_.mutex );
6671 error( RtAudioError::SYSTEM_ERROR );
6674 result = dsCaptureBuffer->GetCurrentPosition( NULL, &safeReadPointer );
6675 if ( FAILED( result ) ) {
6676 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
6677 errorText_ = errorStream_.str();
6678 MUTEX_UNLOCK( &stream_.mutex );
6679 error( RtAudioError::SYSTEM_ERROR );
6682 if ( safeWritePointer != startSafeWritePointer && safeReadPointer != startSafeReadPointer ) break;
6686 //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
6688 handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];
6689 if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];
6690 handle->bufferPointer[1] = safeReadPointer;
6692 else if ( stream_.mode == OUTPUT ) {
6694 // Set the proper nextWritePosition after initial startup.
6695 LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6696 result = dsWriteBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer );
6697 if ( FAILED( result ) ) {
6698 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
6699 errorText_ = errorStream_.str();
6700 MUTEX_UNLOCK( &stream_.mutex );
6701 error( RtAudioError::SYSTEM_ERROR );
6704 handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];
6705 if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];
6708 buffersRolling = true;
6711 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
6713 LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6715 if ( handle->drainCounter > 1 ) { // write zeros to the output stream
6716 bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
6717 bufferBytes *= formatBytes( stream_.userFormat );
6718 memset( stream_.userBuffer[0], 0, bufferBytes );
6721 // Setup parameters and do buffer conversion if necessary.
6722 if ( stream_.doConvertBuffer[0] ) {
6723 buffer = stream_.deviceBuffer;
6724 convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
6725 bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[0];
6726 bufferBytes *= formatBytes( stream_.deviceFormat[0] );
6729 buffer = stream_.userBuffer[0];
6730 bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
6731 bufferBytes *= formatBytes( stream_.userFormat );
6734 // No byte swapping necessary in DirectSound implementation.
6736 // Ahhh ... windoze. 16-bit data is signed but 8-bit data is
6737 // unsigned. So, we need to convert our signed 8-bit data here to
6739 if ( stream_.deviceFormat[0] == RTAUDIO_SINT8 )
6740 for ( int i=0; i<bufferBytes; i++ ) buffer[i] = (unsigned char) ( buffer[i] + 128 );
6742 DWORD dsBufferSize = handle->dsBufferSize[0];
6743 nextWritePointer = handle->bufferPointer[0];
6745 DWORD endWrite, leadPointer;
6747 // Find out where the read and "safe write" pointers are.
6748 result = dsBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer );
6749 if ( FAILED( result ) ) {
6750 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
6751 errorText_ = errorStream_.str();
6752 MUTEX_UNLOCK( &stream_.mutex );
6753 error( RtAudioError::SYSTEM_ERROR );
6757 // We will copy our output buffer into the region between
6758 // safeWritePointer and leadPointer. If leadPointer is not
6759 // beyond the next endWrite position, wait until it is.
6760 leadPointer = safeWritePointer + handle->dsPointerLeadTime[0];
6761 //std::cout << "safeWritePointer = " << safeWritePointer << ", leadPointer = " << leadPointer << ", nextWritePointer = " << nextWritePointer << std::endl;
6762 if ( leadPointer > dsBufferSize ) leadPointer -= dsBufferSize;
6763 if ( leadPointer < nextWritePointer ) leadPointer += dsBufferSize; // unwrap offset
6764 endWrite = nextWritePointer + bufferBytes;
6766 // Check whether the entire write region is behind the play pointer.
6767 if ( leadPointer >= endWrite ) break;
6769 // If we are here, then we must wait until the leadPointer advances
6770 // beyond the end of our next write region. We use the
6771 // Sleep() function to suspend operation until that happens.
6772 double millis = ( endWrite - leadPointer ) * 1000.0;
6773 millis /= ( formatBytes( stream_.deviceFormat[0]) * stream_.nDeviceChannels[0] * stream_.sampleRate);
6774 if ( millis < 1.0 ) millis = 1.0;
6775 Sleep( (DWORD) millis );
6778 if ( dsPointerBetween( nextWritePointer, safeWritePointer, currentWritePointer, dsBufferSize )
6779 || dsPointerBetween( endWrite, safeWritePointer, currentWritePointer, dsBufferSize ) ) {
6780 // We've strayed into the forbidden zone ... resync the read pointer.
6781 handle->xrun[0] = true;
6782 nextWritePointer = safeWritePointer + handle->dsPointerLeadTime[0] - bufferBytes;
6783 if ( nextWritePointer >= dsBufferSize ) nextWritePointer -= dsBufferSize;
6784 handle->bufferPointer[0] = nextWritePointer;
6785 endWrite = nextWritePointer + bufferBytes;
6788 // Lock free space in the buffer
6789 result = dsBuffer->Lock( nextWritePointer, bufferBytes, &buffer1,
6790 &bufferSize1, &buffer2, &bufferSize2, 0 );
6791 if ( FAILED( result ) ) {
6792 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking buffer during playback!";
6793 errorText_ = errorStream_.str();
6794 MUTEX_UNLOCK( &stream_.mutex );
6795 error( RtAudioError::SYSTEM_ERROR );
6799 // Copy our buffer into the DS buffer
6800 CopyMemory( buffer1, buffer, bufferSize1 );
6801 if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 );
6803 // Update our buffer offset and unlock sound buffer
6804 dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
6805 if ( FAILED( result ) ) {
6806 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!";
6807 errorText_ = errorStream_.str();
6808 MUTEX_UNLOCK( &stream_.mutex );
6809 error( RtAudioError::SYSTEM_ERROR );
6812 nextWritePointer = ( nextWritePointer + bufferSize1 + bufferSize2 ) % dsBufferSize;
6813 handle->bufferPointer[0] = nextWritePointer;
6816 // Don't bother draining input
6817 if ( handle->drainCounter ) {
6818 handle->drainCounter++;
6822 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
6824 // Setup parameters.
6825 if ( stream_.doConvertBuffer[1] ) {
6826 buffer = stream_.deviceBuffer;
6827 bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[1];
6828 bufferBytes *= formatBytes( stream_.deviceFormat[1] );
6831 buffer = stream_.userBuffer[1];
6832 bufferBytes = stream_.bufferSize * stream_.nUserChannels[1];
6833 bufferBytes *= formatBytes( stream_.userFormat );
6836 LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6837 long nextReadPointer = handle->bufferPointer[1];
6838 DWORD dsBufferSize = handle->dsBufferSize[1];
6840 // Find out where the write and "safe read" pointers are.
6841 result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer );
6842 if ( FAILED( result ) ) {
6843 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
6844 errorText_ = errorStream_.str();
6845 MUTEX_UNLOCK( &stream_.mutex );
6846 error( RtAudioError::SYSTEM_ERROR );
6850 if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset
6851 DWORD endRead = nextReadPointer + bufferBytes;
6853 // Handling depends on whether we are INPUT or DUPLEX.
6854 // If we're in INPUT mode then waiting is a good thing. If we're in DUPLEX mode,
6855 // then a wait here will drag the write pointers into the forbidden zone.
6857 // In DUPLEX mode, rather than wait, we will back off the read pointer until
6858 // it's in a safe position. This causes dropouts, but it seems to be the only
6859 // practical way to sync up the read and write pointers reliably, given the
6860 // the very complex relationship between phase and increment of the read and write
6863 // In order to minimize audible dropouts in DUPLEX mode, we will
6864 // provide a pre-roll period of 0.5 seconds in which we return
6865 // zeros from the read buffer while the pointers sync up.
6867 if ( stream_.mode == DUPLEX ) {
6868 if ( safeReadPointer < endRead ) {
6869 if ( duplexPrerollBytes <= 0 ) {
6870 // Pre-roll time over. Be more agressive.
6871 int adjustment = endRead-safeReadPointer;
6873 handle->xrun[1] = true;
6875 // - large adjustments: we've probably run out of CPU cycles, so just resync exactly,
6876 // and perform fine adjustments later.
6877 // - small adjustments: back off by twice as much.
6878 if ( adjustment >= 2*bufferBytes )
6879 nextReadPointer = safeReadPointer-2*bufferBytes;
6881 nextReadPointer = safeReadPointer-bufferBytes-adjustment;
6883 if ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;
6887 // In pre=roll time. Just do it.
6888 nextReadPointer = safeReadPointer - bufferBytes;
6889 while ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;
6891 endRead = nextReadPointer + bufferBytes;
6894 else { // mode == INPUT
6895 while ( safeReadPointer < endRead && stream_.callbackInfo.isRunning ) {
6896 // See comments for playback.
6897 double millis = (endRead - safeReadPointer) * 1000.0;
6898 millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate);
6899 if ( millis < 1.0 ) millis = 1.0;
6900 Sleep( (DWORD) millis );
6902 // Wake up and find out where we are now.
6903 result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer );
6904 if ( FAILED( result ) ) {
6905 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
6906 errorText_ = errorStream_.str();
6907 MUTEX_UNLOCK( &stream_.mutex );
6908 error( RtAudioError::SYSTEM_ERROR );
6912 if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset
6916 // Lock free space in the buffer
6917 result = dsBuffer->Lock( nextReadPointer, bufferBytes, &buffer1,
6918 &bufferSize1, &buffer2, &bufferSize2, 0 );
6919 if ( FAILED( result ) ) {
6920 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking capture buffer!";
6921 errorText_ = errorStream_.str();
6922 MUTEX_UNLOCK( &stream_.mutex );
6923 error( RtAudioError::SYSTEM_ERROR );
6927 if ( duplexPrerollBytes <= 0 ) {
6928 // Copy our buffer into the DS buffer
6929 CopyMemory( buffer, buffer1, bufferSize1 );
6930 if ( buffer2 != NULL ) CopyMemory( buffer+bufferSize1, buffer2, bufferSize2 );
6933 memset( buffer, 0, bufferSize1 );
6934 if ( buffer2 != NULL ) memset( buffer + bufferSize1, 0, bufferSize2 );
6935 duplexPrerollBytes -= bufferSize1 + bufferSize2;
6938 // Update our buffer offset and unlock sound buffer
6939 nextReadPointer = ( nextReadPointer + bufferSize1 + bufferSize2 ) % dsBufferSize;
6940 dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
6941 if ( FAILED( result ) ) {
6942 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking capture buffer!";
6943 errorText_ = errorStream_.str();
6944 MUTEX_UNLOCK( &stream_.mutex );
6945 error( RtAudioError::SYSTEM_ERROR );
6948 handle->bufferPointer[1] = nextReadPointer;
6950 // No byte swapping necessary in DirectSound implementation.
6952 // If necessary, convert 8-bit data from unsigned to signed.
6953 if ( stream_.deviceFormat[1] == RTAUDIO_SINT8 )
6954 for ( int j=0; j<bufferBytes; j++ ) buffer[j] = (signed char) ( buffer[j] - 128 );
6956 // Do buffer conversion if necessary.
6957 if ( stream_.doConvertBuffer[1] )
6958 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
6962 MUTEX_UNLOCK( &stream_.mutex );
6963 RtApi::tickStreamTime();
6966 // Definitions for utility functions and callbacks
6967 // specific to the DirectSound implementation.
6969 static unsigned __stdcall callbackHandler( void *ptr )
6971 CallbackInfo *info = (CallbackInfo *) ptr;
6972 RtApiDs *object = (RtApiDs *) info->object;
6973 bool* isRunning = &info->isRunning;
6975 while ( *isRunning == true ) {
6976 object->callbackEvent();
6983 static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
6984 LPCTSTR description,
6988 struct DsProbeData& probeInfo = *(struct DsProbeData*) lpContext;
6989 std::vector<struct DsDevice>& dsDevices = *probeInfo.dsDevices;
6992 bool validDevice = false;
6993 if ( probeInfo.isInput == true ) {
6995 LPDIRECTSOUNDCAPTURE object;
6997 hr = DirectSoundCaptureCreate( lpguid, &object, NULL );
6998 if ( hr != DS_OK ) return TRUE;
7000 caps.dwSize = sizeof(caps);
7001 hr = object->GetCaps( &caps );
7002 if ( hr == DS_OK ) {
7003 if ( caps.dwChannels > 0 && caps.dwFormats > 0 )
7010 LPDIRECTSOUND object;
7011 hr = DirectSoundCreate( lpguid, &object, NULL );
7012 if ( hr != DS_OK ) return TRUE;
7014 caps.dwSize = sizeof(caps);
7015 hr = object->GetCaps( &caps );
7016 if ( hr == DS_OK ) {
7017 if ( caps.dwFlags & DSCAPS_PRIMARYMONO || caps.dwFlags & DSCAPS_PRIMARYSTEREO )
7023 // If good device, then save its name and guid.
7024 std::string name = convertCharPointerToStdString( description );
7025 //if ( name == "Primary Sound Driver" || name == "Primary Sound Capture Driver" )
7026 if ( lpguid == NULL )
7027 name = "Default Device";
7028 if ( validDevice ) {
7029 for ( unsigned int i=0; i<dsDevices.size(); i++ ) {
7030 if ( dsDevices[i].name == name ) {
7031 dsDevices[i].found = true;
7032 if ( probeInfo.isInput ) {
7033 dsDevices[i].id[1] = lpguid;
7034 dsDevices[i].validId[1] = true;
7037 dsDevices[i].id[0] = lpguid;
7038 dsDevices[i].validId[0] = true;
7046 device.found = true;
7047 if ( probeInfo.isInput ) {
7048 device.id[1] = lpguid;
7049 device.validId[1] = true;
7052 device.id[0] = lpguid;
7053 device.validId[0] = true;
7055 dsDevices.push_back( device );
7061 static const char* getErrorString( int code )
7065 case DSERR_ALLOCATED:
7066 return "Already allocated";
7068 case DSERR_CONTROLUNAVAIL:
7069 return "Control unavailable";
7071 case DSERR_INVALIDPARAM:
7072 return "Invalid parameter";
7074 case DSERR_INVALIDCALL:
7075 return "Invalid call";
7078 return "Generic error";
7080 case DSERR_PRIOLEVELNEEDED:
7081 return "Priority level needed";
7083 case DSERR_OUTOFMEMORY:
7084 return "Out of memory";
7086 case DSERR_BADFORMAT:
7087 return "The sample rate or the channel format is not supported";
7089 case DSERR_UNSUPPORTED:
7090 return "Not supported";
7092 case DSERR_NODRIVER:
7095 case DSERR_ALREADYINITIALIZED:
7096 return "Already initialized";
7098 case DSERR_NOAGGREGATION:
7099 return "No aggregation";
7101 case DSERR_BUFFERLOST:
7102 return "Buffer lost";
7104 case DSERR_OTHERAPPHASPRIO:
7105 return "Another application already has priority";
7107 case DSERR_UNINITIALIZED:
7108 return "Uninitialized";
7111 return "DirectSound unknown error";
7114 //******************** End of __WINDOWS_DS__ *********************//
7118 #if defined(__LINUX_ALSA__)
7120 #include <alsa/asoundlib.h>
7123 // A structure to hold various information related to the ALSA API
7126 snd_pcm_t *handles[2];
7129 pthread_cond_t runnable_cv;
7133 :synchronized(false), runnable(false) { xrun[0] = false; xrun[1] = false; }
7136 static void *alsaCallbackHandler( void * ptr );
7138 RtApiAlsa :: RtApiAlsa()
7140 // Nothing to do here.
7143 RtApiAlsa :: ~RtApiAlsa()
7145 if ( stream_.state != STREAM_CLOSED ) closeStream();
7148 unsigned int RtApiAlsa :: getDeviceCount( void )
7150 unsigned nDevices = 0;
7151 int result, subdevice, card;
7155 // Count cards and devices
7157 snd_card_next( &card );
7158 while ( card >= 0 ) {
7159 sprintf( name, "hw:%d", card );
7160 result = snd_ctl_open( &handle, name, 0 );
7162 errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << ".";
7163 errorText_ = errorStream_.str();
7164 error( RtAudioError::WARNING );
7169 result = snd_ctl_pcm_next_device( handle, &subdevice );
7171 errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
7172 errorText_ = errorStream_.str();
7173 error( RtAudioError::WARNING );
7176 if ( subdevice < 0 )
7181 snd_ctl_close( handle );
7182 snd_card_next( &card );
7185 result = snd_ctl_open( &handle, "default", 0 );
7188 snd_ctl_close( handle );
7194 RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device )
7196 RtAudio::DeviceInfo info;
7197 info.probed = false;
7199 unsigned nDevices = 0;
7200 int result, subdevice, card;
7204 // Count cards and devices
7207 snd_card_next( &card );
7208 while ( card >= 0 ) {
7209 sprintf( name, "hw:%d", card );
7210 result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
7212 errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << ".";
7213 errorText_ = errorStream_.str();
7214 error( RtAudioError::WARNING );
7219 result = snd_ctl_pcm_next_device( chandle, &subdevice );
7221 errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
7222 errorText_ = errorStream_.str();
7223 error( RtAudioError::WARNING );
7226 if ( subdevice < 0 ) break;
7227 if ( nDevices == device ) {
7228 sprintf( name, "hw:%d,%d", card, subdevice );
7234 snd_ctl_close( chandle );
7235 snd_card_next( &card );
7238 result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
7239 if ( result == 0 ) {
7240 if ( nDevices == device ) {
7241 strcpy( name, "default" );
7247 if ( nDevices == 0 ) {
7248 errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!";
7249 error( RtAudioError::INVALID_USE );
7253 if ( device >= nDevices ) {
7254 errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!";
7255 error( RtAudioError::INVALID_USE );
7261 // If a stream is already open, we cannot probe the stream devices.
7262 // Thus, use the saved results.
7263 if ( stream_.state != STREAM_CLOSED &&
7264 ( stream_.device[0] == device || stream_.device[1] == device ) ) {
7265 snd_ctl_close( chandle );
7266 if ( device >= devices_.size() ) {
7267 errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened.";
7268 error( RtAudioError::WARNING );
7271 return devices_[ device ];
7274 int openMode = SND_PCM_ASYNC;
7275 snd_pcm_stream_t stream;
7276 snd_pcm_info_t *pcminfo;
7277 snd_pcm_info_alloca( &pcminfo );
7279 snd_pcm_hw_params_t *params;
7280 snd_pcm_hw_params_alloca( ¶ms );
7282 // First try for playback unless default device (which has subdev -1)
7283 stream = SND_PCM_STREAM_PLAYBACK;
7284 snd_pcm_info_set_stream( pcminfo, stream );
7285 if ( subdevice != -1 ) {
7286 snd_pcm_info_set_device( pcminfo, subdevice );
7287 snd_pcm_info_set_subdevice( pcminfo, 0 );
7289 result = snd_ctl_pcm_info( chandle, pcminfo );
7291 // Device probably doesn't support playback.
7296 result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK );
7298 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
7299 errorText_ = errorStream_.str();
7300 error( RtAudioError::WARNING );
7304 // The device is open ... fill the parameter structure.
7305 result = snd_pcm_hw_params_any( phandle, params );
7307 snd_pcm_close( phandle );
7308 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
7309 errorText_ = errorStream_.str();
7310 error( RtAudioError::WARNING );
7314 // Get output channel information.
7316 result = snd_pcm_hw_params_get_channels_max( params, &value );
7318 snd_pcm_close( phandle );
7319 errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << ".";
7320 errorText_ = errorStream_.str();
7321 error( RtAudioError::WARNING );
7324 info.outputChannels = value;
7325 snd_pcm_close( phandle );
7328 stream = SND_PCM_STREAM_CAPTURE;
7329 snd_pcm_info_set_stream( pcminfo, stream );
7331 // Now try for capture unless default device (with subdev = -1)
7332 if ( subdevice != -1 ) {
7333 result = snd_ctl_pcm_info( chandle, pcminfo );
7334 snd_ctl_close( chandle );
7336 // Device probably doesn't support capture.
7337 if ( info.outputChannels == 0 ) return info;
7338 goto probeParameters;
7342 snd_ctl_close( chandle );
7344 result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
7346 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
7347 errorText_ = errorStream_.str();
7348 error( RtAudioError::WARNING );
7349 if ( info.outputChannels == 0 ) return info;
7350 goto probeParameters;
7353 // The device is open ... fill the parameter structure.
7354 result = snd_pcm_hw_params_any( phandle, params );
7356 snd_pcm_close( phandle );
7357 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
7358 errorText_ = errorStream_.str();
7359 error( RtAudioError::WARNING );
7360 if ( info.outputChannels == 0 ) return info;
7361 goto probeParameters;
7364 result = snd_pcm_hw_params_get_channels_max( params, &value );
7366 snd_pcm_close( phandle );
7367 errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << ".";
7368 errorText_ = errorStream_.str();
7369 error( RtAudioError::WARNING );
7370 if ( info.outputChannels == 0 ) return info;
7371 goto probeParameters;
7373 info.inputChannels = value;
7374 snd_pcm_close( phandle );
7376 // If device opens for both playback and capture, we determine the channels.
7377 if ( info.outputChannels > 0 && info.inputChannels > 0 )
7378 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
7380 // ALSA doesn't provide default devices so we'll use the first available one.
7381 if ( device == 0 && info.outputChannels > 0 )
7382 info.isDefaultOutput = true;
7383 if ( device == 0 && info.inputChannels > 0 )
7384 info.isDefaultInput = true;
7387 // At this point, we just need to figure out the supported data
7388 // formats and sample rates. We'll proceed by opening the device in
7389 // the direction with the maximum number of channels, or playback if
7390 // they are equal. This might limit our sample rate options, but so
7393 if ( info.outputChannels >= info.inputChannels )
7394 stream = SND_PCM_STREAM_PLAYBACK;
7396 stream = SND_PCM_STREAM_CAPTURE;
7397 snd_pcm_info_set_stream( pcminfo, stream );
7399 result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
7401 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
7402 errorText_ = errorStream_.str();
7403 error( RtAudioError::WARNING );
7407 // The device is open ... fill the parameter structure.
7408 result = snd_pcm_hw_params_any( phandle, params );
7410 snd_pcm_close( phandle );
7411 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
7412 errorText_ = errorStream_.str();
7413 error( RtAudioError::WARNING );
7417 // Test our discrete set of sample rate values.
7418 info.sampleRates.clear();
7419 for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {
7420 if ( snd_pcm_hw_params_test_rate( phandle, params, SAMPLE_RATES[i], 0 ) == 0 ) {
7421 info.sampleRates.push_back( SAMPLE_RATES[i] );
7423 if ( !info.preferredSampleRate || ( SAMPLE_RATES[i] <= 48000 && SAMPLE_RATES[i] > info.preferredSampleRate ) )
7424 info.preferredSampleRate = SAMPLE_RATES[i];
7427 if ( info.sampleRates.size() == 0 ) {
7428 snd_pcm_close( phandle );
7429 errorStream_ << "RtApiAlsa::getDeviceInfo: no supported sample rates found for device (" << name << ").";
7430 errorText_ = errorStream_.str();
7431 error( RtAudioError::WARNING );
7435 // Probe the supported data formats ... we don't care about endian-ness just yet
7436 snd_pcm_format_t format;
7437 info.nativeFormats = 0;
7438 format = SND_PCM_FORMAT_S8;
7439 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7440 info.nativeFormats |= RTAUDIO_SINT8;
7441 format = SND_PCM_FORMAT_S16;
7442 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7443 info.nativeFormats |= RTAUDIO_SINT16;
7444 format = SND_PCM_FORMAT_S24;
7445 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7446 info.nativeFormats |= RTAUDIO_SINT24;
7447 format = SND_PCM_FORMAT_S32;
7448 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7449 info.nativeFormats |= RTAUDIO_SINT32;
7450 format = SND_PCM_FORMAT_FLOAT;
7451 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7452 info.nativeFormats |= RTAUDIO_FLOAT32;
7453 format = SND_PCM_FORMAT_FLOAT64;
7454 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7455 info.nativeFormats |= RTAUDIO_FLOAT64;
7457 // Check that we have at least one supported format
7458 if ( info.nativeFormats == 0 ) {
7459 snd_pcm_close( phandle );
7460 errorStream_ << "RtApiAlsa::getDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio.";
7461 errorText_ = errorStream_.str();
7462 error( RtAudioError::WARNING );
7466 // Get the device name
7468 result = snd_card_get_name( card, &cardname );
7469 if ( result >= 0 ) {
7470 sprintf( name, "hw:%s,%d", cardname, subdevice );
7475 // That's all ... close the device and return
7476 snd_pcm_close( phandle );
7481 void RtApiAlsa :: saveDeviceInfo( void )
7485 unsigned int nDevices = getDeviceCount();
7486 devices_.resize( nDevices );
7487 for ( unsigned int i=0; i<nDevices; i++ )
7488 devices_[i] = getDeviceInfo( i );
7491 bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
7492 unsigned int firstChannel, unsigned int sampleRate,
7493 RtAudioFormat format, unsigned int *bufferSize,
7494 RtAudio::StreamOptions *options )
7497 #if defined(__RTAUDIO_DEBUG__)
7499 snd_output_stdio_attach(&out, stderr, 0);
7502 // I'm not using the "plug" interface ... too much inconsistent behavior.
7504 unsigned nDevices = 0;
7505 int result, subdevice, card;
7509 if ( options && options->flags & RTAUDIO_ALSA_USE_DEFAULT )
7510 snprintf(name, sizeof(name), "%s", "default");
7512 // Count cards and devices
7514 snd_card_next( &card );
7515 while ( card >= 0 ) {
7516 sprintf( name, "hw:%d", card );
7517 result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
7519 errorStream_ << "RtApiAlsa::probeDeviceOpen: control open, card = " << card << ", " << snd_strerror( result ) << ".";
7520 errorText_ = errorStream_.str();
7525 result = snd_ctl_pcm_next_device( chandle, &subdevice );
7526 if ( result < 0 ) break;
7527 if ( subdevice < 0 ) break;
7528 if ( nDevices == device ) {
7529 sprintf( name, "hw:%d,%d", card, subdevice );
7530 snd_ctl_close( chandle );
7535 snd_ctl_close( chandle );
7536 snd_card_next( &card );
7539 result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
7540 if ( result == 0 ) {
7541 if ( nDevices == device ) {
7542 strcpy( name, "default" );
7543 snd_ctl_close( chandle );
7548 snd_ctl_close( chandle );
7550 if ( nDevices == 0 ) {
7551 // This should not happen because a check is made before this function is called.
7552 errorText_ = "RtApiAlsa::probeDeviceOpen: no devices found!";
7556 if ( device >= nDevices ) {
7557 // This should not happen because a check is made before this function is called.
7558 errorText_ = "RtApiAlsa::probeDeviceOpen: device ID is invalid!";
7565 // The getDeviceInfo() function will not work for a device that is
7566 // already open. Thus, we'll probe the system before opening a
7567 // stream and save the results for use by getDeviceInfo().
7568 if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) // only do once
7569 this->saveDeviceInfo();
7571 snd_pcm_stream_t stream;
7572 if ( mode == OUTPUT )
7573 stream = SND_PCM_STREAM_PLAYBACK;
7575 stream = SND_PCM_STREAM_CAPTURE;
7578 int openMode = SND_PCM_ASYNC;
7579 result = snd_pcm_open( &phandle, name, stream, openMode );
7581 if ( mode == OUTPUT )
7582 errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for output.";
7584 errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for input.";
7585 errorText_ = errorStream_.str();
7589 // Fill the parameter structure.
7590 snd_pcm_hw_params_t *hw_params;
7591 snd_pcm_hw_params_alloca( &hw_params );
7592 result = snd_pcm_hw_params_any( phandle, hw_params );
7594 snd_pcm_close( phandle );
7595 errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") parameters, " << snd_strerror( result ) << ".";
7596 errorText_ = errorStream_.str();
7600 #if defined(__RTAUDIO_DEBUG__)
7601 fprintf( stderr, "\nRtApiAlsa: dump hardware params just after device open:\n\n" );
7602 snd_pcm_hw_params_dump( hw_params, out );
7605 // Set access ... check user preference.
7606 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) {
7607 stream_.userInterleaved = false;
7608 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
7610 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
7611 stream_.deviceInterleaved[mode] = true;
7614 stream_.deviceInterleaved[mode] = false;
7617 stream_.userInterleaved = true;
7618 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
7620 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
7621 stream_.deviceInterleaved[mode] = false;
7624 stream_.deviceInterleaved[mode] = true;
7628 snd_pcm_close( phandle );
7629 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") access, " << snd_strerror( result ) << ".";
7630 errorText_ = errorStream_.str();
7634 // Determine how to set the device format.
7635 stream_.userFormat = format;
7636 snd_pcm_format_t deviceFormat = SND_PCM_FORMAT_UNKNOWN;
7638 if ( format == RTAUDIO_SINT8 )
7639 deviceFormat = SND_PCM_FORMAT_S8;
7640 else if ( format == RTAUDIO_SINT16 )
7641 deviceFormat = SND_PCM_FORMAT_S16;
7642 else if ( format == RTAUDIO_SINT24 )
7643 deviceFormat = SND_PCM_FORMAT_S24;
7644 else if ( format == RTAUDIO_SINT32 )
7645 deviceFormat = SND_PCM_FORMAT_S32;
7646 else if ( format == RTAUDIO_FLOAT32 )
7647 deviceFormat = SND_PCM_FORMAT_FLOAT;
7648 else if ( format == RTAUDIO_FLOAT64 )
7649 deviceFormat = SND_PCM_FORMAT_FLOAT64;
7651 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat) == 0) {
7652 stream_.deviceFormat[mode] = format;
7656 // The user requested format is not natively supported by the device.
7657 deviceFormat = SND_PCM_FORMAT_FLOAT64;
7658 if ( snd_pcm_hw_params_test_format( phandle, hw_params, deviceFormat ) == 0 ) {
7659 stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
7663 deviceFormat = SND_PCM_FORMAT_FLOAT;
7664 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7665 stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
7669 deviceFormat = SND_PCM_FORMAT_S32;
7670 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7671 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
7675 deviceFormat = SND_PCM_FORMAT_S24;
7676 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7677 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
7681 deviceFormat = SND_PCM_FORMAT_S16;
7682 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7683 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
7687 deviceFormat = SND_PCM_FORMAT_S8;
7688 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7689 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
7693 // If we get here, no supported format was found.
7694 snd_pcm_close( phandle );
7695 errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device " << device << " data format not supported by RtAudio.";
7696 errorText_ = errorStream_.str();
7700 result = snd_pcm_hw_params_set_format( phandle, hw_params, deviceFormat );
7702 snd_pcm_close( phandle );
7703 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") data format, " << snd_strerror( result ) << ".";
7704 errorText_ = errorStream_.str();
7708 // Determine whether byte-swaping is necessary.
7709 stream_.doByteSwap[mode] = false;
7710 if ( deviceFormat != SND_PCM_FORMAT_S8 ) {
7711 result = snd_pcm_format_cpu_endian( deviceFormat );
7713 stream_.doByteSwap[mode] = true;
7714 else if (result < 0) {
7715 snd_pcm_close( phandle );
7716 errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") endian-ness, " << snd_strerror( result ) << ".";
7717 errorText_ = errorStream_.str();
7722 // Set the sample rate.
7723 result = snd_pcm_hw_params_set_rate_near( phandle, hw_params, (unsigned int*) &sampleRate, 0 );
7725 snd_pcm_close( phandle );
7726 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting sample rate on device (" << name << "), " << snd_strerror( result ) << ".";
7727 errorText_ = errorStream_.str();
7731 // Determine the number of channels for this device. We support a possible
7732 // minimum device channel number > than the value requested by the user.
7733 stream_.nUserChannels[mode] = channels;
7735 result = snd_pcm_hw_params_get_channels_max( hw_params, &value );
7736 unsigned int deviceChannels = value;
7737 if ( result < 0 || deviceChannels < channels + firstChannel ) {
7738 snd_pcm_close( phandle );
7739 errorStream_ << "RtApiAlsa::probeDeviceOpen: requested channel parameters not supported by device (" << name << "), " << snd_strerror( result ) << ".";
7740 errorText_ = errorStream_.str();
7744 result = snd_pcm_hw_params_get_channels_min( hw_params, &value );
7746 snd_pcm_close( phandle );
7747 errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting minimum channels for device (" << name << "), " << snd_strerror( result ) << ".";
7748 errorText_ = errorStream_.str();
7751 deviceChannels = value;
7752 if ( deviceChannels < channels + firstChannel ) deviceChannels = channels + firstChannel;
7753 stream_.nDeviceChannels[mode] = deviceChannels;
7755 // Set the device channels.
7756 result = snd_pcm_hw_params_set_channels( phandle, hw_params, deviceChannels );
7758 snd_pcm_close( phandle );
7759 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting channels for device (" << name << "), " << snd_strerror( result ) << ".";
7760 errorText_ = errorStream_.str();
7764 // Set the buffer (or period) size.
7766 snd_pcm_uframes_t periodSize = *bufferSize;
7767 result = snd_pcm_hw_params_set_period_size_near( phandle, hw_params, &periodSize, &dir );
7769 snd_pcm_close( phandle );
7770 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting period size for device (" << name << "), " << snd_strerror( result ) << ".";
7771 errorText_ = errorStream_.str();
7774 *bufferSize = periodSize;
7776 // Set the buffer number, which in ALSA is referred to as the "period".
7777 unsigned int periods = 0;
7778 if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2;
7779 if ( options && options->numberOfBuffers > 0 ) periods = options->numberOfBuffers;
7780 if ( periods < 2 ) periods = 4; // a fairly safe default value
7781 result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir );
7783 snd_pcm_close( phandle );
7784 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << ".";
7785 errorText_ = errorStream_.str();
7789 // If attempting to setup a duplex stream, the bufferSize parameter
7790 // MUST be the same in both directions!
7791 if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {
7792 snd_pcm_close( phandle );
7793 errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ").";
7794 errorText_ = errorStream_.str();
7798 stream_.bufferSize = *bufferSize;
7800 // Install the hardware configuration
7801 result = snd_pcm_hw_params( phandle, hw_params );
7803 snd_pcm_close( phandle );
7804 errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing hardware configuration on device (" << name << "), " << snd_strerror( result ) << ".";
7805 errorText_ = errorStream_.str();
7809 #if defined(__RTAUDIO_DEBUG__)
7810 fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n");
7811 snd_pcm_hw_params_dump( hw_params, out );
7814 // Set the software configuration to fill buffers with zeros and prevent device stopping on xruns.
7815 snd_pcm_sw_params_t *sw_params = NULL;
7816 snd_pcm_sw_params_alloca( &sw_params );
7817 snd_pcm_sw_params_current( phandle, sw_params );
7818 snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize );
7819 snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, ULONG_MAX );
7820 snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 );
7822 // The following two settings were suggested by Theo Veenker
7823 //snd_pcm_sw_params_set_avail_min( phandle, sw_params, *bufferSize );
7824 //snd_pcm_sw_params_set_xfer_align( phandle, sw_params, 1 );
7826 // here are two options for a fix
7827 //snd_pcm_sw_params_set_silence_size( phandle, sw_params, ULONG_MAX );
7828 snd_pcm_uframes_t val;
7829 snd_pcm_sw_params_get_boundary( sw_params, &val );
7830 snd_pcm_sw_params_set_silence_size( phandle, sw_params, val );
7832 result = snd_pcm_sw_params( phandle, sw_params );
7834 snd_pcm_close( phandle );
7835 errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing software configuration on device (" << name << "), " << snd_strerror( result ) << ".";
7836 errorText_ = errorStream_.str();
7840 #if defined(__RTAUDIO_DEBUG__)
7841 fprintf(stderr, "\nRtApiAlsa: dump software params after installation:\n\n");
7842 snd_pcm_sw_params_dump( sw_params, out );
7845 // Set flags for buffer conversion
7846 stream_.doConvertBuffer[mode] = false;
7847 if ( stream_.userFormat != stream_.deviceFormat[mode] )
7848 stream_.doConvertBuffer[mode] = true;
7849 if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
7850 stream_.doConvertBuffer[mode] = true;
7851 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
7852 stream_.nUserChannels[mode] > 1 )
7853 stream_.doConvertBuffer[mode] = true;
7855 // Allocate the ApiHandle if necessary and then save.
7856 AlsaHandle *apiInfo = 0;
7857 if ( stream_.apiHandle == 0 ) {
7859 apiInfo = (AlsaHandle *) new AlsaHandle;
7861 catch ( std::bad_alloc& ) {
7862 errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating AlsaHandle memory.";
7866 if ( pthread_cond_init( &apiInfo->runnable_cv, NULL ) ) {
7867 errorText_ = "RtApiAlsa::probeDeviceOpen: error initializing pthread condition variable.";
7871 stream_.apiHandle = (void *) apiInfo;
7872 apiInfo->handles[0] = 0;
7873 apiInfo->handles[1] = 0;
7876 apiInfo = (AlsaHandle *) stream_.apiHandle;
7878 apiInfo->handles[mode] = phandle;
7881 // Allocate necessary internal buffers.
7882 unsigned long bufferBytes;
7883 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
7884 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
7885 if ( stream_.userBuffer[mode] == NULL ) {
7886 errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating user buffer memory.";
7890 if ( stream_.doConvertBuffer[mode] ) {
7892 bool makeBuffer = true;
7893 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
7894 if ( mode == INPUT ) {
7895 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
7896 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
7897 if ( bufferBytes <= bytesOut ) makeBuffer = false;
7902 bufferBytes *= *bufferSize;
7903 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
7904 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
7905 if ( stream_.deviceBuffer == NULL ) {
7906 errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating device buffer memory.";
7912 stream_.sampleRate = sampleRate;
7913 stream_.nBuffers = periods;
7914 stream_.device[mode] = device;
7915 stream_.state = STREAM_STOPPED;
7917 // Setup the buffer conversion information structure.
7918 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
7920 // Setup thread if necessary.
7921 if ( stream_.mode == OUTPUT && mode == INPUT ) {
7922 // We had already set up an output stream.
7923 stream_.mode = DUPLEX;
7924 // Link the streams if possible.
7925 apiInfo->synchronized = false;
7926 if ( snd_pcm_link( apiInfo->handles[0], apiInfo->handles[1] ) == 0 )
7927 apiInfo->synchronized = true;
7929 errorText_ = "RtApiAlsa::probeDeviceOpen: unable to synchronize input and output devices.";
7930 error( RtAudioError::WARNING );
7934 stream_.mode = mode;
7936 // Setup callback thread.
7937 stream_.callbackInfo.object = (void *) this;
7939 // Set the thread attributes for joinable and realtime scheduling
7940 // priority (optional). The higher priority will only take affect
7941 // if the program is run as root or suid. Note, under Linux
7942 // processes with CAP_SYS_NICE privilege, a user can change
7943 // scheduling policy and priority (thus need not be root). See
7944 // POSIX "capabilities".
7945 pthread_attr_t attr;
7946 pthread_attr_init( &attr );
7947 pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
7948 #ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
7949 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
7950 stream_.callbackInfo.doRealtime = true;
7951 struct sched_param param;
7952 int priority = options->priority;
7953 int min = sched_get_priority_min( SCHED_RR );
7954 int max = sched_get_priority_max( SCHED_RR );
7955 if ( priority < min ) priority = min;
7956 else if ( priority > max ) priority = max;
7957 param.sched_priority = priority;
7959 // Set the policy BEFORE the priority. Otherwise it fails.
7960 pthread_attr_setschedpolicy(&attr, SCHED_RR);
7961 pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM);
7962 // This is definitely required. Otherwise it fails.
7963 pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED);
7964 pthread_attr_setschedparam(&attr, ¶m);
7967 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
7969 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
7972 stream_.callbackInfo.isRunning = true;
7973 result = pthread_create( &stream_.callbackInfo.thread, &attr, alsaCallbackHandler, &stream_.callbackInfo );
7974 pthread_attr_destroy( &attr );
7976 // Failed. Try instead with default attributes.
7977 result = pthread_create( &stream_.callbackInfo.thread, NULL, alsaCallbackHandler, &stream_.callbackInfo );
7979 stream_.callbackInfo.isRunning = false;
7980 errorText_ = "RtApiAlsa::error creating callback thread!";
7990 pthread_cond_destroy( &apiInfo->runnable_cv );
7991 if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
7992 if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
7994 stream_.apiHandle = 0;
7997 if ( phandle) snd_pcm_close( phandle );
7999 for ( int i=0; i<2; i++ ) {
8000 if ( stream_.userBuffer[i] ) {
8001 free( stream_.userBuffer[i] );
8002 stream_.userBuffer[i] = 0;
8006 if ( stream_.deviceBuffer ) {
8007 free( stream_.deviceBuffer );
8008 stream_.deviceBuffer = 0;
8011 stream_.state = STREAM_CLOSED;
8015 void RtApiAlsa :: closeStream()
8017 if ( stream_.state == STREAM_CLOSED ) {
8018 errorText_ = "RtApiAlsa::closeStream(): no open stream to close!";
8019 error( RtAudioError::WARNING );
8023 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
8024 stream_.callbackInfo.isRunning = false;
8025 MUTEX_LOCK( &stream_.mutex );
8026 if ( stream_.state == STREAM_STOPPED ) {
8027 apiInfo->runnable = true;
8028 pthread_cond_signal( &apiInfo->runnable_cv );
8030 MUTEX_UNLOCK( &stream_.mutex );
8031 pthread_join( stream_.callbackInfo.thread, NULL );
8033 if ( stream_.state == STREAM_RUNNING ) {
8034 stream_.state = STREAM_STOPPED;
8035 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
8036 snd_pcm_drop( apiInfo->handles[0] );
8037 if ( stream_.mode == INPUT || stream_.mode == DUPLEX )
8038 snd_pcm_drop( apiInfo->handles[1] );
8042 pthread_cond_destroy( &apiInfo->runnable_cv );
8043 if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
8044 if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
8046 stream_.apiHandle = 0;
8049 for ( int i=0; i<2; i++ ) {
8050 if ( stream_.userBuffer[i] ) {
8051 free( stream_.userBuffer[i] );
8052 stream_.userBuffer[i] = 0;
8056 if ( stream_.deviceBuffer ) {
8057 free( stream_.deviceBuffer );
8058 stream_.deviceBuffer = 0;
8061 stream_.mode = UNINITIALIZED;
8062 stream_.state = STREAM_CLOSED;
8065 void RtApiAlsa :: startStream()
8067 // This method calls snd_pcm_prepare if the device isn't already in that state.
8070 if ( stream_.state == STREAM_RUNNING ) {
8071 errorText_ = "RtApiAlsa::startStream(): the stream is already running!";
8072 error( RtAudioError::WARNING );
8076 MUTEX_LOCK( &stream_.mutex );
8079 snd_pcm_state_t state;
8080 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
8081 snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
8082 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
8083 state = snd_pcm_state( handle[0] );
8084 if ( state != SND_PCM_STATE_PREPARED ) {
8085 result = snd_pcm_prepare( handle[0] );
8087 errorStream_ << "RtApiAlsa::startStream: error preparing output pcm device, " << snd_strerror( result ) << ".";
8088 errorText_ = errorStream_.str();
8094 if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
8095 result = snd_pcm_drop(handle[1]); // fix to remove stale data received since device has been open
8096 state = snd_pcm_state( handle[1] );
8097 if ( state != SND_PCM_STATE_PREPARED ) {
8098 result = snd_pcm_prepare( handle[1] );
8100 errorStream_ << "RtApiAlsa::startStream: error preparing input pcm device, " << snd_strerror( result ) << ".";
8101 errorText_ = errorStream_.str();
8107 stream_.state = STREAM_RUNNING;
8110 apiInfo->runnable = true;
8111 pthread_cond_signal( &apiInfo->runnable_cv );
8112 MUTEX_UNLOCK( &stream_.mutex );
8114 if ( result >= 0 ) return;
8115 error( RtAudioError::SYSTEM_ERROR );
8118 void RtApiAlsa :: stopStream()
8121 if ( stream_.state == STREAM_STOPPED ) {
8122 errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!";
8123 error( RtAudioError::WARNING );
8127 stream_.state = STREAM_STOPPED;
8128 MUTEX_LOCK( &stream_.mutex );
8131 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
8132 snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
8133 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
8134 if ( apiInfo->synchronized )
8135 result = snd_pcm_drop( handle[0] );
8137 result = snd_pcm_drain( handle[0] );
8139 errorStream_ << "RtApiAlsa::stopStream: error draining output pcm device, " << snd_strerror( result ) << ".";
8140 errorText_ = errorStream_.str();
8145 if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
8146 result = snd_pcm_drop( handle[1] );
8148 errorStream_ << "RtApiAlsa::stopStream: error stopping input pcm device, " << snd_strerror( result ) << ".";
8149 errorText_ = errorStream_.str();
8155 apiInfo->runnable = false; // fixes high CPU usage when stopped
8156 MUTEX_UNLOCK( &stream_.mutex );
8158 if ( result >= 0 ) return;
8159 error( RtAudioError::SYSTEM_ERROR );
8162 void RtApiAlsa :: abortStream()
8165 if ( stream_.state == STREAM_STOPPED ) {
8166 errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!";
8167 error( RtAudioError::WARNING );
8171 stream_.state = STREAM_STOPPED;
8172 MUTEX_LOCK( &stream_.mutex );
8175 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
8176 snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
8177 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
8178 result = snd_pcm_drop( handle[0] );
8180 errorStream_ << "RtApiAlsa::abortStream: error aborting output pcm device, " << snd_strerror( result ) << ".";
8181 errorText_ = errorStream_.str();
8186 if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
8187 result = snd_pcm_drop( handle[1] );
8189 errorStream_ << "RtApiAlsa::abortStream: error aborting input pcm device, " << snd_strerror( result ) << ".";
8190 errorText_ = errorStream_.str();
8196 apiInfo->runnable = false; // fixes high CPU usage when stopped
8197 MUTEX_UNLOCK( &stream_.mutex );
8199 if ( result >= 0 ) return;
8200 error( RtAudioError::SYSTEM_ERROR );
8203 void RtApiAlsa :: callbackEvent()
8205 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
8206 if ( stream_.state == STREAM_STOPPED ) {
8207 MUTEX_LOCK( &stream_.mutex );
8208 while ( !apiInfo->runnable )
8209 pthread_cond_wait( &apiInfo->runnable_cv, &stream_.mutex );
8211 if ( stream_.state != STREAM_RUNNING ) {
8212 MUTEX_UNLOCK( &stream_.mutex );
8215 MUTEX_UNLOCK( &stream_.mutex );
8218 if ( stream_.state == STREAM_CLOSED ) {
8219 errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!";
8220 error( RtAudioError::WARNING );
8224 int doStopStream = 0;
8225 RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
8226 double streamTime = getStreamTime();
8227 RtAudioStreamStatus status = 0;
8228 if ( stream_.mode != INPUT && apiInfo->xrun[0] == true ) {
8229 status |= RTAUDIO_OUTPUT_UNDERFLOW;
8230 apiInfo->xrun[0] = false;
8232 if ( stream_.mode != OUTPUT && apiInfo->xrun[1] == true ) {
8233 status |= RTAUDIO_INPUT_OVERFLOW;
8234 apiInfo->xrun[1] = false;
8236 doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
8237 stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
8239 if ( doStopStream == 2 ) {
8244 MUTEX_LOCK( &stream_.mutex );
8246 // The state might change while waiting on a mutex.
8247 if ( stream_.state == STREAM_STOPPED ) goto unlock;
8253 snd_pcm_sframes_t frames;
8254 RtAudioFormat format;
8255 handle = (snd_pcm_t **) apiInfo->handles;
8257 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
8259 // Setup parameters.
8260 if ( stream_.doConvertBuffer[1] ) {
8261 buffer = stream_.deviceBuffer;
8262 channels = stream_.nDeviceChannels[1];
8263 format = stream_.deviceFormat[1];
8266 buffer = stream_.userBuffer[1];
8267 channels = stream_.nUserChannels[1];
8268 format = stream_.userFormat;
8271 // Read samples from device in interleaved/non-interleaved format.
8272 if ( stream_.deviceInterleaved[1] )
8273 result = snd_pcm_readi( handle[1], buffer, stream_.bufferSize );
8275 void *bufs[channels];
8276 size_t offset = stream_.bufferSize * formatBytes( format );
8277 for ( int i=0; i<channels; i++ )
8278 bufs[i] = (void *) (buffer + (i * offset));
8279 result = snd_pcm_readn( handle[1], bufs, stream_.bufferSize );
8282 if ( result < (int) stream_.bufferSize ) {
8283 // Either an error or overrun occured.
8284 if ( result == -EPIPE ) {
8285 snd_pcm_state_t state = snd_pcm_state( handle[1] );
8286 if ( state == SND_PCM_STATE_XRUN ) {
8287 apiInfo->xrun[1] = true;
8288 result = snd_pcm_prepare( handle[1] );
8290 errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after overrun, " << snd_strerror( result ) << ".";
8291 errorText_ = errorStream_.str();
8295 errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
8296 errorText_ = errorStream_.str();
8300 errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << ".";
8301 errorText_ = errorStream_.str();
8303 error( RtAudioError::WARNING );
8307 // Do byte swapping if necessary.
8308 if ( stream_.doByteSwap[1] )
8309 byteSwapBuffer( buffer, stream_.bufferSize * channels, format );
8311 // Do buffer conversion if necessary.
8312 if ( stream_.doConvertBuffer[1] )
8313 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
8315 // Check stream latency
8316 result = snd_pcm_delay( handle[1], &frames );
8317 if ( result == 0 && frames > 0 ) stream_.latency[1] = frames;
8322 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
8324 // Setup parameters and do buffer conversion if necessary.
8325 if ( stream_.doConvertBuffer[0] ) {
8326 buffer = stream_.deviceBuffer;
8327 convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
8328 channels = stream_.nDeviceChannels[0];
8329 format = stream_.deviceFormat[0];
8332 buffer = stream_.userBuffer[0];
8333 channels = stream_.nUserChannels[0];
8334 format = stream_.userFormat;
8337 // Do byte swapping if necessary.
8338 if ( stream_.doByteSwap[0] )
8339 byteSwapBuffer(buffer, stream_.bufferSize * channels, format);
8341 // Write samples to device in interleaved/non-interleaved format.
8342 if ( stream_.deviceInterleaved[0] )
8343 result = snd_pcm_writei( handle[0], buffer, stream_.bufferSize );
8345 void *bufs[channels];
8346 size_t offset = stream_.bufferSize * formatBytes( format );
8347 for ( int i=0; i<channels; i++ )
8348 bufs[i] = (void *) (buffer + (i * offset));
8349 result = snd_pcm_writen( handle[0], bufs, stream_.bufferSize );
8352 if ( result < (int) stream_.bufferSize ) {
8353 // Either an error or underrun occured.
8354 if ( result == -EPIPE ) {
8355 snd_pcm_state_t state = snd_pcm_state( handle[0] );
8356 if ( state == SND_PCM_STATE_XRUN ) {
8357 apiInfo->xrun[0] = true;
8358 result = snd_pcm_prepare( handle[0] );
8360 errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after underrun, " << snd_strerror( result ) << ".";
8361 errorText_ = errorStream_.str();
8364 errorText_ = "RtApiAlsa::callbackEvent: audio write error, underrun.";
8367 errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
8368 errorText_ = errorStream_.str();
8372 errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << ".";
8373 errorText_ = errorStream_.str();
8375 error( RtAudioError::WARNING );
8379 // Check stream latency
8380 result = snd_pcm_delay( handle[0], &frames );
8381 if ( result == 0 && frames > 0 ) stream_.latency[0] = frames;
8385 MUTEX_UNLOCK( &stream_.mutex );
8387 RtApi::tickStreamTime();
8388 if ( doStopStream == 1 ) this->stopStream();
8391 static void *alsaCallbackHandler( void *ptr )
8393 CallbackInfo *info = (CallbackInfo *) ptr;
8394 RtApiAlsa *object = (RtApiAlsa *) info->object;
8395 bool *isRunning = &info->isRunning;
8397 #ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
8398 if ( info->doRealtime ) {
8399 std::cerr << "RtAudio alsa: " <<
8400 (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") <<
8401 "running realtime scheduling" << std::endl;
8405 while ( *isRunning == true ) {
8406 pthread_testcancel();
8407 object->callbackEvent();
8410 pthread_exit( NULL );
8413 //******************** End of __LINUX_ALSA__ *********************//
8416 #if defined(__LINUX_PULSE__)
8418 // Code written by Peter Meerwald, pmeerw@pmeerw.net
8419 // and Tristan Matthews.
8421 #include <pulse/error.h>
8422 #include <pulse/simple.h>
8425 static const unsigned int SUPPORTED_SAMPLERATES[] = { 8000, 16000, 22050, 32000,
8426 44100, 48000, 96000, 0};
8428 struct rtaudio_pa_format_mapping_t {
8429 RtAudioFormat rtaudio_format;
8430 pa_sample_format_t pa_format;
8433 static const rtaudio_pa_format_mapping_t supported_sampleformats[] = {
8434 {RTAUDIO_SINT16, PA_SAMPLE_S16LE},
8435 {RTAUDIO_SINT32, PA_SAMPLE_S32LE},
8436 {RTAUDIO_FLOAT32, PA_SAMPLE_FLOAT32LE},
8437 {0, PA_SAMPLE_INVALID}};
8439 struct PulseAudioHandle {
8443 pthread_cond_t runnable_cv;
8445 PulseAudioHandle() : s_play(0), s_rec(0), runnable(false) { }
8448 RtApiPulse::~RtApiPulse()
8450 if ( stream_.state != STREAM_CLOSED )
8454 unsigned int RtApiPulse::getDeviceCount( void )
8459 RtAudio::DeviceInfo RtApiPulse::getDeviceInfo( unsigned int /*device*/ )
8461 RtAudio::DeviceInfo info;
8463 info.name = "PulseAudio";
8464 info.outputChannels = 2;
8465 info.inputChannels = 2;
8466 info.duplexChannels = 2;
8467 info.isDefaultOutput = true;
8468 info.isDefaultInput = true;
8470 for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr )
8471 info.sampleRates.push_back( *sr );
8473 info.preferredSampleRate = 48000;
8474 info.nativeFormats = RTAUDIO_SINT16 | RTAUDIO_SINT32 | RTAUDIO_FLOAT32;
8479 static void *pulseaudio_callback( void * user )
8481 CallbackInfo *cbi = static_cast<CallbackInfo *>( user );
8482 RtApiPulse *context = static_cast<RtApiPulse *>( cbi->object );
8483 volatile bool *isRunning = &cbi->isRunning;
8485 #ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
8486 if (cbi->doRealtime) {
8487 std::cerr << "RtAudio pulse: " <<
8488 (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") <<
8489 "running realtime scheduling" << std::endl;
8493 while ( *isRunning ) {
8494 pthread_testcancel();
8495 context->callbackEvent();
8498 pthread_exit( NULL );
8501 void RtApiPulse::closeStream( void )
8503 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8505 stream_.callbackInfo.isRunning = false;
8507 MUTEX_LOCK( &stream_.mutex );
8508 if ( stream_.state == STREAM_STOPPED ) {
8509 pah->runnable = true;
8510 pthread_cond_signal( &pah->runnable_cv );
8512 MUTEX_UNLOCK( &stream_.mutex );
8514 pthread_join( pah->thread, 0 );
8515 if ( pah->s_play ) {
8516 pa_simple_flush( pah->s_play, NULL );
8517 pa_simple_free( pah->s_play );
8520 pa_simple_free( pah->s_rec );
8522 pthread_cond_destroy( &pah->runnable_cv );
8524 stream_.apiHandle = 0;
8527 if ( stream_.userBuffer[0] ) {
8528 free( stream_.userBuffer[0] );
8529 stream_.userBuffer[0] = 0;
8531 if ( stream_.userBuffer[1] ) {
8532 free( stream_.userBuffer[1] );
8533 stream_.userBuffer[1] = 0;
8536 stream_.state = STREAM_CLOSED;
8537 stream_.mode = UNINITIALIZED;
8540 void RtApiPulse::callbackEvent( void )
8542 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8544 if ( stream_.state == STREAM_STOPPED ) {
8545 MUTEX_LOCK( &stream_.mutex );
8546 while ( !pah->runnable )
8547 pthread_cond_wait( &pah->runnable_cv, &stream_.mutex );
8549 if ( stream_.state != STREAM_RUNNING ) {
8550 MUTEX_UNLOCK( &stream_.mutex );
8553 MUTEX_UNLOCK( &stream_.mutex );
8556 if ( stream_.state == STREAM_CLOSED ) {
8557 errorText_ = "RtApiPulse::callbackEvent(): the stream is closed ... "
8558 "this shouldn't happen!";
8559 error( RtAudioError::WARNING );
8563 RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
8564 double streamTime = getStreamTime();
8565 RtAudioStreamStatus status = 0;
8566 int doStopStream = callback( stream_.userBuffer[OUTPUT], stream_.userBuffer[INPUT],
8567 stream_.bufferSize, streamTime, status,
8568 stream_.callbackInfo.userData );
8570 if ( doStopStream == 2 ) {
8575 MUTEX_LOCK( &stream_.mutex );
8576 void *pulse_in = stream_.doConvertBuffer[INPUT] ? stream_.deviceBuffer : stream_.userBuffer[INPUT];
8577 void *pulse_out = stream_.doConvertBuffer[OUTPUT] ? stream_.deviceBuffer : stream_.userBuffer[OUTPUT];
8579 if ( stream_.state != STREAM_RUNNING )
8584 if (stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
8585 if ( stream_.doConvertBuffer[OUTPUT] ) {
8586 convertBuffer( stream_.deviceBuffer,
8587 stream_.userBuffer[OUTPUT],
8588 stream_.convertInfo[OUTPUT] );
8589 bytes = stream_.nDeviceChannels[OUTPUT] * stream_.bufferSize *
8590 formatBytes( stream_.deviceFormat[OUTPUT] );
8592 bytes = stream_.nUserChannels[OUTPUT] * stream_.bufferSize *
8593 formatBytes( stream_.userFormat );
8595 if ( pa_simple_write( pah->s_play, pulse_out, bytes, &pa_error ) < 0 ) {
8596 errorStream_ << "RtApiPulse::callbackEvent: audio write error, " <<
8597 pa_strerror( pa_error ) << ".";
8598 errorText_ = errorStream_.str();
8599 error( RtAudioError::WARNING );
8603 if ( stream_.mode == INPUT || stream_.mode == DUPLEX) {
8604 if ( stream_.doConvertBuffer[INPUT] )
8605 bytes = stream_.nDeviceChannels[INPUT] * stream_.bufferSize *
8606 formatBytes( stream_.deviceFormat[INPUT] );
8608 bytes = stream_.nUserChannels[INPUT] * stream_.bufferSize *
8609 formatBytes( stream_.userFormat );
8611 if ( pa_simple_read( pah->s_rec, pulse_in, bytes, &pa_error ) < 0 ) {
8612 errorStream_ << "RtApiPulse::callbackEvent: audio read error, " <<
8613 pa_strerror( pa_error ) << ".";
8614 errorText_ = errorStream_.str();
8615 error( RtAudioError::WARNING );
8617 if ( stream_.doConvertBuffer[INPUT] ) {
8618 convertBuffer( stream_.userBuffer[INPUT],
8619 stream_.deviceBuffer,
8620 stream_.convertInfo[INPUT] );
8625 MUTEX_UNLOCK( &stream_.mutex );
8626 RtApi::tickStreamTime();
8628 if ( doStopStream == 1 )
8632 void RtApiPulse::startStream( void )
8634 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8636 if ( stream_.state == STREAM_CLOSED ) {
8637 errorText_ = "RtApiPulse::startStream(): the stream is not open!";
8638 error( RtAudioError::INVALID_USE );
8641 if ( stream_.state == STREAM_RUNNING ) {
8642 errorText_ = "RtApiPulse::startStream(): the stream is already running!";
8643 error( RtAudioError::WARNING );
8647 MUTEX_LOCK( &stream_.mutex );
8649 stream_.state = STREAM_RUNNING;
8651 pah->runnable = true;
8652 pthread_cond_signal( &pah->runnable_cv );
8653 MUTEX_UNLOCK( &stream_.mutex );
8656 void RtApiPulse::stopStream( void )
8658 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8660 if ( stream_.state == STREAM_CLOSED ) {
8661 errorText_ = "RtApiPulse::stopStream(): the stream is not open!";
8662 error( RtAudioError::INVALID_USE );
8665 if ( stream_.state == STREAM_STOPPED ) {
8666 errorText_ = "RtApiPulse::stopStream(): the stream is already stopped!";
8667 error( RtAudioError::WARNING );
8671 stream_.state = STREAM_STOPPED;
8672 MUTEX_LOCK( &stream_.mutex );
8674 if ( pah && pah->s_play ) {
8676 if ( pa_simple_drain( pah->s_play, &pa_error ) < 0 ) {
8677 errorStream_ << "RtApiPulse::stopStream: error draining output device, " <<
8678 pa_strerror( pa_error ) << ".";
8679 errorText_ = errorStream_.str();
8680 MUTEX_UNLOCK( &stream_.mutex );
8681 error( RtAudioError::SYSTEM_ERROR );
8686 stream_.state = STREAM_STOPPED;
8687 MUTEX_UNLOCK( &stream_.mutex );
8690 void RtApiPulse::abortStream( void )
8692 PulseAudioHandle *pah = static_cast<PulseAudioHandle*>( stream_.apiHandle );
8694 if ( stream_.state == STREAM_CLOSED ) {
8695 errorText_ = "RtApiPulse::abortStream(): the stream is not open!";
8696 error( RtAudioError::INVALID_USE );
8699 if ( stream_.state == STREAM_STOPPED ) {
8700 errorText_ = "RtApiPulse::abortStream(): the stream is already stopped!";
8701 error( RtAudioError::WARNING );
8705 stream_.state = STREAM_STOPPED;
8706 MUTEX_LOCK( &stream_.mutex );
8708 if ( pah && pah->s_play ) {
8710 if ( pa_simple_flush( pah->s_play, &pa_error ) < 0 ) {
8711 errorStream_ << "RtApiPulse::abortStream: error flushing output device, " <<
8712 pa_strerror( pa_error ) << ".";
8713 errorText_ = errorStream_.str();
8714 MUTEX_UNLOCK( &stream_.mutex );
8715 error( RtAudioError::SYSTEM_ERROR );
8720 stream_.state = STREAM_STOPPED;
8721 MUTEX_UNLOCK( &stream_.mutex );
8724 bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode,
8725 unsigned int channels, unsigned int firstChannel,
8726 unsigned int sampleRate, RtAudioFormat format,
8727 unsigned int *bufferSize, RtAudio::StreamOptions *options )
8729 PulseAudioHandle *pah = 0;
8730 unsigned long bufferBytes = 0;
8733 if ( device != 0 ) return false;
8734 if ( mode != INPUT && mode != OUTPUT ) return false;
8735 if ( channels != 1 && channels != 2 ) {
8736 errorText_ = "RtApiPulse::probeDeviceOpen: unsupported number of channels.";
8739 ss.channels = channels;
8741 if ( firstChannel != 0 ) return false;
8743 bool sr_found = false;
8744 for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr ) {
8745 if ( sampleRate == *sr ) {
8747 stream_.sampleRate = sampleRate;
8748 ss.rate = sampleRate;
8753 errorText_ = "RtApiPulse::probeDeviceOpen: unsupported sample rate.";
8758 for ( const rtaudio_pa_format_mapping_t *sf = supported_sampleformats;
8759 sf->rtaudio_format && sf->pa_format != PA_SAMPLE_INVALID; ++sf ) {
8760 if ( format == sf->rtaudio_format ) {
8762 stream_.userFormat = sf->rtaudio_format;
8763 stream_.deviceFormat[mode] = stream_.userFormat;
8764 ss.format = sf->pa_format;
8768 if ( !sf_found ) { // Use internal data format conversion.
8769 stream_.userFormat = format;
8770 stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
8771 ss.format = PA_SAMPLE_FLOAT32LE;
8774 // Set other stream parameters.
8775 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
8776 else stream_.userInterleaved = true;
8777 stream_.deviceInterleaved[mode] = true;
8778 stream_.nBuffers = 1;
8779 stream_.doByteSwap[mode] = false;
8780 stream_.nUserChannels[mode] = channels;
8781 stream_.nDeviceChannels[mode] = channels + firstChannel;
8782 stream_.channelOffset[mode] = 0;
8783 std::string streamName = "RtAudio";
8785 // Set flags for buffer conversion.
8786 stream_.doConvertBuffer[mode] = false;
8787 if ( stream_.userFormat != stream_.deviceFormat[mode] )
8788 stream_.doConvertBuffer[mode] = true;
8789 if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
8790 stream_.doConvertBuffer[mode] = true;
8792 // Allocate necessary internal buffers.
8793 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
8794 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
8795 if ( stream_.userBuffer[mode] == NULL ) {
8796 errorText_ = "RtApiPulse::probeDeviceOpen: error allocating user buffer memory.";
8799 stream_.bufferSize = *bufferSize;
8801 if ( stream_.doConvertBuffer[mode] ) {
8803 bool makeBuffer = true;
8804 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
8805 if ( mode == INPUT ) {
8806 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
8807 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
8808 if ( bufferBytes <= bytesOut ) makeBuffer = false;
8813 bufferBytes *= *bufferSize;
8814 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
8815 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
8816 if ( stream_.deviceBuffer == NULL ) {
8817 errorText_ = "RtApiPulse::probeDeviceOpen: error allocating device buffer memory.";
8823 stream_.device[mode] = device;
8825 // Setup the buffer conversion information structure.
8826 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
8828 if ( !stream_.apiHandle ) {
8829 PulseAudioHandle *pah = new PulseAudioHandle;
8831 errorText_ = "RtApiPulse::probeDeviceOpen: error allocating memory for handle.";
8835 stream_.apiHandle = pah;
8836 if ( pthread_cond_init( &pah->runnable_cv, NULL ) != 0 ) {
8837 errorText_ = "RtApiPulse::probeDeviceOpen: error creating condition variable.";
8841 pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8844 if ( options && !options->streamName.empty() ) streamName = options->streamName;
8847 pa_buffer_attr buffer_attr;
8848 buffer_attr.fragsize = bufferBytes;
8849 buffer_attr.maxlength = -1;
8851 pah->s_rec = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_RECORD, NULL, "Record", &ss, NULL, &buffer_attr, &error );
8852 if ( !pah->s_rec ) {
8853 errorText_ = "RtApiPulse::probeDeviceOpen: error connecting input to PulseAudio server.";
8858 pah->s_play = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_PLAYBACK, NULL, "Playback", &ss, NULL, NULL, &error );
8859 if ( !pah->s_play ) {
8860 errorText_ = "RtApiPulse::probeDeviceOpen: error connecting output to PulseAudio server.";
8868 if ( stream_.mode == UNINITIALIZED )
8869 stream_.mode = mode;
8870 else if ( stream_.mode == mode )
8873 stream_.mode = DUPLEX;
8875 if ( !stream_.callbackInfo.isRunning ) {
8876 stream_.callbackInfo.object = this;
8878 stream_.state = STREAM_STOPPED;
8879 // Set the thread attributes for joinable and realtime scheduling
8880 // priority (optional). The higher priority will only take affect
8881 // if the program is run as root or suid. Note, under Linux
8882 // processes with CAP_SYS_NICE privilege, a user can change
8883 // scheduling policy and priority (thus need not be root). See
8884 // POSIX "capabilities".
8885 pthread_attr_t attr;
8886 pthread_attr_init( &attr );
8887 pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
8888 #ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
8889 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
8890 stream_.callbackInfo.doRealtime = true;
8891 struct sched_param param;
8892 int priority = options->priority;
8893 int min = sched_get_priority_min( SCHED_RR );
8894 int max = sched_get_priority_max( SCHED_RR );
8895 if ( priority < min ) priority = min;
8896 else if ( priority > max ) priority = max;
8897 param.sched_priority = priority;
8899 // Set the policy BEFORE the priority. Otherwise it fails.
8900 pthread_attr_setschedpolicy(&attr, SCHED_RR);
8901 pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM);
8902 // This is definitely required. Otherwise it fails.
8903 pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED);
8904 pthread_attr_setschedparam(&attr, ¶m);
8907 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
8909 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
8912 stream_.callbackInfo.isRunning = true;
8913 int result = pthread_create( &pah->thread, &attr, pulseaudio_callback, (void *)&stream_.callbackInfo);
8914 pthread_attr_destroy(&attr);
8916 // Failed. Try instead with default attributes.
8917 result = pthread_create( &pah->thread, NULL, pulseaudio_callback, (void *)&stream_.callbackInfo);
8919 stream_.callbackInfo.isRunning = false;
8920 errorText_ = "RtApiPulse::probeDeviceOpen: error creating thread.";
8929 if ( pah && stream_.callbackInfo.isRunning ) {
8930 pthread_cond_destroy( &pah->runnable_cv );
8932 stream_.apiHandle = 0;
8935 for ( int i=0; i<2; i++ ) {
8936 if ( stream_.userBuffer[i] ) {
8937 free( stream_.userBuffer[i] );
8938 stream_.userBuffer[i] = 0;
8942 if ( stream_.deviceBuffer ) {
8943 free( stream_.deviceBuffer );
8944 stream_.deviceBuffer = 0;
8947 stream_.state = STREAM_CLOSED;
8951 //******************** End of __LINUX_PULSE__ *********************//
8954 #if defined(__LINUX_OSS__)
8957 #include <sys/ioctl.h>
8960 #include <sys/soundcard.h>
8964 static void *ossCallbackHandler(void * ptr);
8966 // A structure to hold various information related to the OSS API
8969 int id[2]; // device ids
8972 pthread_cond_t runnable;
8975 :triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
8978 RtApiOss :: RtApiOss()
8980 // Nothing to do here.
8983 RtApiOss :: ~RtApiOss()
8985 if ( stream_.state != STREAM_CLOSED ) closeStream();
8988 unsigned int RtApiOss :: getDeviceCount( void )
8990 int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
8991 if ( mixerfd == -1 ) {
8992 errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'.";
8993 error( RtAudioError::WARNING );
8997 oss_sysinfo sysinfo;
8998 if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) {
9000 errorText_ = "RtApiOss::getDeviceCount: error getting sysinfo, OSS version >= 4.0 is required.";
9001 error( RtAudioError::WARNING );
9006 return sysinfo.numaudios;
9009 RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device )
9011 RtAudio::DeviceInfo info;
9012 info.probed = false;
9014 int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
9015 if ( mixerfd == -1 ) {
9016 errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'.";
9017 error( RtAudioError::WARNING );
9021 oss_sysinfo sysinfo;
9022 int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
9023 if ( result == -1 ) {
9025 errorText_ = "RtApiOss::getDeviceInfo: error getting sysinfo, OSS version >= 4.0 is required.";
9026 error( RtAudioError::WARNING );
9030 unsigned nDevices = sysinfo.numaudios;
9031 if ( nDevices == 0 ) {
9033 errorText_ = "RtApiOss::getDeviceInfo: no devices found!";
9034 error( RtAudioError::INVALID_USE );
9038 if ( device >= nDevices ) {
9040 errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!";
9041 error( RtAudioError::INVALID_USE );
9045 oss_audioinfo ainfo;
9047 result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
9049 if ( result == -1 ) {
9050 errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
9051 errorText_ = errorStream_.str();
9052 error( RtAudioError::WARNING );
9057 if ( ainfo.caps & PCM_CAP_OUTPUT ) info.outputChannels = ainfo.max_channels;
9058 if ( ainfo.caps & PCM_CAP_INPUT ) info.inputChannels = ainfo.max_channels;
9059 if ( ainfo.caps & PCM_CAP_DUPLEX ) {
9060 if ( info.outputChannels > 0 && info.inputChannels > 0 && ainfo.caps & PCM_CAP_DUPLEX )
9061 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
9064 // Probe data formats ... do for input
9065 unsigned long mask = ainfo.iformats;
9066 if ( mask & AFMT_S16_LE || mask & AFMT_S16_BE )
9067 info.nativeFormats |= RTAUDIO_SINT16;
9068 if ( mask & AFMT_S8 )
9069 info.nativeFormats |= RTAUDIO_SINT8;
9070 if ( mask & AFMT_S32_LE || mask & AFMT_S32_BE )
9071 info.nativeFormats |= RTAUDIO_SINT32;
9073 if ( mask & AFMT_FLOAT )
9074 info.nativeFormats |= RTAUDIO_FLOAT32;
9076 if ( mask & AFMT_S24_LE || mask & AFMT_S24_BE )
9077 info.nativeFormats |= RTAUDIO_SINT24;
9079 // Check that we have at least one supported format
9080 if ( info.nativeFormats == 0 ) {
9081 errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio.";
9082 errorText_ = errorStream_.str();
9083 error( RtAudioError::WARNING );
9087 // Probe the supported sample rates.
9088 info.sampleRates.clear();
9089 if ( ainfo.nrates ) {
9090 for ( unsigned int i=0; i<ainfo.nrates; i++ ) {
9091 for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
9092 if ( ainfo.rates[i] == SAMPLE_RATES[k] ) {
9093 info.sampleRates.push_back( SAMPLE_RATES[k] );
9095 if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
9096 info.preferredSampleRate = SAMPLE_RATES[k];
9104 // Check min and max rate values;
9105 for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
9106 if ( ainfo.min_rate <= (int) SAMPLE_RATES[k] && ainfo.max_rate >= (int) SAMPLE_RATES[k] ) {
9107 info.sampleRates.push_back( SAMPLE_RATES[k] );
9109 if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
9110 info.preferredSampleRate = SAMPLE_RATES[k];
9115 if ( info.sampleRates.size() == 0 ) {
9116 errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ").";
9117 errorText_ = errorStream_.str();
9118 error( RtAudioError::WARNING );
9122 info.name = ainfo.name;
9129 bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
9130 unsigned int firstChannel, unsigned int sampleRate,
9131 RtAudioFormat format, unsigned int *bufferSize,
9132 RtAudio::StreamOptions *options )
9134 int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
9135 if ( mixerfd == -1 ) {
9136 errorText_ = "RtApiOss::probeDeviceOpen: error opening '/dev/mixer'.";
9140 oss_sysinfo sysinfo;
9141 int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
9142 if ( result == -1 ) {
9144 errorText_ = "RtApiOss::probeDeviceOpen: error getting sysinfo, OSS version >= 4.0 is required.";
9148 unsigned nDevices = sysinfo.numaudios;
9149 if ( nDevices == 0 ) {
9150 // This should not happen because a check is made before this function is called.
9152 errorText_ = "RtApiOss::probeDeviceOpen: no devices found!";
9156 if ( device >= nDevices ) {
9157 // This should not happen because a check is made before this function is called.
9159 errorText_ = "RtApiOss::probeDeviceOpen: device ID is invalid!";
9163 oss_audioinfo ainfo;
9165 result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
9167 if ( result == -1 ) {
9168 errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
9169 errorText_ = errorStream_.str();
9173 // Check if device supports input or output
9174 if ( ( mode == OUTPUT && !( ainfo.caps & PCM_CAP_OUTPUT ) ) ||
9175 ( mode == INPUT && !( ainfo.caps & PCM_CAP_INPUT ) ) ) {
9176 if ( mode == OUTPUT )
9177 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support output.";
9179 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support input.";
9180 errorText_ = errorStream_.str();
9185 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9186 if ( mode == OUTPUT )
9188 else { // mode == INPUT
9189 if (stream_.mode == OUTPUT && stream_.device[0] == device) {
9190 // We just set the same device for playback ... close and reopen for duplex (OSS only).
9191 close( handle->id[0] );
9193 if ( !( ainfo.caps & PCM_CAP_DUPLEX ) ) {
9194 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support duplex mode.";
9195 errorText_ = errorStream_.str();
9198 // Check that the number previously set channels is the same.
9199 if ( stream_.nUserChannels[0] != channels ) {
9200 errorStream_ << "RtApiOss::probeDeviceOpen: input/output channels must be equal for OSS duplex device (" << ainfo.name << ").";
9201 errorText_ = errorStream_.str();
9210 // Set exclusive access if specified.
9211 if ( options && options->flags & RTAUDIO_HOG_DEVICE ) flags |= O_EXCL;
9213 // Try to open the device.
9215 fd = open( ainfo.devnode, flags, 0 );
9217 if ( errno == EBUSY )
9218 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") is busy.";
9220 errorStream_ << "RtApiOss::probeDeviceOpen: error opening device (" << ainfo.name << ").";
9221 errorText_ = errorStream_.str();
9225 // For duplex operation, specifically set this mode (this doesn't seem to work).
9227 if ( flags | O_RDWR ) {
9228 result = ioctl( fd, SNDCTL_DSP_SETDUPLEX, NULL );
9229 if ( result == -1) {
9230 errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ").";
9231 errorText_ = errorStream_.str();
9237 // Check the device channel support.
9238 stream_.nUserChannels[mode] = channels;
9239 if ( ainfo.max_channels < (int)(channels + firstChannel) ) {
9241 errorStream_ << "RtApiOss::probeDeviceOpen: the device (" << ainfo.name << ") does not support requested channel parameters.";
9242 errorText_ = errorStream_.str();
9246 // Set the number of channels.
9247 int deviceChannels = channels + firstChannel;
9248 result = ioctl( fd, SNDCTL_DSP_CHANNELS, &deviceChannels );
9249 if ( result == -1 || deviceChannels < (int)(channels + firstChannel) ) {
9251 errorStream_ << "RtApiOss::probeDeviceOpen: error setting channel parameters on device (" << ainfo.name << ").";
9252 errorText_ = errorStream_.str();
9255 stream_.nDeviceChannels[mode] = deviceChannels;
9257 // Get the data format mask
9259 result = ioctl( fd, SNDCTL_DSP_GETFMTS, &mask );
9260 if ( result == -1 ) {
9262 errorStream_ << "RtApiOss::probeDeviceOpen: error getting device (" << ainfo.name << ") data formats.";
9263 errorText_ = errorStream_.str();
9267 // Determine how to set the device format.
9268 stream_.userFormat = format;
9269 int deviceFormat = -1;
9270 stream_.doByteSwap[mode] = false;
9271 if ( format == RTAUDIO_SINT8 ) {
9272 if ( mask & AFMT_S8 ) {
9273 deviceFormat = AFMT_S8;
9274 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
9277 else if ( format == RTAUDIO_SINT16 ) {
9278 if ( mask & AFMT_S16_NE ) {
9279 deviceFormat = AFMT_S16_NE;
9280 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
9282 else if ( mask & AFMT_S16_OE ) {
9283 deviceFormat = AFMT_S16_OE;
9284 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
9285 stream_.doByteSwap[mode] = true;
9288 else if ( format == RTAUDIO_SINT24 ) {
9289 if ( mask & AFMT_S24_NE ) {
9290 deviceFormat = AFMT_S24_NE;
9291 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
9293 else if ( mask & AFMT_S24_OE ) {
9294 deviceFormat = AFMT_S24_OE;
9295 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
9296 stream_.doByteSwap[mode] = true;
9299 else if ( format == RTAUDIO_SINT32 ) {
9300 if ( mask & AFMT_S32_NE ) {
9301 deviceFormat = AFMT_S32_NE;
9302 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
9304 else if ( mask & AFMT_S32_OE ) {
9305 deviceFormat = AFMT_S32_OE;
9306 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
9307 stream_.doByteSwap[mode] = true;
9311 if ( deviceFormat == -1 ) {
9312 // The user requested format is not natively supported by the device.
9313 if ( mask & AFMT_S16_NE ) {
9314 deviceFormat = AFMT_S16_NE;
9315 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
9317 else if ( mask & AFMT_S32_NE ) {
9318 deviceFormat = AFMT_S32_NE;
9319 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
9321 else if ( mask & AFMT_S24_NE ) {
9322 deviceFormat = AFMT_S24_NE;
9323 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
9325 else if ( mask & AFMT_S16_OE ) {
9326 deviceFormat = AFMT_S16_OE;
9327 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
9328 stream_.doByteSwap[mode] = true;
9330 else if ( mask & AFMT_S32_OE ) {
9331 deviceFormat = AFMT_S32_OE;
9332 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
9333 stream_.doByteSwap[mode] = true;
9335 else if ( mask & AFMT_S24_OE ) {
9336 deviceFormat = AFMT_S24_OE;
9337 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
9338 stream_.doByteSwap[mode] = true;
9340 else if ( mask & AFMT_S8) {
9341 deviceFormat = AFMT_S8;
9342 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
9346 if ( stream_.deviceFormat[mode] == 0 ) {
9347 // This really shouldn't happen ...
9349 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") data format not supported by RtAudio.";
9350 errorText_ = errorStream_.str();
9354 // Set the data format.
9355 int temp = deviceFormat;
9356 result = ioctl( fd, SNDCTL_DSP_SETFMT, &deviceFormat );
9357 if ( result == -1 || deviceFormat != temp ) {
9359 errorStream_ << "RtApiOss::probeDeviceOpen: error setting data format on device (" << ainfo.name << ").";
9360 errorText_ = errorStream_.str();
9364 // Attempt to set the buffer size. According to OSS, the minimum
9365 // number of buffers is two. The supposed minimum buffer size is 16
9366 // bytes, so that will be our lower bound. The argument to this
9367 // call is in the form 0xMMMMSSSS (hex), where the buffer size (in
9368 // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM.
9369 // We'll check the actual value used near the end of the setup
9371 int ossBufferBytes = *bufferSize * formatBytes( stream_.deviceFormat[mode] ) * deviceChannels;
9372 if ( ossBufferBytes < 16 ) ossBufferBytes = 16;
9374 if ( options ) buffers = options->numberOfBuffers;
9375 if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) buffers = 2;
9376 if ( buffers < 2 ) buffers = 3;
9377 temp = ((int) buffers << 16) + (int)( log10( (double)ossBufferBytes ) / log10( 2.0 ) );
9378 result = ioctl( fd, SNDCTL_DSP_SETFRAGMENT, &temp );
9379 if ( result == -1 ) {
9381 errorStream_ << "RtApiOss::probeDeviceOpen: error setting buffer size on device (" << ainfo.name << ").";
9382 errorText_ = errorStream_.str();
9385 stream_.nBuffers = buffers;
9387 // Save buffer size (in sample frames).
9388 *bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels );
9389 stream_.bufferSize = *bufferSize;
9391 // Set the sample rate.
9392 int srate = sampleRate;
9393 result = ioctl( fd, SNDCTL_DSP_SPEED, &srate );
9394 if ( result == -1 ) {
9396 errorStream_ << "RtApiOss::probeDeviceOpen: error setting sample rate (" << sampleRate << ") on device (" << ainfo.name << ").";
9397 errorText_ = errorStream_.str();
9401 // Verify the sample rate setup worked.
9402 if ( abs( srate - (int)sampleRate ) > 100 ) {
9404 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ").";
9405 errorText_ = errorStream_.str();
9408 stream_.sampleRate = sampleRate;
9410 if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device) {
9411 // We're doing duplex setup here.
9412 stream_.deviceFormat[0] = stream_.deviceFormat[1];
9413 stream_.nDeviceChannels[0] = deviceChannels;
9416 // Set interleaving parameters.
9417 stream_.userInterleaved = true;
9418 stream_.deviceInterleaved[mode] = true;
9419 if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
9420 stream_.userInterleaved = false;
9422 // Set flags for buffer conversion
9423 stream_.doConvertBuffer[mode] = false;
9424 if ( stream_.userFormat != stream_.deviceFormat[mode] )
9425 stream_.doConvertBuffer[mode] = true;
9426 if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
9427 stream_.doConvertBuffer[mode] = true;
9428 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
9429 stream_.nUserChannels[mode] > 1 )
9430 stream_.doConvertBuffer[mode] = true;
9432 // Allocate the stream handles if necessary and then save.
9433 if ( stream_.apiHandle == 0 ) {
9435 handle = new OssHandle;
9437 catch ( std::bad_alloc& ) {
9438 errorText_ = "RtApiOss::probeDeviceOpen: error allocating OssHandle memory.";
9442 if ( pthread_cond_init( &handle->runnable, NULL ) ) {
9443 errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable.";
9447 stream_.apiHandle = (void *) handle;
9450 handle = (OssHandle *) stream_.apiHandle;
9452 handle->id[mode] = fd;
9454 // Allocate necessary internal buffers.
9455 unsigned long bufferBytes;
9456 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
9457 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
9458 if ( stream_.userBuffer[mode] == NULL ) {
9459 errorText_ = "RtApiOss::probeDeviceOpen: error allocating user buffer memory.";
9463 if ( stream_.doConvertBuffer[mode] ) {
9465 bool makeBuffer = true;
9466 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
9467 if ( mode == INPUT ) {
9468 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
9469 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
9470 if ( bufferBytes <= bytesOut ) makeBuffer = false;
9475 bufferBytes *= *bufferSize;
9476 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
9477 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
9478 if ( stream_.deviceBuffer == NULL ) {
9479 errorText_ = "RtApiOss::probeDeviceOpen: error allocating device buffer memory.";
9485 stream_.device[mode] = device;
9486 stream_.state = STREAM_STOPPED;
9488 // Setup the buffer conversion information structure.
9489 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
9491 // Setup thread if necessary.
9492 if ( stream_.mode == OUTPUT && mode == INPUT ) {
9493 // We had already set up an output stream.
9494 stream_.mode = DUPLEX;
9495 if ( stream_.device[0] == device ) handle->id[0] = fd;
9498 stream_.mode = mode;
9500 // Setup callback thread.
9501 stream_.callbackInfo.object = (void *) this;
9503 // Set the thread attributes for joinable and realtime scheduling
9504 // priority. The higher priority will only take affect if the
9505 // program is run as root or suid.
9506 pthread_attr_t attr;
9507 pthread_attr_init( &attr );
9508 pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
9509 #ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
9510 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
9511 stream_.callbackInfo.doRealtime = true;
9512 struct sched_param param;
9513 int priority = options->priority;
9514 int min = sched_get_priority_min( SCHED_RR );
9515 int max = sched_get_priority_max( SCHED_RR );
9516 if ( priority < min ) priority = min;
9517 else if ( priority > max ) priority = max;
9518 param.sched_priority = priority;
9520 // Set the policy BEFORE the priority. Otherwise it fails.
9521 pthread_attr_setschedpolicy(&attr, SCHED_RR);
9522 pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM);
9523 // This is definitely required. Otherwise it fails.
9524 pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED);
9525 pthread_attr_setschedparam(&attr, ¶m);
9528 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
9530 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
9533 stream_.callbackInfo.isRunning = true;
9534 result = pthread_create( &stream_.callbackInfo.thread, &attr, ossCallbackHandler, &stream_.callbackInfo );
9535 pthread_attr_destroy( &attr );
9537 // Failed. Try instead with default attributes.
9538 result = pthread_create( &stream_.callbackInfo.thread, NULL, ossCallbackHandler, &stream_.callbackInfo );
9540 stream_.callbackInfo.isRunning = false;
9541 errorText_ = "RtApiOss::error creating callback thread!";
9551 pthread_cond_destroy( &handle->runnable );
9552 if ( handle->id[0] ) close( handle->id[0] );
9553 if ( handle->id[1] ) close( handle->id[1] );
9555 stream_.apiHandle = 0;
9558 for ( int i=0; i<2; i++ ) {
9559 if ( stream_.userBuffer[i] ) {
9560 free( stream_.userBuffer[i] );
9561 stream_.userBuffer[i] = 0;
9565 if ( stream_.deviceBuffer ) {
9566 free( stream_.deviceBuffer );
9567 stream_.deviceBuffer = 0;
9570 stream_.state = STREAM_CLOSED;
9574 void RtApiOss :: closeStream()
9576 if ( stream_.state == STREAM_CLOSED ) {
9577 errorText_ = "RtApiOss::closeStream(): no open stream to close!";
9578 error( RtAudioError::WARNING );
9582 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9583 stream_.callbackInfo.isRunning = false;
9584 MUTEX_LOCK( &stream_.mutex );
9585 if ( stream_.state == STREAM_STOPPED )
9586 pthread_cond_signal( &handle->runnable );
9587 MUTEX_UNLOCK( &stream_.mutex );
9588 pthread_join( stream_.callbackInfo.thread, NULL );
9590 if ( stream_.state == STREAM_RUNNING ) {
9591 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
9592 ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
9594 ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
9595 stream_.state = STREAM_STOPPED;
9599 pthread_cond_destroy( &handle->runnable );
9600 if ( handle->id[0] ) close( handle->id[0] );
9601 if ( handle->id[1] ) close( handle->id[1] );
9603 stream_.apiHandle = 0;
9606 for ( int i=0; i<2; i++ ) {
9607 if ( stream_.userBuffer[i] ) {
9608 free( stream_.userBuffer[i] );
9609 stream_.userBuffer[i] = 0;
9613 if ( stream_.deviceBuffer ) {
9614 free( stream_.deviceBuffer );
9615 stream_.deviceBuffer = 0;
9618 stream_.mode = UNINITIALIZED;
9619 stream_.state = STREAM_CLOSED;
9622 void RtApiOss :: startStream()
9625 if ( stream_.state == STREAM_RUNNING ) {
9626 errorText_ = "RtApiOss::startStream(): the stream is already running!";
9627 error( RtAudioError::WARNING );
9631 MUTEX_LOCK( &stream_.mutex );
9633 stream_.state = STREAM_RUNNING;
9635 // No need to do anything else here ... OSS automatically starts
9636 // when fed samples.
9638 MUTEX_UNLOCK( &stream_.mutex );
9640 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9641 pthread_cond_signal( &handle->runnable );
9644 void RtApiOss :: stopStream()
9647 if ( stream_.state == STREAM_STOPPED ) {
9648 errorText_ = "RtApiOss::stopStream(): the stream is already stopped!";
9649 error( RtAudioError::WARNING );
9653 MUTEX_LOCK( &stream_.mutex );
9655 // The state might change while waiting on a mutex.
9656 if ( stream_.state == STREAM_STOPPED ) {
9657 MUTEX_UNLOCK( &stream_.mutex );
9662 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9663 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
9665 // Flush the output with zeros a few times.
9668 RtAudioFormat format;
9670 if ( stream_.doConvertBuffer[0] ) {
9671 buffer = stream_.deviceBuffer;
9672 samples = stream_.bufferSize * stream_.nDeviceChannels[0];
9673 format = stream_.deviceFormat[0];
9676 buffer = stream_.userBuffer[0];
9677 samples = stream_.bufferSize * stream_.nUserChannels[0];
9678 format = stream_.userFormat;
9681 memset( buffer, 0, samples * formatBytes(format) );
9682 for ( unsigned int i=0; i<stream_.nBuffers+1; i++ ) {
9683 result = write( handle->id[0], buffer, samples * formatBytes(format) );
9684 if ( result == -1 ) {
9685 errorText_ = "RtApiOss::stopStream: audio write error.";
9686 error( RtAudioError::WARNING );
9690 result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
9691 if ( result == -1 ) {
9692 errorStream_ << "RtApiOss::stopStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
9693 errorText_ = errorStream_.str();
9696 handle->triggered = false;
9699 if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
9700 result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
9701 if ( result == -1 ) {
9702 errorStream_ << "RtApiOss::stopStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
9703 errorText_ = errorStream_.str();
9709 stream_.state = STREAM_STOPPED;
9710 MUTEX_UNLOCK( &stream_.mutex );
9712 if ( result != -1 ) return;
9713 error( RtAudioError::SYSTEM_ERROR );
9716 void RtApiOss :: abortStream()
9719 if ( stream_.state == STREAM_STOPPED ) {
9720 errorText_ = "RtApiOss::abortStream(): the stream is already stopped!";
9721 error( RtAudioError::WARNING );
9725 MUTEX_LOCK( &stream_.mutex );
9727 // The state might change while waiting on a mutex.
9728 if ( stream_.state == STREAM_STOPPED ) {
9729 MUTEX_UNLOCK( &stream_.mutex );
9734 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9735 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
9736 result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
9737 if ( result == -1 ) {
9738 errorStream_ << "RtApiOss::abortStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
9739 errorText_ = errorStream_.str();
9742 handle->triggered = false;
9745 if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
9746 result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
9747 if ( result == -1 ) {
9748 errorStream_ << "RtApiOss::abortStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
9749 errorText_ = errorStream_.str();
9755 stream_.state = STREAM_STOPPED;
9756 MUTEX_UNLOCK( &stream_.mutex );
9758 if ( result != -1 ) return;
9759 error( RtAudioError::SYSTEM_ERROR );
9762 void RtApiOss :: callbackEvent()
9764 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9765 if ( stream_.state == STREAM_STOPPED ) {
9766 MUTEX_LOCK( &stream_.mutex );
9767 pthread_cond_wait( &handle->runnable, &stream_.mutex );
9768 if ( stream_.state != STREAM_RUNNING ) {
9769 MUTEX_UNLOCK( &stream_.mutex );
9772 MUTEX_UNLOCK( &stream_.mutex );
9775 if ( stream_.state == STREAM_CLOSED ) {
9776 errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!";
9777 error( RtAudioError::WARNING );
9781 // Invoke user callback to get fresh output data.
9782 int doStopStream = 0;
9783 RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
9784 double streamTime = getStreamTime();
9785 RtAudioStreamStatus status = 0;
9786 if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
9787 status |= RTAUDIO_OUTPUT_UNDERFLOW;
9788 handle->xrun[0] = false;
9790 if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
9791 status |= RTAUDIO_INPUT_OVERFLOW;
9792 handle->xrun[1] = false;
9794 doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
9795 stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
9796 if ( doStopStream == 2 ) {
9797 this->abortStream();
9801 MUTEX_LOCK( &stream_.mutex );
9803 // The state might change while waiting on a mutex.
9804 if ( stream_.state == STREAM_STOPPED ) goto unlock;
9809 RtAudioFormat format;
9811 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
9813 // Setup parameters and do buffer conversion if necessary.
9814 if ( stream_.doConvertBuffer[0] ) {
9815 buffer = stream_.deviceBuffer;
9816 convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
9817 samples = stream_.bufferSize * stream_.nDeviceChannels[0];
9818 format = stream_.deviceFormat[0];
9821 buffer = stream_.userBuffer[0];
9822 samples = stream_.bufferSize * stream_.nUserChannels[0];
9823 format = stream_.userFormat;
9826 // Do byte swapping if necessary.
9827 if ( stream_.doByteSwap[0] )
9828 byteSwapBuffer( buffer, samples, format );
9830 if ( stream_.mode == DUPLEX && handle->triggered == false ) {
9832 ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
9833 result = write( handle->id[0], buffer, samples * formatBytes(format) );
9834 trig = PCM_ENABLE_INPUT|PCM_ENABLE_OUTPUT;
9835 ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
9836 handle->triggered = true;
9839 // Write samples to device.
9840 result = write( handle->id[0], buffer, samples * formatBytes(format) );
9842 if ( result == -1 ) {
9843 // We'll assume this is an underrun, though there isn't a
9844 // specific means for determining that.
9845 handle->xrun[0] = true;
9846 errorText_ = "RtApiOss::callbackEvent: audio write error.";
9847 error( RtAudioError::WARNING );
9848 // Continue on to input section.
9852 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
9854 // Setup parameters.
9855 if ( stream_.doConvertBuffer[1] ) {
9856 buffer = stream_.deviceBuffer;
9857 samples = stream_.bufferSize * stream_.nDeviceChannels[1];
9858 format = stream_.deviceFormat[1];
9861 buffer = stream_.userBuffer[1];
9862 samples = stream_.bufferSize * stream_.nUserChannels[1];
9863 format = stream_.userFormat;
9866 // Read samples from device.
9867 result = read( handle->id[1], buffer, samples * formatBytes(format) );
9869 if ( result == -1 ) {
9870 // We'll assume this is an overrun, though there isn't a
9871 // specific means for determining that.
9872 handle->xrun[1] = true;
9873 errorText_ = "RtApiOss::callbackEvent: audio read error.";
9874 error( RtAudioError::WARNING );
9878 // Do byte swapping if necessary.
9879 if ( stream_.doByteSwap[1] )
9880 byteSwapBuffer( buffer, samples, format );
9882 // Do buffer conversion if necessary.
9883 if ( stream_.doConvertBuffer[1] )
9884 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
9888 MUTEX_UNLOCK( &stream_.mutex );
9890 RtApi::tickStreamTime();
9891 if ( doStopStream == 1 ) this->stopStream();
9894 static void *ossCallbackHandler( void *ptr )
9896 CallbackInfo *info = (CallbackInfo *) ptr;
9897 RtApiOss *object = (RtApiOss *) info->object;
9898 bool *isRunning = &info->isRunning;
9900 #ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
9901 if (info->doRealtime) {
9902 std::cerr << "RtAudio oss: " <<
9903 (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") <<
9904 "running realtime scheduling" << std::endl;
9908 while ( *isRunning == true ) {
9909 pthread_testcancel();
9910 object->callbackEvent();
9913 pthread_exit( NULL );
9916 //******************** End of __LINUX_OSS__ *********************//
9920 // *************************************************** //
9922 // Protected common (OS-independent) RtAudio methods.
9924 // *************************************************** //
9926 // This method can be modified to control the behavior of error
9927 // message printing.
9928 void RtApi :: error( RtAudioError::Type type )
9930 errorStream_.str(""); // clear the ostringstream
9932 RtAudioErrorCallback errorCallback = (RtAudioErrorCallback) stream_.callbackInfo.errorCallback;
9933 if ( errorCallback ) {
9934 // abortStream() can generate new error messages. Ignore them. Just keep original one.
9936 if ( firstErrorOccurred_ )
9939 firstErrorOccurred_ = true;
9940 const std::string errorMessage = errorText_;
9942 if ( type != RtAudioError::WARNING && stream_.state != STREAM_STOPPED) {
9943 stream_.callbackInfo.isRunning = false; // exit from the thread
9947 errorCallback( type, errorMessage );
9948 firstErrorOccurred_ = false;
9952 if ( type == RtAudioError::WARNING && showWarnings_ == true )
9953 std::cerr << '\n' << errorText_ << "\n\n";
9954 else if ( type != RtAudioError::WARNING )
9955 throw( RtAudioError( errorText_, type ) );
9958 void RtApi :: verifyStream()
9960 if ( stream_.state == STREAM_CLOSED ) {
9961 errorText_ = "RtApi:: a stream is not open!";
9962 error( RtAudioError::INVALID_USE );
9966 void RtApi :: clearStreamInfo()
9968 stream_.mode = UNINITIALIZED;
9969 stream_.state = STREAM_CLOSED;
9970 stream_.sampleRate = 0;
9971 stream_.bufferSize = 0;
9972 stream_.nBuffers = 0;
9973 stream_.userFormat = 0;
9974 stream_.userInterleaved = true;
9975 stream_.streamTime = 0.0;
9976 stream_.apiHandle = 0;
9977 stream_.deviceBuffer = 0;
9978 stream_.callbackInfo.callback = 0;
9979 stream_.callbackInfo.userData = 0;
9980 stream_.callbackInfo.isRunning = false;
9981 stream_.callbackInfo.errorCallback = 0;
9982 for ( int i=0; i<2; i++ ) {
9983 stream_.device[i] = 11111;
9984 stream_.doConvertBuffer[i] = false;
9985 stream_.deviceInterleaved[i] = true;
9986 stream_.doByteSwap[i] = false;
9987 stream_.nUserChannels[i] = 0;
9988 stream_.nDeviceChannels[i] = 0;
9989 stream_.channelOffset[i] = 0;
9990 stream_.deviceFormat[i] = 0;
9991 stream_.latency[i] = 0;
9992 stream_.userBuffer[i] = 0;
9993 stream_.convertInfo[i].channels = 0;
9994 stream_.convertInfo[i].inJump = 0;
9995 stream_.convertInfo[i].outJump = 0;
9996 stream_.convertInfo[i].inFormat = 0;
9997 stream_.convertInfo[i].outFormat = 0;
9998 stream_.convertInfo[i].inOffset.clear();
9999 stream_.convertInfo[i].outOffset.clear();
10003 unsigned int RtApi :: formatBytes( RtAudioFormat format )
10005 if ( format == RTAUDIO_SINT16 )
10007 else if ( format == RTAUDIO_SINT32 || format == RTAUDIO_FLOAT32 )
10009 else if ( format == RTAUDIO_FLOAT64 )
10011 else if ( format == RTAUDIO_SINT24 )
10013 else if ( format == RTAUDIO_SINT8 )
10016 errorText_ = "RtApi::formatBytes: undefined format.";
10017 error( RtAudioError::WARNING );
10022 void RtApi :: setConvertInfo( StreamMode mode, unsigned int firstChannel )
10024 if ( mode == INPUT ) { // convert device to user buffer
10025 stream_.convertInfo[mode].inJump = stream_.nDeviceChannels[1];
10026 stream_.convertInfo[mode].outJump = stream_.nUserChannels[1];
10027 stream_.convertInfo[mode].inFormat = stream_.deviceFormat[1];
10028 stream_.convertInfo[mode].outFormat = stream_.userFormat;
10030 else { // convert user to device buffer
10031 stream_.convertInfo[mode].inJump = stream_.nUserChannels[0];
10032 stream_.convertInfo[mode].outJump = stream_.nDeviceChannels[0];
10033 stream_.convertInfo[mode].inFormat = stream_.userFormat;
10034 stream_.convertInfo[mode].outFormat = stream_.deviceFormat[0];
10037 if ( stream_.convertInfo[mode].inJump < stream_.convertInfo[mode].outJump )
10038 stream_.convertInfo[mode].channels = stream_.convertInfo[mode].inJump;
10040 stream_.convertInfo[mode].channels = stream_.convertInfo[mode].outJump;
10042 // Set up the interleave/deinterleave offsets.
10043 if ( stream_.deviceInterleaved[mode] != stream_.userInterleaved ) {
10044 if ( ( mode == OUTPUT && stream_.deviceInterleaved[mode] ) ||
10045 ( mode == INPUT && stream_.userInterleaved ) ) {
10046 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
10047 stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
10048 stream_.convertInfo[mode].outOffset.push_back( k );
10049 stream_.convertInfo[mode].inJump = 1;
10053 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
10054 stream_.convertInfo[mode].inOffset.push_back( k );
10055 stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
10056 stream_.convertInfo[mode].outJump = 1;
10060 else { // no (de)interleaving
10061 if ( stream_.userInterleaved ) {
10062 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
10063 stream_.convertInfo[mode].inOffset.push_back( k );
10064 stream_.convertInfo[mode].outOffset.push_back( k );
10068 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
10069 stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
10070 stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
10071 stream_.convertInfo[mode].inJump = 1;
10072 stream_.convertInfo[mode].outJump = 1;
10077 // Add channel offset.
10078 if ( firstChannel > 0 ) {
10079 if ( stream_.deviceInterleaved[mode] ) {
10080 if ( mode == OUTPUT ) {
10081 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
10082 stream_.convertInfo[mode].outOffset[k] += firstChannel;
10085 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
10086 stream_.convertInfo[mode].inOffset[k] += firstChannel;
10090 if ( mode == OUTPUT ) {
10091 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
10092 stream_.convertInfo[mode].outOffset[k] += ( firstChannel * stream_.bufferSize );
10095 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
10096 stream_.convertInfo[mode].inOffset[k] += ( firstChannel * stream_.bufferSize );
10102 void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info )
10104 // This function does format conversion, input/output channel compensation, and
10105 // data interleaving/deinterleaving. 24-bit integers are assumed to occupy
10106 // the lower three bytes of a 32-bit integer.
10108 // Clear our device buffer when in/out duplex device channels are different
10109 if ( outBuffer == stream_.deviceBuffer && stream_.mode == DUPLEX &&
10110 ( stream_.nDeviceChannels[0] < stream_.nDeviceChannels[1] ) )
10111 memset( outBuffer, 0, stream_.bufferSize * info.outJump * formatBytes( info.outFormat ) );
10114 if (info.outFormat == RTAUDIO_FLOAT64) {
10116 Float64 *out = (Float64 *)outBuffer;
10118 if (info.inFormat == RTAUDIO_SINT8) {
10119 signed char *in = (signed char *)inBuffer;
10120 scale = 1.0 / 127.5;
10121 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10122 for (j=0; j<info.channels; j++) {
10123 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
10124 out[info.outOffset[j]] += 0.5;
10125 out[info.outOffset[j]] *= scale;
10128 out += info.outJump;
10131 else if (info.inFormat == RTAUDIO_SINT16) {
10132 Int16 *in = (Int16 *)inBuffer;
10133 scale = 1.0 / 32767.5;
10134 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10135 for (j=0; j<info.channels; j++) {
10136 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
10137 out[info.outOffset[j]] += 0.5;
10138 out[info.outOffset[j]] *= scale;
10141 out += info.outJump;
10144 else if (info.inFormat == RTAUDIO_SINT24) {
10145 Int24 *in = (Int24 *)inBuffer;
10146 scale = 1.0 / 8388607.5;
10147 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10148 for (j=0; j<info.channels; j++) {
10149 out[info.outOffset[j]] = (Float64) (in[info.inOffset[j]].asInt());
10150 out[info.outOffset[j]] += 0.5;
10151 out[info.outOffset[j]] *= scale;
10154 out += info.outJump;
10157 else if (info.inFormat == RTAUDIO_SINT32) {
10158 Int32 *in = (Int32 *)inBuffer;
10159 scale = 1.0 / 2147483647.5;
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]];
10163 out[info.outOffset[j]] += 0.5;
10164 out[info.outOffset[j]] *= scale;
10167 out += info.outJump;
10170 else if (info.inFormat == RTAUDIO_FLOAT32) {
10171 Float32 *in = (Float32 *)inBuffer;
10172 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10173 for (j=0; j<info.channels; j++) {
10174 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
10177 out += info.outJump;
10180 else if (info.inFormat == RTAUDIO_FLOAT64) {
10181 // Channel compensation and/or (de)interleaving only.
10182 Float64 *in = (Float64 *)inBuffer;
10183 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10184 for (j=0; j<info.channels; j++) {
10185 out[info.outOffset[j]] = in[info.inOffset[j]];
10188 out += info.outJump;
10192 else if (info.outFormat == RTAUDIO_FLOAT32) {
10194 Float32 *out = (Float32 *)outBuffer;
10196 if (info.inFormat == RTAUDIO_SINT8) {
10197 signed char *in = (signed char *)inBuffer;
10198 scale = (Float32) ( 1.0 / 127.5 );
10199 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10200 for (j=0; j<info.channels; j++) {
10201 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
10202 out[info.outOffset[j]] += 0.5;
10203 out[info.outOffset[j]] *= scale;
10206 out += info.outJump;
10209 else if (info.inFormat == RTAUDIO_SINT16) {
10210 Int16 *in = (Int16 *)inBuffer;
10211 scale = (Float32) ( 1.0 / 32767.5 );
10212 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10213 for (j=0; j<info.channels; j++) {
10214 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
10215 out[info.outOffset[j]] += 0.5;
10216 out[info.outOffset[j]] *= scale;
10219 out += info.outJump;
10222 else if (info.inFormat == RTAUDIO_SINT24) {
10223 Int24 *in = (Int24 *)inBuffer;
10224 scale = (Float32) ( 1.0 / 8388607.5 );
10225 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10226 for (j=0; j<info.channels; j++) {
10227 out[info.outOffset[j]] = (Float32) (in[info.inOffset[j]].asInt());
10228 out[info.outOffset[j]] += 0.5;
10229 out[info.outOffset[j]] *= scale;
10232 out += info.outJump;
10235 else if (info.inFormat == RTAUDIO_SINT32) {
10236 Int32 *in = (Int32 *)inBuffer;
10237 scale = (Float32) ( 1.0 / 2147483647.5 );
10238 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10239 for (j=0; j<info.channels; j++) {
10240 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
10241 out[info.outOffset[j]] += 0.5;
10242 out[info.outOffset[j]] *= scale;
10245 out += info.outJump;
10248 else if (info.inFormat == RTAUDIO_FLOAT32) {
10249 // Channel compensation and/or (de)interleaving only.
10250 Float32 *in = (Float32 *)inBuffer;
10251 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10252 for (j=0; j<info.channels; j++) {
10253 out[info.outOffset[j]] = in[info.inOffset[j]];
10256 out += info.outJump;
10259 else if (info.inFormat == RTAUDIO_FLOAT64) {
10260 Float64 *in = (Float64 *)inBuffer;
10261 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10262 for (j=0; j<info.channels; j++) {
10263 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
10266 out += info.outJump;
10270 else if (info.outFormat == RTAUDIO_SINT32) {
10271 Int32 *out = (Int32 *)outBuffer;
10272 if (info.inFormat == RTAUDIO_SINT8) {
10273 signed char *in = (signed char *)inBuffer;
10274 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10275 for (j=0; j<info.channels; j++) {
10276 out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
10277 out[info.outOffset[j]] <<= 24;
10280 out += info.outJump;
10283 else if (info.inFormat == RTAUDIO_SINT16) {
10284 Int16 *in = (Int16 *)inBuffer;
10285 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10286 for (j=0; j<info.channels; j++) {
10287 out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
10288 out[info.outOffset[j]] <<= 16;
10291 out += info.outJump;
10294 else if (info.inFormat == RTAUDIO_SINT24) {
10295 Int24 *in = (Int24 *)inBuffer;
10296 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10297 for (j=0; j<info.channels; j++) {
10298 out[info.outOffset[j]] = (Int32) in[info.inOffset[j]].asInt();
10299 out[info.outOffset[j]] <<= 8;
10302 out += info.outJump;
10305 else if (info.inFormat == RTAUDIO_SINT32) {
10306 // Channel compensation and/or (de)interleaving only.
10307 Int32 *in = (Int32 *)inBuffer;
10308 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10309 for (j=0; j<info.channels; j++) {
10310 out[info.outOffset[j]] = in[info.inOffset[j]];
10313 out += info.outJump;
10316 else if (info.inFormat == RTAUDIO_FLOAT32) {
10317 Float32 *in = (Float32 *)inBuffer;
10318 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10319 for (j=0; j<info.channels; j++) {
10320 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
10323 out += info.outJump;
10326 else if (info.inFormat == RTAUDIO_FLOAT64) {
10327 Float64 *in = (Float64 *)inBuffer;
10328 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10329 for (j=0; j<info.channels; j++) {
10330 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
10333 out += info.outJump;
10337 else if (info.outFormat == RTAUDIO_SINT24) {
10338 Int24 *out = (Int24 *)outBuffer;
10339 if (info.inFormat == RTAUDIO_SINT8) {
10340 signed char *in = (signed char *)inBuffer;
10341 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10342 for (j=0; j<info.channels; j++) {
10343 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] << 16);
10344 //out[info.outOffset[j]] <<= 16;
10347 out += info.outJump;
10350 else if (info.inFormat == RTAUDIO_SINT16) {
10351 Int16 *in = (Int16 *)inBuffer;
10352 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10353 for (j=0; j<info.channels; j++) {
10354 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] << 8);
10355 //out[info.outOffset[j]] <<= 8;
10358 out += info.outJump;
10361 else if (info.inFormat == RTAUDIO_SINT24) {
10362 // Channel compensation and/or (de)interleaving only.
10363 Int24 *in = (Int24 *)inBuffer;
10364 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10365 for (j=0; j<info.channels; j++) {
10366 out[info.outOffset[j]] = in[info.inOffset[j]];
10369 out += info.outJump;
10372 else if (info.inFormat == RTAUDIO_SINT32) {
10373 Int32 *in = (Int32 *)inBuffer;
10374 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10375 for (j=0; j<info.channels; j++) {
10376 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] >> 8);
10377 //out[info.outOffset[j]] >>= 8;
10380 out += info.outJump;
10383 else if (info.inFormat == RTAUDIO_FLOAT32) {
10384 Float32 *in = (Float32 *)inBuffer;
10385 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10386 for (j=0; j<info.channels; j++) {
10387 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
10390 out += info.outJump;
10393 else if (info.inFormat == RTAUDIO_FLOAT64) {
10394 Float64 *in = (Float64 *)inBuffer;
10395 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10396 for (j=0; j<info.channels; j++) {
10397 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
10400 out += info.outJump;
10404 else if (info.outFormat == RTAUDIO_SINT16) {
10405 Int16 *out = (Int16 *)outBuffer;
10406 if (info.inFormat == RTAUDIO_SINT8) {
10407 signed char *in = (signed char *)inBuffer;
10408 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10409 for (j=0; j<info.channels; j++) {
10410 out[info.outOffset[j]] = (Int16) in[info.inOffset[j]];
10411 out[info.outOffset[j]] <<= 8;
10414 out += info.outJump;
10417 else if (info.inFormat == RTAUDIO_SINT16) {
10418 // Channel compensation and/or (de)interleaving only.
10419 Int16 *in = (Int16 *)inBuffer;
10420 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10421 for (j=0; j<info.channels; j++) {
10422 out[info.outOffset[j]] = in[info.inOffset[j]];
10425 out += info.outJump;
10428 else if (info.inFormat == RTAUDIO_SINT24) {
10429 Int24 *in = (Int24 *)inBuffer;
10430 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10431 for (j=0; j<info.channels; j++) {
10432 out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]].asInt() >> 8);
10435 out += info.outJump;
10438 else if (info.inFormat == RTAUDIO_SINT32) {
10439 Int32 *in = (Int32 *)inBuffer;
10440 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10441 for (j=0; j<info.channels; j++) {
10442 out[info.outOffset[j]] = (Int16) ((in[info.inOffset[j]] >> 16) & 0x0000ffff);
10445 out += info.outJump;
10448 else if (info.inFormat == RTAUDIO_FLOAT32) {
10449 Float32 *in = (Float32 *)inBuffer;
10450 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10451 for (j=0; j<info.channels; j++) {
10452 out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
10455 out += info.outJump;
10458 else if (info.inFormat == RTAUDIO_FLOAT64) {
10459 Float64 *in = (Float64 *)inBuffer;
10460 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10461 for (j=0; j<info.channels; j++) {
10462 out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
10465 out += info.outJump;
10469 else if (info.outFormat == RTAUDIO_SINT8) {
10470 signed char *out = (signed char *)outBuffer;
10471 if (info.inFormat == RTAUDIO_SINT8) {
10472 // Channel compensation and/or (de)interleaving only.
10473 signed char *in = (signed char *)inBuffer;
10474 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10475 for (j=0; j<info.channels; j++) {
10476 out[info.outOffset[j]] = in[info.inOffset[j]];
10479 out += info.outJump;
10482 if (info.inFormat == RTAUDIO_SINT16) {
10483 Int16 *in = (Int16 *)inBuffer;
10484 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10485 for (j=0; j<info.channels; j++) {
10486 out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 8) & 0x00ff);
10489 out += info.outJump;
10492 else if (info.inFormat == RTAUDIO_SINT24) {
10493 Int24 *in = (Int24 *)inBuffer;
10494 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10495 for (j=0; j<info.channels; j++) {
10496 out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]].asInt() >> 16);
10499 out += info.outJump;
10502 else if (info.inFormat == RTAUDIO_SINT32) {
10503 Int32 *in = (Int32 *)inBuffer;
10504 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10505 for (j=0; j<info.channels; j++) {
10506 out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 24) & 0x000000ff);
10509 out += info.outJump;
10512 else if (info.inFormat == RTAUDIO_FLOAT32) {
10513 Float32 *in = (Float32 *)inBuffer;
10514 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10515 for (j=0; j<info.channels; j++) {
10516 out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
10519 out += info.outJump;
10522 else if (info.inFormat == RTAUDIO_FLOAT64) {
10523 Float64 *in = (Float64 *)inBuffer;
10524 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10525 for (j=0; j<info.channels; j++) {
10526 out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
10529 out += info.outJump;
10535 //static inline uint16_t bswap_16(uint16_t x) { return (x>>8) | (x<<8); }
10536 //static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); }
10537 //static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); }
10539 void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format )
10545 if ( format == RTAUDIO_SINT16 ) {
10546 for ( unsigned int i=0; i<samples; i++ ) {
10547 // Swap 1st and 2nd bytes.
10552 // Increment 2 bytes.
10556 else if ( format == RTAUDIO_SINT32 ||
10557 format == RTAUDIO_FLOAT32 ) {
10558 for ( unsigned int i=0; i<samples; i++ ) {
10559 // Swap 1st and 4th bytes.
10564 // Swap 2nd and 3rd bytes.
10570 // Increment 3 more bytes.
10574 else if ( format == RTAUDIO_SINT24 ) {
10575 for ( unsigned int i=0; i<samples; i++ ) {
10576 // Swap 1st and 3rd bytes.
10581 // Increment 2 more bytes.
10585 else if ( format == RTAUDIO_FLOAT64 ) {
10586 for ( unsigned int i=0; i<samples; i++ ) {
10587 // Swap 1st and 8th bytes
10592 // Swap 2nd and 7th bytes
10598 // Swap 3rd and 6th bytes
10604 // Swap 4th and 5th bytes
10610 // Increment 5 more bytes.
10616 // Indentation settings for Vim and Emacs
10618 // Local Variables:
10619 // c-basic-offset: 2
10620 // indent-tabs-mode: nil
10623 // vim: et sts=2 sw=2