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. eg. Creatives ASIO driver
3225 // in that case, let's be naïve and try that instead
3226 *bufferSize = preferSize;
3227 stream_.bufferSize = *bufferSize;
3228 result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks );
3231 if ( result != ASE_OK ) {
3232 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") creating buffers.";
3233 errorText_ = errorStream_.str();
3236 buffersAllocated = true;
3237 stream_.state = STREAM_STOPPED;
3239 // Set flags for buffer conversion.
3240 stream_.doConvertBuffer[mode] = false;
3241 if ( stream_.userFormat != stream_.deviceFormat[mode] )
3242 stream_.doConvertBuffer[mode] = true;
3243 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
3244 stream_.nUserChannels[mode] > 1 )
3245 stream_.doConvertBuffer[mode] = true;
3247 // Allocate necessary internal buffers
3248 unsigned long bufferBytes;
3249 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
3250 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
3251 if ( stream_.userBuffer[mode] == NULL ) {
3252 errorText_ = "RtApiAsio::probeDeviceOpen: error allocating user buffer memory.";
3256 if ( stream_.doConvertBuffer[mode] ) {
3258 bool makeBuffer = true;
3259 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
3260 if ( isDuplexInput && stream_.deviceBuffer ) {
3261 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
3262 if ( bufferBytes <= bytesOut ) makeBuffer = false;
3266 bufferBytes *= *bufferSize;
3267 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
3268 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
3269 if ( stream_.deviceBuffer == NULL ) {
3270 errorText_ = "RtApiAsio::probeDeviceOpen: error allocating device buffer memory.";
3276 // Determine device latencies
3277 long inputLatency, outputLatency;
3278 result = ASIOGetLatencies( &inputLatency, &outputLatency );
3279 if ( result != ASE_OK ) {
3280 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting latency.";
3281 errorText_ = errorStream_.str();
3282 error( RtAudioError::WARNING); // warn but don't fail
3285 stream_.latency[0] = outputLatency;
3286 stream_.latency[1] = inputLatency;
3289 // Setup the buffer conversion information structure. We don't use
3290 // buffers to do channel offsets, so we override that parameter
3292 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );
3297 if ( !isDuplexInput ) {
3298 // the cleanup for error in the duplex input, is done by RtApi::openStream
3299 // So we clean up for single channel only
3301 if ( buffersAllocated )
3302 ASIODisposeBuffers();
3304 drivers.removeCurrentDriver();
3307 CloseHandle( handle->condition );
3308 if ( handle->bufferInfos )
3309 free( handle->bufferInfos );
3312 stream_.apiHandle = 0;
3316 if ( stream_.userBuffer[mode] ) {
3317 free( stream_.userBuffer[mode] );
3318 stream_.userBuffer[mode] = 0;
3321 if ( stream_.deviceBuffer ) {
3322 free( stream_.deviceBuffer );
3323 stream_.deviceBuffer = 0;
3328 }////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
3330 void RtApiAsio :: closeStream()
3332 if ( stream_.state == STREAM_CLOSED ) {
3333 errorText_ = "RtApiAsio::closeStream(): no open stream to close!";
3334 error( RtAudioError::WARNING );
3338 if ( stream_.state == STREAM_RUNNING ) {
3339 stream_.state = STREAM_STOPPED;
3342 ASIODisposeBuffers();
3343 drivers.removeCurrentDriver();
3345 AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
3347 CloseHandle( handle->condition );
3348 if ( handle->bufferInfos )
3349 free( handle->bufferInfos );
3351 stream_.apiHandle = 0;
3354 for ( int i=0; i<2; i++ ) {
3355 if ( stream_.userBuffer[i] ) {
3356 free( stream_.userBuffer[i] );
3357 stream_.userBuffer[i] = 0;
3361 if ( stream_.deviceBuffer ) {
3362 free( stream_.deviceBuffer );
3363 stream_.deviceBuffer = 0;
3366 stream_.mode = UNINITIALIZED;
3367 stream_.state = STREAM_CLOSED;
3370 bool stopThreadCalled = false;
3372 void RtApiAsio :: startStream()
3375 if ( stream_.state == STREAM_RUNNING ) {
3376 errorText_ = "RtApiAsio::startStream(): the stream is already running!";
3377 error( RtAudioError::WARNING );
3381 AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
3382 ASIOError result = ASIOStart();
3383 if ( result != ASE_OK ) {
3384 errorStream_ << "RtApiAsio::startStream: error (" << getAsioErrorString( result ) << ") starting device.";
3385 errorText_ = errorStream_.str();
3389 handle->drainCounter = 0;
3390 handle->internalDrain = false;
3391 ResetEvent( handle->condition );
3392 stream_.state = STREAM_RUNNING;
3396 stopThreadCalled = false;
3398 if ( result == ASE_OK ) return;
3399 error( RtAudioError::SYSTEM_ERROR );
3402 void RtApiAsio :: stopStream()
3405 if ( stream_.state == STREAM_STOPPED ) {
3406 errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!";
3407 error( RtAudioError::WARNING );
3411 AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
3412 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
3413 if ( handle->drainCounter == 0 ) {
3414 handle->drainCounter = 2;
3415 WaitForSingleObject( handle->condition, INFINITE ); // block until signaled
3419 stream_.state = STREAM_STOPPED;
3421 ASIOError result = ASIOStop();
3422 if ( result != ASE_OK ) {
3423 errorStream_ << "RtApiAsio::stopStream: error (" << getAsioErrorString( result ) << ") stopping device.";
3424 errorText_ = errorStream_.str();
3427 if ( result == ASE_OK ) return;
3428 error( RtAudioError::SYSTEM_ERROR );
3431 void RtApiAsio :: abortStream()
3434 if ( stream_.state == STREAM_STOPPED ) {
3435 errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!";
3436 error( RtAudioError::WARNING );
3440 // The following lines were commented-out because some behavior was
3441 // noted where the device buffers need to be zeroed to avoid
3442 // continuing sound, even when the device buffers are completely
3443 // disposed. So now, calling abort is the same as calling stop.
3444 // AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
3445 // handle->drainCounter = 2;
3449 // This function will be called by a spawned thread when the user
3450 // callback function signals that the stream should be stopped or
3451 // aborted. It is necessary to handle it this way because the
3452 // callbackEvent() function must return before the ASIOStop()
3453 // function will return.
3454 static unsigned __stdcall asioStopStream( void *ptr )
3456 CallbackInfo *info = (CallbackInfo *) ptr;
3457 RtApiAsio *object = (RtApiAsio *) info->object;
3459 object->stopStream();
3464 bool RtApiAsio :: callbackEvent( long bufferIndex )
3466 if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;
3467 if ( stream_.state == STREAM_CLOSED ) {
3468 errorText_ = "RtApiAsio::callbackEvent(): the stream is closed ... this shouldn't happen!";
3469 error( RtAudioError::WARNING );
3473 CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
3474 AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
3476 // Check if we were draining the stream and signal if finished.
3477 if ( handle->drainCounter > 3 ) {
3479 stream_.state = STREAM_STOPPING;
3480 if ( handle->internalDrain == false )
3481 SetEvent( handle->condition );
3482 else { // spawn a thread to stop the stream
3484 stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream,
3485 &stream_.callbackInfo, 0, &threadId );
3490 // Invoke user callback to get fresh output data UNLESS we are
3492 if ( handle->drainCounter == 0 ) {
3493 RtAudioCallback callback = (RtAudioCallback) info->callback;
3494 double streamTime = getStreamTime();
3495 RtAudioStreamStatus status = 0;
3496 if ( stream_.mode != INPUT && asioXRun == true ) {
3497 status |= RTAUDIO_OUTPUT_UNDERFLOW;
3500 if ( stream_.mode != OUTPUT && asioXRun == true ) {
3501 status |= RTAUDIO_INPUT_OVERFLOW;
3504 int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
3505 stream_.bufferSize, streamTime, status, info->userData );
3506 if ( cbReturnValue == 2 ) {
3507 stream_.state = STREAM_STOPPING;
3508 handle->drainCounter = 2;
3510 stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream,
3511 &stream_.callbackInfo, 0, &threadId );
3514 else if ( cbReturnValue == 1 ) {
3515 handle->drainCounter = 1;
3516 handle->internalDrain = true;
3520 unsigned int nChannels, bufferBytes, i, j;
3521 nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
3522 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
3524 bufferBytes = stream_.bufferSize * formatBytes( stream_.deviceFormat[0] );
3526 if ( handle->drainCounter > 1 ) { // write zeros to the output stream
3528 for ( i=0, j=0; i<nChannels; i++ ) {
3529 if ( handle->bufferInfos[i].isInput != ASIOTrue )
3530 memset( handle->bufferInfos[i].buffers[bufferIndex], 0, bufferBytes );
3534 else if ( stream_.doConvertBuffer[0] ) {
3536 convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
3537 if ( stream_.doByteSwap[0] )
3538 byteSwapBuffer( stream_.deviceBuffer,
3539 stream_.bufferSize * stream_.nDeviceChannels[0],
3540 stream_.deviceFormat[0] );
3542 for ( i=0, j=0; i<nChannels; i++ ) {
3543 if ( handle->bufferInfos[i].isInput != ASIOTrue )
3544 memcpy( handle->bufferInfos[i].buffers[bufferIndex],
3545 &stream_.deviceBuffer[j++*bufferBytes], bufferBytes );
3551 if ( stream_.doByteSwap[0] )
3552 byteSwapBuffer( stream_.userBuffer[0],
3553 stream_.bufferSize * stream_.nUserChannels[0],
3554 stream_.userFormat );
3556 for ( i=0, j=0; i<nChannels; i++ ) {
3557 if ( handle->bufferInfos[i].isInput != ASIOTrue )
3558 memcpy( handle->bufferInfos[i].buffers[bufferIndex],
3559 &stream_.userBuffer[0][bufferBytes*j++], bufferBytes );
3565 // Don't bother draining input
3566 if ( handle->drainCounter ) {
3567 handle->drainCounter++;
3571 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
3573 bufferBytes = stream_.bufferSize * formatBytes(stream_.deviceFormat[1]);
3575 if (stream_.doConvertBuffer[1]) {
3577 // Always interleave ASIO input data.
3578 for ( i=0, j=0; i<nChannels; i++ ) {
3579 if ( handle->bufferInfos[i].isInput == ASIOTrue )
3580 memcpy( &stream_.deviceBuffer[j++*bufferBytes],
3581 handle->bufferInfos[i].buffers[bufferIndex],
3585 if ( stream_.doByteSwap[1] )
3586 byteSwapBuffer( stream_.deviceBuffer,
3587 stream_.bufferSize * stream_.nDeviceChannels[1],
3588 stream_.deviceFormat[1] );
3589 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
3593 for ( i=0, j=0; i<nChannels; i++ ) {
3594 if ( handle->bufferInfos[i].isInput == ASIOTrue ) {
3595 memcpy( &stream_.userBuffer[1][bufferBytes*j++],
3596 handle->bufferInfos[i].buffers[bufferIndex],
3601 if ( stream_.doByteSwap[1] )
3602 byteSwapBuffer( stream_.userBuffer[1],
3603 stream_.bufferSize * stream_.nUserChannels[1],
3604 stream_.userFormat );
3609 // The following call was suggested by Malte Clasen. While the API
3610 // documentation indicates it should not be required, some device
3611 // drivers apparently do not function correctly without it.
3614 RtApi::tickStreamTime();
3618 static void sampleRateChanged( ASIOSampleRate sRate )
3620 // The ASIO documentation says that this usually only happens during
3621 // external sync. Audio processing is not stopped by the driver,
3622 // actual sample rate might not have even changed, maybe only the
3623 // sample rate status of an AES/EBU or S/PDIF digital input at the
3626 RtApi *object = (RtApi *) asioCallbackInfo->object;
3628 object->stopStream();
3630 catch ( RtAudioError &exception ) {
3631 std::cerr << "\nRtApiAsio: sampleRateChanged() error (" << exception.getMessage() << ")!\n" << std::endl;
3635 std::cerr << "\nRtApiAsio: driver reports sample rate changed to " << sRate << " ... stream stopped!!!\n" << std::endl;
3638 static long asioMessages( long selector, long value, void* /*message*/, double* /*opt*/ )
3642 switch( selector ) {
3643 case kAsioSelectorSupported:
3644 if ( value == kAsioResetRequest
3645 || value == kAsioEngineVersion
3646 || value == kAsioResyncRequest
3647 || value == kAsioLatenciesChanged
3648 // The following three were added for ASIO 2.0, you don't
3649 // necessarily have to support them.
3650 || value == kAsioSupportsTimeInfo
3651 || value == kAsioSupportsTimeCode
3652 || value == kAsioSupportsInputMonitor)
3655 case kAsioResetRequest:
3656 // Defer the task and perform the reset of the driver during the
3657 // next "safe" situation. You cannot reset the driver right now,
3658 // as this code is called from the driver. Reset the driver is
3659 // done by completely destruct is. I.e. ASIOStop(),
3660 // ASIODisposeBuffers(), Destruction Afterwards you initialize the
3662 std::cerr << "\nRtApiAsio: driver reset requested!!!" << std::endl;
3665 case kAsioResyncRequest:
3666 // This informs the application that the driver encountered some
3667 // non-fatal data loss. It is used for synchronization purposes
3668 // of different media. Added mainly to work around the Win16Mutex
3669 // problems in Windows 95/98 with the Windows Multimedia system,
3670 // which could lose data because the Mutex was held too long by
3671 // another thread. However a driver can issue it in other
3673 // std::cerr << "\nRtApiAsio: driver resync requested!!!" << std::endl;
3677 case kAsioLatenciesChanged:
3678 // This will inform the host application that the drivers were
3679 // latencies changed. Beware, it this does not mean that the
3680 // buffer sizes have changed! You might need to update internal
3682 std::cerr << "\nRtApiAsio: driver latency may have changed!!!" << std::endl;
3685 case kAsioEngineVersion:
3686 // Return the supported ASIO version of the host application. If
3687 // a host application does not implement this selector, ASIO 1.0
3688 // is assumed by the driver.
3691 case kAsioSupportsTimeInfo:
3692 // Informs the driver whether the
3693 // asioCallbacks.bufferSwitchTimeInfo() callback is supported.
3694 // For compatibility with ASIO 1.0 drivers the host application
3695 // should always support the "old" bufferSwitch method, too.
3698 case kAsioSupportsTimeCode:
3699 // Informs the driver whether application is interested in time
3700 // code info. If an application does not need to know about time
3701 // code, the driver has less work to do.
3708 static const char* getAsioErrorString( ASIOError result )
3716 static const Messages m[] =
3718 { ASE_NotPresent, "Hardware input or output is not present or available." },
3719 { ASE_HWMalfunction, "Hardware is malfunctioning." },
3720 { ASE_InvalidParameter, "Invalid input parameter." },
3721 { ASE_InvalidMode, "Invalid mode." },
3722 { ASE_SPNotAdvancing, "Sample position not advancing." },
3723 { ASE_NoClock, "Sample clock or rate cannot be determined or is not present." },
3724 { ASE_NoMemory, "Not enough memory to complete the request." }
3727 for ( unsigned int i = 0; i < sizeof(m)/sizeof(m[0]); ++i )
3728 if ( m[i].value == result ) return m[i].message;
3730 return "Unknown error.";
3733 //******************** End of __WINDOWS_ASIO__ *********************//
3737 #if defined(__WINDOWS_WASAPI__) // Windows WASAPI API
3739 // Authored by Marcus Tomlinson <themarcustomlinson@gmail.com>, April 2014
3740 // - Introduces support for the Windows WASAPI API
3741 // - Aims to deliver bit streams to and from hardware at the lowest possible latency, via the absolute minimum buffer sizes required
3742 // - Provides flexible stream configuration to an otherwise strict and inflexible WASAPI interface
3743 // - Includes automatic internal conversion of sample rate and buffer size between hardware and the user
3750 #include <mferror.h>
3752 #include <mftransform.h>
3753 #include <wmcodecdsp.h>
3755 #include <audioclient.h>
3757 #include <mmdeviceapi.h>
3758 #include <functiondiscoverykeys_devpkey.h>
3760 #ifndef MF_E_TRANSFORM_NEED_MORE_INPUT
3761 #define MF_E_TRANSFORM_NEED_MORE_INPUT _HRESULT_TYPEDEF_(0xc00d6d72)
3764 #ifndef MFSTARTUP_NOSOCKET
3765 #define MFSTARTUP_NOSOCKET 0x1
3769 #pragma comment( lib, "ksuser" )
3770 #pragma comment( lib, "mfplat.lib" )
3771 #pragma comment( lib, "mfuuid.lib" )
3772 #pragma comment( lib, "wmcodecdspuuid" )
3775 //=============================================================================
3777 #define SAFE_RELEASE( objectPtr )\
3780 objectPtr->Release();\
3784 typedef HANDLE ( __stdcall *TAvSetMmThreadCharacteristicsPtr )( LPCWSTR TaskName, LPDWORD TaskIndex );
3786 //-----------------------------------------------------------------------------
3788 // WASAPI dictates stream sample rate, format, channel count, and in some cases, buffer size.
3789 // Therefore we must perform all necessary conversions to user buffers in order to satisfy these
3790 // requirements. WasapiBuffer ring buffers are used between HwIn->UserIn and UserOut->HwOut to
3791 // provide intermediate storage for read / write synchronization.
3805 // sets the length of the internal ring buffer
3806 void setBufferSize( unsigned int bufferSize, unsigned int formatBytes ) {
3809 buffer_ = ( char* ) calloc( bufferSize, formatBytes );
3811 bufferSize_ = bufferSize;
3816 // attempt to push a buffer into the ring buffer at the current "in" index
3817 bool pushBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format )
3819 if ( !buffer || // incoming buffer is NULL
3820 bufferSize == 0 || // incoming buffer has no data
3821 bufferSize > bufferSize_ ) // incoming buffer too large
3826 unsigned int relOutIndex = outIndex_;
3827 unsigned int inIndexEnd = inIndex_ + bufferSize;
3828 if ( relOutIndex < inIndex_ && inIndexEnd >= bufferSize_ ) {
3829 relOutIndex += bufferSize_;
3832 // "in" index can end on the "out" index but cannot begin at it
3833 if ( inIndex_ <= relOutIndex && inIndexEnd > relOutIndex ) {
3834 return false; // not enough space between "in" index and "out" index
3837 // copy buffer from external to internal
3838 int fromZeroSize = inIndex_ + bufferSize - bufferSize_;
3839 fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize;
3840 int fromInSize = bufferSize - fromZeroSize;
3845 memcpy( &( ( char* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( char ) );
3846 memcpy( buffer_, &( ( char* ) buffer )[fromInSize], fromZeroSize * sizeof( char ) );
3848 case RTAUDIO_SINT16:
3849 memcpy( &( ( short* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( short ) );
3850 memcpy( buffer_, &( ( short* ) buffer )[fromInSize], fromZeroSize * sizeof( short ) );
3852 case RTAUDIO_SINT24:
3853 memcpy( &( ( S24* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( S24 ) );
3854 memcpy( buffer_, &( ( S24* ) buffer )[fromInSize], fromZeroSize * sizeof( S24 ) );
3856 case RTAUDIO_SINT32:
3857 memcpy( &( ( int* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( int ) );
3858 memcpy( buffer_, &( ( int* ) buffer )[fromInSize], fromZeroSize * sizeof( int ) );
3860 case RTAUDIO_FLOAT32:
3861 memcpy( &( ( float* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( float ) );
3862 memcpy( buffer_, &( ( float* ) buffer )[fromInSize], fromZeroSize * sizeof( float ) );
3864 case RTAUDIO_FLOAT64:
3865 memcpy( &( ( double* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( double ) );
3866 memcpy( buffer_, &( ( double* ) buffer )[fromInSize], fromZeroSize * sizeof( double ) );
3870 // update "in" index
3871 inIndex_ += bufferSize;
3872 inIndex_ %= bufferSize_;
3877 // attempt to pull a buffer from the ring buffer from the current "out" index
3878 bool pullBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format )
3880 if ( !buffer || // incoming buffer is NULL
3881 bufferSize == 0 || // incoming buffer has no data
3882 bufferSize > bufferSize_ ) // incoming buffer too large
3887 unsigned int relInIndex = inIndex_;
3888 unsigned int outIndexEnd = outIndex_ + bufferSize;
3889 if ( relInIndex < outIndex_ && outIndexEnd >= bufferSize_ ) {
3890 relInIndex += bufferSize_;
3893 // "out" index can begin at and end on the "in" index
3894 if ( outIndex_ < relInIndex && outIndexEnd > relInIndex ) {
3895 return false; // not enough space between "out" index and "in" index
3898 // copy buffer from internal to external
3899 int fromZeroSize = outIndex_ + bufferSize - bufferSize_;
3900 fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize;
3901 int fromOutSize = bufferSize - fromZeroSize;
3906 memcpy( buffer, &( ( char* ) buffer_ )[outIndex_], fromOutSize * sizeof( char ) );
3907 memcpy( &( ( char* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( char ) );
3909 case RTAUDIO_SINT16:
3910 memcpy( buffer, &( ( short* ) buffer_ )[outIndex_], fromOutSize * sizeof( short ) );
3911 memcpy( &( ( short* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( short ) );
3913 case RTAUDIO_SINT24:
3914 memcpy( buffer, &( ( S24* ) buffer_ )[outIndex_], fromOutSize * sizeof( S24 ) );
3915 memcpy( &( ( S24* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( S24 ) );
3917 case RTAUDIO_SINT32:
3918 memcpy( buffer, &( ( int* ) buffer_ )[outIndex_], fromOutSize * sizeof( int ) );
3919 memcpy( &( ( int* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( int ) );
3921 case RTAUDIO_FLOAT32:
3922 memcpy( buffer, &( ( float* ) buffer_ )[outIndex_], fromOutSize * sizeof( float ) );
3923 memcpy( &( ( float* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( float ) );
3925 case RTAUDIO_FLOAT64:
3926 memcpy( buffer, &( ( double* ) buffer_ )[outIndex_], fromOutSize * sizeof( double ) );
3927 memcpy( &( ( double* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( double ) );
3931 // update "out" index
3932 outIndex_ += bufferSize;
3933 outIndex_ %= bufferSize_;
3940 unsigned int bufferSize_;
3941 unsigned int inIndex_;
3942 unsigned int outIndex_;
3945 //-----------------------------------------------------------------------------
3947 // In order to satisfy WASAPI's buffer requirements, we need a means of converting sample rate
3948 // between HW and the user. The WasapiResampler class is used to perform this conversion between
3949 // HwIn->UserIn and UserOut->HwOut during the stream callback loop.
3950 class WasapiResampler
3953 WasapiResampler( bool isFloat, unsigned int bitsPerSample, unsigned int channelCount,
3954 unsigned int inSampleRate, unsigned int outSampleRate )
3955 : _bytesPerSample( bitsPerSample / 8 )
3956 , _channelCount( channelCount )
3957 , _sampleRatio( ( float ) outSampleRate / inSampleRate )
3958 , _transformUnk( NULL )
3959 , _transform( NULL )
3960 , _mediaType( NULL )
3961 , _inputMediaType( NULL )
3962 , _outputMediaType( NULL )
3964 #ifdef __IWMResamplerProps_FWD_DEFINED__
3965 , _resamplerProps( NULL )
3968 // 1. Initialization
3970 MFStartup( MF_VERSION, MFSTARTUP_NOSOCKET );
3972 // 2. Create Resampler Transform Object
3974 CoCreateInstance( CLSID_CResamplerMediaObject, NULL, CLSCTX_INPROC_SERVER,
3975 IID_IUnknown, ( void** ) &_transformUnk );
3977 _transformUnk->QueryInterface( IID_PPV_ARGS( &_transform ) );
3979 #ifdef __IWMResamplerProps_FWD_DEFINED__
3980 _transformUnk->QueryInterface( IID_PPV_ARGS( &_resamplerProps ) );
3981 _resamplerProps->SetHalfFilterLength( 60 ); // best conversion quality
3984 // 3. Specify input / output format
3986 MFCreateMediaType( &_mediaType );
3987 _mediaType->SetGUID( MF_MT_MAJOR_TYPE, MFMediaType_Audio );
3988 _mediaType->SetGUID( MF_MT_SUBTYPE, isFloat ? MFAudioFormat_Float : MFAudioFormat_PCM );
3989 _mediaType->SetUINT32( MF_MT_AUDIO_NUM_CHANNELS, channelCount );
3990 _mediaType->SetUINT32( MF_MT_AUDIO_SAMPLES_PER_SECOND, inSampleRate );
3991 _mediaType->SetUINT32( MF_MT_AUDIO_BLOCK_ALIGNMENT, _bytesPerSample * channelCount );
3992 _mediaType->SetUINT32( MF_MT_AUDIO_AVG_BYTES_PER_SECOND, _bytesPerSample * channelCount * inSampleRate );
3993 _mediaType->SetUINT32( MF_MT_AUDIO_BITS_PER_SAMPLE, bitsPerSample );
3994 _mediaType->SetUINT32( MF_MT_ALL_SAMPLES_INDEPENDENT, TRUE );
3996 MFCreateMediaType( &_inputMediaType );
3997 _mediaType->CopyAllItems( _inputMediaType );
3999 _transform->SetInputType( 0, _inputMediaType, 0 );
4001 MFCreateMediaType( &_outputMediaType );
4002 _mediaType->CopyAllItems( _outputMediaType );
4004 _outputMediaType->SetUINT32( MF_MT_AUDIO_SAMPLES_PER_SECOND, outSampleRate );
4005 _outputMediaType->SetUINT32( MF_MT_AUDIO_AVG_BYTES_PER_SECOND, _bytesPerSample * channelCount * outSampleRate );
4007 _transform->SetOutputType( 0, _outputMediaType, 0 );
4009 // 4. Send stream start messages to Resampler
4011 _transform->ProcessMessage( MFT_MESSAGE_COMMAND_FLUSH, 0 );
4012 _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_BEGIN_STREAMING, 0 );
4013 _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_START_OF_STREAM, 0 );
4018 // 8. Send stream stop messages to Resampler
4020 _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_END_OF_STREAM, 0 );
4021 _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_END_STREAMING, 0 );
4027 SAFE_RELEASE( _transformUnk );
4028 SAFE_RELEASE( _transform );
4029 SAFE_RELEASE( _mediaType );
4030 SAFE_RELEASE( _inputMediaType );
4031 SAFE_RELEASE( _outputMediaType );
4033 #ifdef __IWMResamplerProps_FWD_DEFINED__
4034 SAFE_RELEASE( _resamplerProps );
4038 void Convert( char* outBuffer, const char* inBuffer, unsigned int inSampleCount, unsigned int& outSampleCount )
4040 unsigned int inputBufferSize = _bytesPerSample * _channelCount * inSampleCount;
4041 if ( _sampleRatio == 1 )
4043 // no sample rate conversion required
4044 memcpy( outBuffer, inBuffer, inputBufferSize );
4045 outSampleCount = inSampleCount;
4049 unsigned int outputBufferSize = ( unsigned int ) ceilf( inputBufferSize * _sampleRatio ) + ( _bytesPerSample * _channelCount );
4051 IMFMediaBuffer* rInBuffer;
4052 IMFSample* rInSample;
4053 BYTE* rInByteBuffer = NULL;
4055 // 5. Create Sample object from input data
4057 MFCreateMemoryBuffer( inputBufferSize, &rInBuffer );
4059 rInBuffer->Lock( &rInByteBuffer, NULL, NULL );
4060 memcpy( rInByteBuffer, inBuffer, inputBufferSize );
4061 rInBuffer->Unlock();
4062 rInByteBuffer = NULL;
4064 rInBuffer->SetCurrentLength( inputBufferSize );
4066 MFCreateSample( &rInSample );
4067 rInSample->AddBuffer( rInBuffer );
4069 // 6. Pass input data to Resampler
4071 _transform->ProcessInput( 0, rInSample, 0 );
4073 SAFE_RELEASE( rInBuffer );
4074 SAFE_RELEASE( rInSample );
4076 // 7. Perform sample rate conversion
4078 IMFMediaBuffer* rOutBuffer = NULL;
4079 BYTE* rOutByteBuffer = NULL;
4081 MFT_OUTPUT_DATA_BUFFER rOutDataBuffer;
4083 DWORD rBytes = outputBufferSize; // maximum bytes accepted per ProcessOutput
4085 // 7.1 Create Sample object for output data
4087 memset( &rOutDataBuffer, 0, sizeof rOutDataBuffer );
4088 MFCreateSample( &( rOutDataBuffer.pSample ) );
4089 MFCreateMemoryBuffer( rBytes, &rOutBuffer );
4090 rOutDataBuffer.pSample->AddBuffer( rOutBuffer );
4091 rOutDataBuffer.dwStreamID = 0;
4092 rOutDataBuffer.dwStatus = 0;
4093 rOutDataBuffer.pEvents = NULL;
4095 // 7.2 Get output data from Resampler
4097 if ( _transform->ProcessOutput( 0, 1, &rOutDataBuffer, &rStatus ) == MF_E_TRANSFORM_NEED_MORE_INPUT )
4100 SAFE_RELEASE( rOutBuffer );
4101 SAFE_RELEASE( rOutDataBuffer.pSample );
4105 // 7.3 Write output data to outBuffer
4107 SAFE_RELEASE( rOutBuffer );
4108 rOutDataBuffer.pSample->ConvertToContiguousBuffer( &rOutBuffer );
4109 rOutBuffer->GetCurrentLength( &rBytes );
4111 rOutBuffer->Lock( &rOutByteBuffer, NULL, NULL );
4112 memcpy( outBuffer, rOutByteBuffer, rBytes );
4113 rOutBuffer->Unlock();
4114 rOutByteBuffer = NULL;
4116 outSampleCount = rBytes / _bytesPerSample / _channelCount;
4117 SAFE_RELEASE( rOutBuffer );
4118 SAFE_RELEASE( rOutDataBuffer.pSample );
4122 unsigned int _bytesPerSample;
4123 unsigned int _channelCount;
4126 IUnknown* _transformUnk;
4127 IMFTransform* _transform;
4128 IMFMediaType* _mediaType;
4129 IMFMediaType* _inputMediaType;
4130 IMFMediaType* _outputMediaType;
4132 #ifdef __IWMResamplerProps_FWD_DEFINED__
4133 IWMResamplerProps* _resamplerProps;
4137 //-----------------------------------------------------------------------------
4139 // A structure to hold various information related to the WASAPI implementation.
4142 IAudioClient* captureAudioClient;
4143 IAudioClient* renderAudioClient;
4144 IAudioCaptureClient* captureClient;
4145 IAudioRenderClient* renderClient;
4146 HANDLE captureEvent;
4150 : captureAudioClient( NULL ),
4151 renderAudioClient( NULL ),
4152 captureClient( NULL ),
4153 renderClient( NULL ),
4154 captureEvent( NULL ),
4155 renderEvent( NULL ) {}
4158 //=============================================================================
4160 RtApiWasapi::RtApiWasapi()
4161 : coInitialized_( false ), deviceEnumerator_( NULL )
4163 // WASAPI can run either apartment or multi-threaded
4164 HRESULT hr = CoInitialize( NULL );
4165 if ( !FAILED( hr ) )
4166 coInitialized_ = true;
4168 // Instantiate device enumerator
4169 hr = CoCreateInstance( __uuidof( MMDeviceEnumerator ), NULL,
4170 CLSCTX_ALL, __uuidof( IMMDeviceEnumerator ),
4171 ( void** ) &deviceEnumerator_ );
4173 // If this runs on an old Windows, it will fail. Ignore and proceed.
4175 deviceEnumerator_ = NULL;
4178 //-----------------------------------------------------------------------------
4180 RtApiWasapi::~RtApiWasapi()
4182 if ( stream_.state != STREAM_CLOSED )
4185 SAFE_RELEASE( deviceEnumerator_ );
4187 // If this object previously called CoInitialize()
4188 if ( coInitialized_ )
4192 //=============================================================================
4194 unsigned int RtApiWasapi::getDeviceCount( void )
4196 unsigned int captureDeviceCount = 0;
4197 unsigned int renderDeviceCount = 0;
4199 IMMDeviceCollection* captureDevices = NULL;
4200 IMMDeviceCollection* renderDevices = NULL;
4202 if ( !deviceEnumerator_ )
4205 // Count capture devices
4207 HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
4208 if ( FAILED( hr ) ) {
4209 errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device collection.";
4213 hr = captureDevices->GetCount( &captureDeviceCount );
4214 if ( FAILED( hr ) ) {
4215 errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device count.";
4219 // Count render devices
4220 hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
4221 if ( FAILED( hr ) ) {
4222 errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device collection.";
4226 hr = renderDevices->GetCount( &renderDeviceCount );
4227 if ( FAILED( hr ) ) {
4228 errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device count.";
4233 // release all references
4234 SAFE_RELEASE( captureDevices );
4235 SAFE_RELEASE( renderDevices );
4237 if ( errorText_.empty() )
4238 return captureDeviceCount + renderDeviceCount;
4240 error( RtAudioError::DRIVER_ERROR );
4244 //-----------------------------------------------------------------------------
4246 RtAudio::DeviceInfo RtApiWasapi::getDeviceInfo( unsigned int device )
4248 RtAudio::DeviceInfo info;
4249 unsigned int captureDeviceCount = 0;
4250 unsigned int renderDeviceCount = 0;
4251 std::string defaultDeviceName;
4252 bool isCaptureDevice = false;
4254 PROPVARIANT deviceNameProp;
4255 PROPVARIANT defaultDeviceNameProp;
4257 IMMDeviceCollection* captureDevices = NULL;
4258 IMMDeviceCollection* renderDevices = NULL;
4259 IMMDevice* devicePtr = NULL;
4260 IMMDevice* defaultDevicePtr = NULL;
4261 IAudioClient* audioClient = NULL;
4262 IPropertyStore* devicePropStore = NULL;
4263 IPropertyStore* defaultDevicePropStore = NULL;
4265 WAVEFORMATEX* deviceFormat = NULL;
4266 WAVEFORMATEX* closestMatchFormat = NULL;
4269 info.probed = false;
4271 // Count capture devices
4273 RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
4274 HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
4275 if ( FAILED( hr ) ) {
4276 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device collection.";
4280 hr = captureDevices->GetCount( &captureDeviceCount );
4281 if ( FAILED( hr ) ) {
4282 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device count.";
4286 // Count render devices
4287 hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
4288 if ( FAILED( hr ) ) {
4289 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device collection.";
4293 hr = renderDevices->GetCount( &renderDeviceCount );
4294 if ( FAILED( hr ) ) {
4295 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device count.";
4299 // validate device index
4300 if ( device >= captureDeviceCount + renderDeviceCount ) {
4301 errorText_ = "RtApiWasapi::getDeviceInfo: Invalid device index.";
4302 errorType = RtAudioError::INVALID_USE;
4306 // determine whether index falls within capture or render devices
4307 if ( device >= renderDeviceCount ) {
4308 hr = captureDevices->Item( device - renderDeviceCount, &devicePtr );
4309 if ( FAILED( hr ) ) {
4310 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device handle.";
4313 isCaptureDevice = true;
4316 hr = renderDevices->Item( device, &devicePtr );
4317 if ( FAILED( hr ) ) {
4318 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device handle.";
4321 isCaptureDevice = false;
4324 // get default device name
4325 if ( isCaptureDevice ) {
4326 hr = deviceEnumerator_->GetDefaultAudioEndpoint( eCapture, eConsole, &defaultDevicePtr );
4327 if ( FAILED( hr ) ) {
4328 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default capture device handle.";
4333 hr = deviceEnumerator_->GetDefaultAudioEndpoint( eRender, eConsole, &defaultDevicePtr );
4334 if ( FAILED( hr ) ) {
4335 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default render device handle.";
4340 hr = defaultDevicePtr->OpenPropertyStore( STGM_READ, &defaultDevicePropStore );
4341 if ( FAILED( hr ) ) {
4342 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open default device property store.";
4345 PropVariantInit( &defaultDeviceNameProp );
4347 hr = defaultDevicePropStore->GetValue( PKEY_Device_FriendlyName, &defaultDeviceNameProp );
4348 if ( FAILED( hr ) ) {
4349 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default device property: PKEY_Device_FriendlyName.";
4353 defaultDeviceName = convertCharPointerToStdString(defaultDeviceNameProp.pwszVal);
4356 hr = devicePtr->OpenPropertyStore( STGM_READ, &devicePropStore );
4357 if ( FAILED( hr ) ) {
4358 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open device property store.";
4362 PropVariantInit( &deviceNameProp );
4364 hr = devicePropStore->GetValue( PKEY_Device_FriendlyName, &deviceNameProp );
4365 if ( FAILED( hr ) ) {
4366 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device property: PKEY_Device_FriendlyName.";
4370 info.name =convertCharPointerToStdString(deviceNameProp.pwszVal);
4373 if ( isCaptureDevice ) {
4374 info.isDefaultInput = info.name == defaultDeviceName;
4375 info.isDefaultOutput = false;
4378 info.isDefaultInput = false;
4379 info.isDefaultOutput = info.name == defaultDeviceName;
4383 hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, NULL, ( void** ) &audioClient );
4384 if ( FAILED( hr ) ) {
4385 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device audio client.";
4389 hr = audioClient->GetMixFormat( &deviceFormat );
4390 if ( FAILED( hr ) ) {
4391 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device mix format.";
4395 if ( isCaptureDevice ) {
4396 info.inputChannels = deviceFormat->nChannels;
4397 info.outputChannels = 0;
4398 info.duplexChannels = 0;
4401 info.inputChannels = 0;
4402 info.outputChannels = deviceFormat->nChannels;
4403 info.duplexChannels = 0;
4407 info.sampleRates.clear();
4409 // allow support for all sample rates as we have a built-in sample rate converter
4410 for ( unsigned int i = 0; i < MAX_SAMPLE_RATES; i++ ) {
4411 info.sampleRates.push_back( SAMPLE_RATES[i] );
4413 info.preferredSampleRate = deviceFormat->nSamplesPerSec;
4416 info.nativeFormats = 0;
4418 if ( deviceFormat->wFormatTag == WAVE_FORMAT_IEEE_FLOAT ||
4419 ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
4420 ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_IEEE_FLOAT ) )
4422 if ( deviceFormat->wBitsPerSample == 32 ) {
4423 info.nativeFormats |= RTAUDIO_FLOAT32;
4425 else if ( deviceFormat->wBitsPerSample == 64 ) {
4426 info.nativeFormats |= RTAUDIO_FLOAT64;
4429 else if ( deviceFormat->wFormatTag == WAVE_FORMAT_PCM ||
4430 ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
4431 ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_PCM ) )
4433 if ( deviceFormat->wBitsPerSample == 8 ) {
4434 info.nativeFormats |= RTAUDIO_SINT8;
4436 else if ( deviceFormat->wBitsPerSample == 16 ) {
4437 info.nativeFormats |= RTAUDIO_SINT16;
4439 else if ( deviceFormat->wBitsPerSample == 24 ) {
4440 info.nativeFormats |= RTAUDIO_SINT24;
4442 else if ( deviceFormat->wBitsPerSample == 32 ) {
4443 info.nativeFormats |= RTAUDIO_SINT32;
4451 // release all references
4452 PropVariantClear( &deviceNameProp );
4453 PropVariantClear( &defaultDeviceNameProp );
4455 SAFE_RELEASE( captureDevices );
4456 SAFE_RELEASE( renderDevices );
4457 SAFE_RELEASE( devicePtr );
4458 SAFE_RELEASE( defaultDevicePtr );
4459 SAFE_RELEASE( audioClient );
4460 SAFE_RELEASE( devicePropStore );
4461 SAFE_RELEASE( defaultDevicePropStore );
4463 CoTaskMemFree( deviceFormat );
4464 CoTaskMemFree( closestMatchFormat );
4466 if ( !errorText_.empty() )
4471 //-----------------------------------------------------------------------------
4473 unsigned int RtApiWasapi::getDefaultOutputDevice( void )
4475 for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {
4476 if ( getDeviceInfo( i ).isDefaultOutput ) {
4484 //-----------------------------------------------------------------------------
4486 unsigned int RtApiWasapi::getDefaultInputDevice( void )
4488 for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {
4489 if ( getDeviceInfo( i ).isDefaultInput ) {
4497 //-----------------------------------------------------------------------------
4499 void RtApiWasapi::closeStream( void )
4501 if ( stream_.state == STREAM_CLOSED ) {
4502 errorText_ = "RtApiWasapi::closeStream: No open stream to close.";
4503 error( RtAudioError::WARNING );
4507 if ( stream_.state != STREAM_STOPPED )
4510 // clean up stream memory
4511 SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient )
4512 SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient )
4514 SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureClient )
4515 SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderClient )
4517 if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent )
4518 CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent );
4520 if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent )
4521 CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent );
4523 delete ( WasapiHandle* ) stream_.apiHandle;
4524 stream_.apiHandle = NULL;
4526 for ( int i = 0; i < 2; i++ ) {
4527 if ( stream_.userBuffer[i] ) {
4528 free( stream_.userBuffer[i] );
4529 stream_.userBuffer[i] = 0;
4533 if ( stream_.deviceBuffer ) {
4534 free( stream_.deviceBuffer );
4535 stream_.deviceBuffer = 0;
4538 // update stream state
4539 stream_.state = STREAM_CLOSED;
4542 //-----------------------------------------------------------------------------
4544 void RtApiWasapi::startStream( void )
4548 if ( stream_.state == STREAM_RUNNING ) {
4549 errorText_ = "RtApiWasapi::startStream: The stream is already running.";
4550 error( RtAudioError::WARNING );
4554 // update stream state
4555 stream_.state = STREAM_RUNNING;
4557 // create WASAPI stream thread
4558 stream_.callbackInfo.thread = ( ThreadHandle ) CreateThread( NULL, 0, runWasapiThread, this, CREATE_SUSPENDED, NULL );
4560 if ( !stream_.callbackInfo.thread ) {
4561 errorText_ = "RtApiWasapi::startStream: Unable to instantiate callback thread.";
4562 error( RtAudioError::THREAD_ERROR );
4565 SetThreadPriority( ( void* ) stream_.callbackInfo.thread, stream_.callbackInfo.priority );
4566 ResumeThread( ( void* ) stream_.callbackInfo.thread );
4570 //-----------------------------------------------------------------------------
4572 void RtApiWasapi::stopStream( void )
4576 if ( stream_.state == STREAM_STOPPED ) {
4577 errorText_ = "RtApiWasapi::stopStream: The stream is already stopped.";
4578 error( RtAudioError::WARNING );
4582 // inform stream thread by setting stream state to STREAM_STOPPING
4583 stream_.state = STREAM_STOPPING;
4585 // wait until stream thread is stopped
4586 while( stream_.state != STREAM_STOPPED ) {
4590 // Wait for the last buffer to play before stopping.
4591 Sleep( 1000 * stream_.bufferSize / stream_.sampleRate );
4593 // stop capture client if applicable
4594 if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) {
4595 HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop();
4596 if ( FAILED( hr ) ) {
4597 errorText_ = "RtApiWasapi::stopStream: Unable to stop capture stream.";
4598 error( RtAudioError::DRIVER_ERROR );
4603 // stop render client if applicable
4604 if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) {
4605 HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop();
4606 if ( FAILED( hr ) ) {
4607 errorText_ = "RtApiWasapi::stopStream: Unable to stop render stream.";
4608 error( RtAudioError::DRIVER_ERROR );
4613 // close thread handle
4614 if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
4615 errorText_ = "RtApiWasapi::stopStream: Unable to close callback thread.";
4616 error( RtAudioError::THREAD_ERROR );
4620 stream_.callbackInfo.thread = (ThreadHandle) NULL;
4623 //-----------------------------------------------------------------------------
4625 void RtApiWasapi::abortStream( void )
4629 if ( stream_.state == STREAM_STOPPED ) {
4630 errorText_ = "RtApiWasapi::abortStream: The stream is already stopped.";
4631 error( RtAudioError::WARNING );
4635 // inform stream thread by setting stream state to STREAM_STOPPING
4636 stream_.state = STREAM_STOPPING;
4638 // wait until stream thread is stopped
4639 while ( stream_.state != STREAM_STOPPED ) {
4643 // stop capture client if applicable
4644 if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) {
4645 HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop();
4646 if ( FAILED( hr ) ) {
4647 errorText_ = "RtApiWasapi::abortStream: Unable to stop capture stream.";
4648 error( RtAudioError::DRIVER_ERROR );
4653 // stop render client if applicable
4654 if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) {
4655 HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop();
4656 if ( FAILED( hr ) ) {
4657 errorText_ = "RtApiWasapi::abortStream: Unable to stop render stream.";
4658 error( RtAudioError::DRIVER_ERROR );
4663 // close thread handle
4664 if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
4665 errorText_ = "RtApiWasapi::abortStream: Unable to close callback thread.";
4666 error( RtAudioError::THREAD_ERROR );
4670 stream_.callbackInfo.thread = (ThreadHandle) NULL;
4673 //-----------------------------------------------------------------------------
4675 bool RtApiWasapi::probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
4676 unsigned int firstChannel, unsigned int sampleRate,
4677 RtAudioFormat format, unsigned int* bufferSize,
4678 RtAudio::StreamOptions* options )
4680 bool methodResult = FAILURE;
4681 unsigned int captureDeviceCount = 0;
4682 unsigned int renderDeviceCount = 0;
4684 IMMDeviceCollection* captureDevices = NULL;
4685 IMMDeviceCollection* renderDevices = NULL;
4686 IMMDevice* devicePtr = NULL;
4687 WAVEFORMATEX* deviceFormat = NULL;
4688 unsigned int bufferBytes;
4689 stream_.state = STREAM_STOPPED;
4691 // create API Handle if not already created
4692 if ( !stream_.apiHandle )
4693 stream_.apiHandle = ( void* ) new WasapiHandle();
4695 // Count capture devices
4697 RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
4698 HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
4699 if ( FAILED( hr ) ) {
4700 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device collection.";
4704 hr = captureDevices->GetCount( &captureDeviceCount );
4705 if ( FAILED( hr ) ) {
4706 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device count.";
4710 // Count render devices
4711 hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
4712 if ( FAILED( hr ) ) {
4713 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device collection.";
4717 hr = renderDevices->GetCount( &renderDeviceCount );
4718 if ( FAILED( hr ) ) {
4719 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device count.";
4723 // validate device index
4724 if ( device >= captureDeviceCount + renderDeviceCount ) {
4725 errorType = RtAudioError::INVALID_USE;
4726 errorText_ = "RtApiWasapi::probeDeviceOpen: Invalid device index.";
4730 // determine whether index falls within capture or render devices
4731 if ( device >= renderDeviceCount ) {
4732 if ( mode != INPUT ) {
4733 errorType = RtAudioError::INVALID_USE;
4734 errorText_ = "RtApiWasapi::probeDeviceOpen: Capture device selected as output device.";
4738 // retrieve captureAudioClient from devicePtr
4739 IAudioClient*& captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
4741 hr = captureDevices->Item( device - renderDeviceCount, &devicePtr );
4742 if ( FAILED( hr ) ) {
4743 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device handle.";
4747 hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
4748 NULL, ( void** ) &captureAudioClient );
4749 if ( FAILED( hr ) ) {
4750 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client.";
4754 hr = captureAudioClient->GetMixFormat( &deviceFormat );
4755 if ( FAILED( hr ) ) {
4756 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format.";
4760 stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
4761 captureAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
4764 if ( mode != OUTPUT ) {
4765 errorType = RtAudioError::INVALID_USE;
4766 errorText_ = "RtApiWasapi::probeDeviceOpen: Render device selected as input device.";
4770 // retrieve renderAudioClient from devicePtr
4771 IAudioClient*& renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
4773 hr = renderDevices->Item( device, &devicePtr );
4774 if ( FAILED( hr ) ) {
4775 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device handle.";
4779 hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
4780 NULL, ( void** ) &renderAudioClient );
4781 if ( FAILED( hr ) ) {
4782 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client.";
4786 hr = renderAudioClient->GetMixFormat( &deviceFormat );
4787 if ( FAILED( hr ) ) {
4788 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format.";
4792 stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
4793 renderAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
4797 if ( ( stream_.mode == OUTPUT && mode == INPUT ) ||
4798 ( stream_.mode == INPUT && mode == OUTPUT ) ) {
4799 stream_.mode = DUPLEX;
4802 stream_.mode = mode;
4805 stream_.device[mode] = device;
4806 stream_.doByteSwap[mode] = false;
4807 stream_.sampleRate = sampleRate;
4808 stream_.bufferSize = *bufferSize;
4809 stream_.nBuffers = 1;
4810 stream_.nUserChannels[mode] = channels;
4811 stream_.channelOffset[mode] = firstChannel;
4812 stream_.userFormat = format;
4813 stream_.deviceFormat[mode] = getDeviceInfo( device ).nativeFormats;
4815 if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
4816 stream_.userInterleaved = false;
4818 stream_.userInterleaved = true;
4819 stream_.deviceInterleaved[mode] = true;
4821 // Set flags for buffer conversion.
4822 stream_.doConvertBuffer[mode] = false;
4823 if ( stream_.userFormat != stream_.deviceFormat[mode] ||
4824 stream_.nUserChannels[0] != stream_.nDeviceChannels[0] ||
4825 stream_.nUserChannels[1] != stream_.nDeviceChannels[1] )
4826 stream_.doConvertBuffer[mode] = true;
4827 else if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
4828 stream_.nUserChannels[mode] > 1 )
4829 stream_.doConvertBuffer[mode] = true;
4831 if ( stream_.doConvertBuffer[mode] )
4832 setConvertInfo( mode, 0 );
4834 // Allocate necessary internal buffers
4835 bufferBytes = stream_.nUserChannels[mode] * stream_.bufferSize * formatBytes( stream_.userFormat );
4837 stream_.userBuffer[mode] = ( char* ) calloc( bufferBytes, 1 );
4838 if ( !stream_.userBuffer[mode] ) {
4839 errorType = RtAudioError::MEMORY_ERROR;
4840 errorText_ = "RtApiWasapi::probeDeviceOpen: Error allocating user buffer memory.";
4844 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME )
4845 stream_.callbackInfo.priority = 15;
4847 stream_.callbackInfo.priority = 0;
4849 ///! TODO: RTAUDIO_MINIMIZE_LATENCY // Provide stream buffers directly to callback
4850 ///! TODO: RTAUDIO_HOG_DEVICE // Exclusive mode
4852 methodResult = SUCCESS;
4856 SAFE_RELEASE( captureDevices );
4857 SAFE_RELEASE( renderDevices );
4858 SAFE_RELEASE( devicePtr );
4859 CoTaskMemFree( deviceFormat );
4861 // if method failed, close the stream
4862 if ( methodResult == FAILURE )
4865 if ( !errorText_.empty() )
4867 return methodResult;
4870 //=============================================================================
4872 DWORD WINAPI RtApiWasapi::runWasapiThread( void* wasapiPtr )
4875 ( ( RtApiWasapi* ) wasapiPtr )->wasapiThread();
4880 DWORD WINAPI RtApiWasapi::stopWasapiThread( void* wasapiPtr )
4883 ( ( RtApiWasapi* ) wasapiPtr )->stopStream();
4888 DWORD WINAPI RtApiWasapi::abortWasapiThread( void* wasapiPtr )
4891 ( ( RtApiWasapi* ) wasapiPtr )->abortStream();
4896 //-----------------------------------------------------------------------------
4898 void RtApiWasapi::wasapiThread()
4900 // as this is a new thread, we must CoInitialize it
4901 CoInitialize( NULL );
4905 IAudioClient* captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
4906 IAudioClient* renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
4907 IAudioCaptureClient* captureClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureClient;
4908 IAudioRenderClient* renderClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderClient;
4909 HANDLE captureEvent = ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent;
4910 HANDLE renderEvent = ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent;
4912 WAVEFORMATEX* captureFormat = NULL;
4913 WAVEFORMATEX* renderFormat = NULL;
4914 float captureSrRatio = 0.0f;
4915 float renderSrRatio = 0.0f;
4916 WasapiBuffer captureBuffer;
4917 WasapiBuffer renderBuffer;
4918 WasapiResampler* captureResampler = NULL;
4919 WasapiResampler* renderResampler = NULL;
4921 // declare local stream variables
4922 RtAudioCallback callback = ( RtAudioCallback ) stream_.callbackInfo.callback;
4923 BYTE* streamBuffer = NULL;
4924 unsigned long captureFlags = 0;
4925 unsigned int bufferFrameCount = 0;
4926 unsigned int numFramesPadding = 0;
4927 unsigned int convBufferSize = 0;
4928 bool callbackPushed = true;
4929 bool callbackPulled = false;
4930 bool callbackStopped = false;
4931 int callbackResult = 0;
4933 // convBuffer is used to store converted buffers between WASAPI and the user
4934 char* convBuffer = NULL;
4935 unsigned int convBuffSize = 0;
4936 unsigned int deviceBuffSize = 0;
4939 RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
4941 // Attempt to assign "Pro Audio" characteristic to thread
4942 HMODULE AvrtDll = LoadLibrary( (LPCTSTR) "AVRT.dll" );
4944 DWORD taskIndex = 0;
4945 TAvSetMmThreadCharacteristicsPtr AvSetMmThreadCharacteristicsPtr = ( TAvSetMmThreadCharacteristicsPtr ) GetProcAddress( AvrtDll, "AvSetMmThreadCharacteristicsW" );
4946 AvSetMmThreadCharacteristicsPtr( L"Pro Audio", &taskIndex );
4947 FreeLibrary( AvrtDll );
4950 // start capture stream if applicable
4951 if ( captureAudioClient ) {
4952 hr = captureAudioClient->GetMixFormat( &captureFormat );
4953 if ( FAILED( hr ) ) {
4954 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
4958 // init captureResampler
4959 captureResampler = new WasapiResampler( stream_.deviceFormat[INPUT] == RTAUDIO_FLOAT32 || stream_.deviceFormat[INPUT] == RTAUDIO_FLOAT64,
4960 formatBytes( stream_.deviceFormat[INPUT] ) * 8, stream_.nDeviceChannels[INPUT],
4961 captureFormat->nSamplesPerSec, stream_.sampleRate );
4963 captureSrRatio = ( ( float ) captureFormat->nSamplesPerSec / stream_.sampleRate );
4965 // initialize capture stream according to desire buffer size
4966 float desiredBufferSize = stream_.bufferSize * captureSrRatio;
4967 REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / captureFormat->nSamplesPerSec );
4969 if ( !captureClient ) {
4970 hr = captureAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
4971 AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
4972 desiredBufferPeriod,
4973 desiredBufferPeriod,
4976 if ( FAILED( hr ) ) {
4977 errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize capture audio client.";
4981 hr = captureAudioClient->GetService( __uuidof( IAudioCaptureClient ),
4982 ( void** ) &captureClient );
4983 if ( FAILED( hr ) ) {
4984 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture client handle.";
4988 // configure captureEvent to trigger on every available capture buffer
4989 captureEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
4990 if ( !captureEvent ) {
4991 errorType = RtAudioError::SYSTEM_ERROR;
4992 errorText_ = "RtApiWasapi::wasapiThread: Unable to create capture event.";
4996 hr = captureAudioClient->SetEventHandle( captureEvent );
4997 if ( FAILED( hr ) ) {
4998 errorText_ = "RtApiWasapi::wasapiThread: Unable to set capture event handle.";
5002 ( ( WasapiHandle* ) stream_.apiHandle )->captureClient = captureClient;
5003 ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent = captureEvent;
5006 unsigned int inBufferSize = 0;
5007 hr = captureAudioClient->GetBufferSize( &inBufferSize );
5008 if ( FAILED( hr ) ) {
5009 errorText_ = "RtApiWasapi::wasapiThread: Unable to get capture buffer size.";
5013 // scale outBufferSize according to stream->user sample rate ratio
5014 unsigned int outBufferSize = ( unsigned int ) ceilf( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT];
5015 inBufferSize *= stream_.nDeviceChannels[INPUT];
5017 // set captureBuffer size
5018 captureBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[INPUT] ) );
5020 // reset the capture stream
5021 hr = captureAudioClient->Reset();
5022 if ( FAILED( hr ) ) {
5023 errorText_ = "RtApiWasapi::wasapiThread: Unable to reset capture stream.";
5027 // start the capture stream
5028 hr = captureAudioClient->Start();
5029 if ( FAILED( hr ) ) {
5030 errorText_ = "RtApiWasapi::wasapiThread: Unable to start capture stream.";
5035 // start render stream if applicable
5036 if ( renderAudioClient ) {
5037 hr = renderAudioClient->GetMixFormat( &renderFormat );
5038 if ( FAILED( hr ) ) {
5039 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
5043 // init renderResampler
5044 renderResampler = new WasapiResampler( stream_.deviceFormat[OUTPUT] == RTAUDIO_FLOAT32 || stream_.deviceFormat[OUTPUT] == RTAUDIO_FLOAT64,
5045 formatBytes( stream_.deviceFormat[OUTPUT] ) * 8, stream_.nDeviceChannels[OUTPUT],
5046 stream_.sampleRate, renderFormat->nSamplesPerSec );
5048 renderSrRatio = ( ( float ) renderFormat->nSamplesPerSec / stream_.sampleRate );
5050 // initialize render stream according to desire buffer size
5051 float desiredBufferSize = stream_.bufferSize * renderSrRatio;
5052 REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / renderFormat->nSamplesPerSec );
5054 if ( !renderClient ) {
5055 hr = renderAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
5056 AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
5057 desiredBufferPeriod,
5058 desiredBufferPeriod,
5061 if ( FAILED( hr ) ) {
5062 errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize render audio client.";
5066 hr = renderAudioClient->GetService( __uuidof( IAudioRenderClient ),
5067 ( void** ) &renderClient );
5068 if ( FAILED( hr ) ) {
5069 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render client handle.";
5073 // configure renderEvent to trigger on every available render buffer
5074 renderEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
5075 if ( !renderEvent ) {
5076 errorType = RtAudioError::SYSTEM_ERROR;
5077 errorText_ = "RtApiWasapi::wasapiThread: Unable to create render event.";
5081 hr = renderAudioClient->SetEventHandle( renderEvent );
5082 if ( FAILED( hr ) ) {
5083 errorText_ = "RtApiWasapi::wasapiThread: Unable to set render event handle.";
5087 ( ( WasapiHandle* ) stream_.apiHandle )->renderClient = renderClient;
5088 ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent = renderEvent;
5091 unsigned int outBufferSize = 0;
5092 hr = renderAudioClient->GetBufferSize( &outBufferSize );
5093 if ( FAILED( hr ) ) {
5094 errorText_ = "RtApiWasapi::wasapiThread: Unable to get render buffer size.";
5098 // scale inBufferSize according to user->stream sample rate ratio
5099 unsigned int inBufferSize = ( unsigned int ) ceilf( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT];
5100 outBufferSize *= stream_.nDeviceChannels[OUTPUT];
5102 // set renderBuffer size
5103 renderBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[OUTPUT] ) );
5105 // reset the render stream
5106 hr = renderAudioClient->Reset();
5107 if ( FAILED( hr ) ) {
5108 errorText_ = "RtApiWasapi::wasapiThread: Unable to reset render stream.";
5112 // start the render stream
5113 hr = renderAudioClient->Start();
5114 if ( FAILED( hr ) ) {
5115 errorText_ = "RtApiWasapi::wasapiThread: Unable to start render stream.";
5120 // malloc buffer memory
5121 if ( stream_.mode == INPUT )
5123 using namespace std; // for ceilf
5124 convBuffSize = ( size_t ) ( ceilf( stream_.bufferSize * captureSrRatio ) ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
5125 deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
5127 else if ( stream_.mode == OUTPUT )
5129 convBuffSize = ( size_t ) ( ceilf( stream_.bufferSize * renderSrRatio ) ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
5130 deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
5132 else if ( stream_.mode == DUPLEX )
5134 convBuffSize = std::max( ( size_t ) ( ceilf( stream_.bufferSize * captureSrRatio ) ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
5135 ( size_t ) ( ceilf( stream_.bufferSize * renderSrRatio ) ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
5136 deviceBuffSize = std::max( stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
5137 stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
5140 convBuffSize *= 2; // allow overflow for *SrRatio remainders
5141 convBuffer = ( char* ) malloc( convBuffSize );
5142 stream_.deviceBuffer = ( char* ) malloc( deviceBuffSize );
5143 if ( !convBuffer || !stream_.deviceBuffer ) {
5144 errorType = RtAudioError::MEMORY_ERROR;
5145 errorText_ = "RtApiWasapi::wasapiThread: Error allocating device buffer memory.";
5149 // stream process loop
5150 while ( stream_.state != STREAM_STOPPING ) {
5151 if ( !callbackPulled ) {
5154 // 1. Pull callback buffer from inputBuffer
5155 // 2. If 1. was successful: Convert callback buffer to user sample rate and channel count
5156 // Convert callback buffer to user format
5158 if ( captureAudioClient )
5160 int samplesToPull = ( unsigned int ) floorf( stream_.bufferSize * captureSrRatio );
5161 if ( captureSrRatio != 1 )
5163 // account for remainders
5168 while ( convBufferSize < stream_.bufferSize )
5170 // Pull callback buffer from inputBuffer
5171 callbackPulled = captureBuffer.pullBuffer( convBuffer,
5172 samplesToPull * stream_.nDeviceChannels[INPUT],
5173 stream_.deviceFormat[INPUT] );
5175 if ( !callbackPulled )
5180 // Convert callback buffer to user sample rate
5181 unsigned int deviceBufferOffset = convBufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
5182 unsigned int convSamples = 0;
5184 captureResampler->Convert( stream_.deviceBuffer + deviceBufferOffset,
5189 convBufferSize += convSamples;
5190 samplesToPull = 1; // now pull one sample at a time until we have stream_.bufferSize samples
5193 if ( callbackPulled )
5195 if ( stream_.doConvertBuffer[INPUT] ) {
5196 // Convert callback buffer to user format
5197 convertBuffer( stream_.userBuffer[INPUT],
5198 stream_.deviceBuffer,
5199 stream_.convertInfo[INPUT] );
5202 // no further conversion, simple copy deviceBuffer to userBuffer
5203 memcpy( stream_.userBuffer[INPUT],
5204 stream_.deviceBuffer,
5205 stream_.bufferSize * stream_.nUserChannels[INPUT] * formatBytes( stream_.userFormat ) );
5210 // if there is no capture stream, set callbackPulled flag
5211 callbackPulled = true;
5216 // 1. Execute user callback method
5217 // 2. Handle return value from callback
5219 // if callback has not requested the stream to stop
5220 if ( callbackPulled && !callbackStopped ) {
5221 // Execute user callback method
5222 callbackResult = callback( stream_.userBuffer[OUTPUT],
5223 stream_.userBuffer[INPUT],
5226 captureFlags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY ? RTAUDIO_INPUT_OVERFLOW : 0,
5227 stream_.callbackInfo.userData );
5229 // Handle return value from callback
5230 if ( callbackResult == 1 ) {
5231 // instantiate a thread to stop this thread
5232 HANDLE threadHandle = CreateThread( NULL, 0, stopWasapiThread, this, 0, NULL );
5233 if ( !threadHandle ) {
5234 errorType = RtAudioError::THREAD_ERROR;
5235 errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream stop thread.";
5238 else if ( !CloseHandle( threadHandle ) ) {
5239 errorType = RtAudioError::THREAD_ERROR;
5240 errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream stop thread handle.";
5244 callbackStopped = true;
5246 else if ( callbackResult == 2 ) {
5247 // instantiate a thread to stop this thread
5248 HANDLE threadHandle = CreateThread( NULL, 0, abortWasapiThread, this, 0, NULL );
5249 if ( !threadHandle ) {
5250 errorType = RtAudioError::THREAD_ERROR;
5251 errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream abort thread.";
5254 else if ( !CloseHandle( threadHandle ) ) {
5255 errorType = RtAudioError::THREAD_ERROR;
5256 errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream abort thread handle.";
5260 callbackStopped = true;
5267 // 1. Convert callback buffer to stream format
5268 // 2. Convert callback buffer to stream sample rate and channel count
5269 // 3. Push callback buffer into outputBuffer
5271 if ( renderAudioClient && callbackPulled )
5273 // if the last call to renderBuffer.PushBuffer() was successful
5274 if ( callbackPushed || convBufferSize == 0 )
5276 if ( stream_.doConvertBuffer[OUTPUT] )
5278 // Convert callback buffer to stream format
5279 convertBuffer( stream_.deviceBuffer,
5280 stream_.userBuffer[OUTPUT],
5281 stream_.convertInfo[OUTPUT] );
5285 // Convert callback buffer to stream sample rate
5286 renderResampler->Convert( convBuffer,
5287 stream_.deviceBuffer,
5292 // Push callback buffer into outputBuffer
5293 callbackPushed = renderBuffer.pushBuffer( convBuffer,
5294 convBufferSize * stream_.nDeviceChannels[OUTPUT],
5295 stream_.deviceFormat[OUTPUT] );
5298 // if there is no render stream, set callbackPushed flag
5299 callbackPushed = true;
5304 // 1. Get capture buffer from stream
5305 // 2. Push capture buffer into inputBuffer
5306 // 3. If 2. was successful: Release capture buffer
5308 if ( captureAudioClient ) {
5309 // if the callback input buffer was not pulled from captureBuffer, wait for next capture event
5310 if ( !callbackPulled ) {
5311 WaitForSingleObject( captureEvent, INFINITE );
5314 // Get capture buffer from stream
5315 hr = captureClient->GetBuffer( &streamBuffer,
5317 &captureFlags, NULL, NULL );
5318 if ( FAILED( hr ) ) {
5319 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture buffer.";
5323 if ( bufferFrameCount != 0 ) {
5324 // Push capture buffer into inputBuffer
5325 if ( captureBuffer.pushBuffer( ( char* ) streamBuffer,
5326 bufferFrameCount * stream_.nDeviceChannels[INPUT],
5327 stream_.deviceFormat[INPUT] ) )
5329 // Release capture buffer
5330 hr = captureClient->ReleaseBuffer( bufferFrameCount );
5331 if ( FAILED( hr ) ) {
5332 errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
5338 // Inform WASAPI that capture was unsuccessful
5339 hr = captureClient->ReleaseBuffer( 0 );
5340 if ( FAILED( hr ) ) {
5341 errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
5348 // Inform WASAPI that capture was unsuccessful
5349 hr = captureClient->ReleaseBuffer( 0 );
5350 if ( FAILED( hr ) ) {
5351 errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
5359 // 1. Get render buffer from stream
5360 // 2. Pull next buffer from outputBuffer
5361 // 3. If 2. was successful: Fill render buffer with next buffer
5362 // Release render buffer
5364 if ( renderAudioClient ) {
5365 // if the callback output buffer was not pushed to renderBuffer, wait for next render event
5366 if ( callbackPulled && !callbackPushed ) {
5367 WaitForSingleObject( renderEvent, INFINITE );
5370 // Get render buffer from stream
5371 hr = renderAudioClient->GetBufferSize( &bufferFrameCount );
5372 if ( FAILED( hr ) ) {
5373 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer size.";
5377 hr = renderAudioClient->GetCurrentPadding( &numFramesPadding );
5378 if ( FAILED( hr ) ) {
5379 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer padding.";
5383 bufferFrameCount -= numFramesPadding;
5385 if ( bufferFrameCount != 0 ) {
5386 hr = renderClient->GetBuffer( bufferFrameCount, &streamBuffer );
5387 if ( FAILED( hr ) ) {
5388 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer.";
5392 // Pull next buffer from outputBuffer
5393 // Fill render buffer with next buffer
5394 if ( renderBuffer.pullBuffer( ( char* ) streamBuffer,
5395 bufferFrameCount * stream_.nDeviceChannels[OUTPUT],
5396 stream_.deviceFormat[OUTPUT] ) )
5398 // Release render buffer
5399 hr = renderClient->ReleaseBuffer( bufferFrameCount, 0 );
5400 if ( FAILED( hr ) ) {
5401 errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
5407 // Inform WASAPI that render was unsuccessful
5408 hr = renderClient->ReleaseBuffer( 0, 0 );
5409 if ( FAILED( hr ) ) {
5410 errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
5417 // Inform WASAPI that render was unsuccessful
5418 hr = renderClient->ReleaseBuffer( 0, 0 );
5419 if ( FAILED( hr ) ) {
5420 errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
5426 // if the callback buffer was pushed renderBuffer reset callbackPulled flag
5427 if ( callbackPushed ) {
5428 // unsetting the callbackPulled flag lets the stream know that
5429 // the audio device is ready for another callback output buffer.
5430 callbackPulled = false;
5433 RtApi::tickStreamTime();
5440 CoTaskMemFree( captureFormat );
5441 CoTaskMemFree( renderFormat );
5443 free ( convBuffer );
5444 delete renderResampler;
5445 delete captureResampler;
5449 if ( !errorText_.empty() )
5452 // update stream state
5453 stream_.state = STREAM_STOPPED;
5456 //******************** End of __WINDOWS_WASAPI__ *********************//
5460 #if defined(__WINDOWS_DS__) // Windows DirectSound API
5462 // Modified by Robin Davies, October 2005
5463 // - Improvements to DirectX pointer chasing.
5464 // - Bug fix for non-power-of-two Asio granularity used by Edirol PCR-A30.
5465 // - Auto-call CoInitialize for DSOUND and ASIO platforms.
5466 // Various revisions for RtAudio 4.0 by Gary Scavone, April 2007
5467 // Changed device query structure for RtAudio 4.0.7, January 2010
5469 #include <windows.h>
5470 #include <process.h>
5471 #include <mmsystem.h>
5475 #include <algorithm>
5477 #if defined(__MINGW32__)
5478 // missing from latest mingw winapi
5479 #define WAVE_FORMAT_96M08 0x00010000 /* 96 kHz, Mono, 8-bit */
5480 #define WAVE_FORMAT_96S08 0x00020000 /* 96 kHz, Stereo, 8-bit */
5481 #define WAVE_FORMAT_96M16 0x00040000 /* 96 kHz, Mono, 16-bit */
5482 #define WAVE_FORMAT_96S16 0x00080000 /* 96 kHz, Stereo, 16-bit */
5485 #define MINIMUM_DEVICE_BUFFER_SIZE 32768
5487 #ifdef _MSC_VER // if Microsoft Visual C++
5488 #pragma comment( lib, "winmm.lib" ) // then, auto-link winmm.lib. Otherwise, it has to be added manually.
5491 static inline DWORD dsPointerBetween( DWORD pointer, DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize )
5493 if ( pointer > bufferSize ) pointer -= bufferSize;
5494 if ( laterPointer < earlierPointer ) laterPointer += bufferSize;
5495 if ( pointer < earlierPointer ) pointer += bufferSize;
5496 return pointer >= earlierPointer && pointer < laterPointer;
5499 // A structure to hold various information related to the DirectSound
5500 // API implementation.
5502 unsigned int drainCounter; // Tracks callback counts when draining
5503 bool internalDrain; // Indicates if stop is initiated from callback or not.
5507 UINT bufferPointer[2];
5508 DWORD dsBufferSize[2];
5509 DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by.
5513 :drainCounter(0), internalDrain(false) { id[0] = 0; id[1] = 0; buffer[0] = 0; buffer[1] = 0; xrun[0] = false; xrun[1] = false; bufferPointer[0] = 0; bufferPointer[1] = 0; }
5516 // Declarations for utility functions, callbacks, and structures
5517 // specific to the DirectSound implementation.
5518 static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
5519 LPCTSTR description,
5523 static const char* getErrorString( int code );
5525 static unsigned __stdcall callbackHandler( void *ptr );
5534 : found(false) { validId[0] = false; validId[1] = false; }
5537 struct DsProbeData {
5539 std::vector<struct DsDevice>* dsDevices;
5542 RtApiDs :: RtApiDs()
5544 // Dsound will run both-threaded. If CoInitialize fails, then just
5545 // accept whatever the mainline chose for a threading model.
5546 coInitialized_ = false;
5547 HRESULT hr = CoInitialize( NULL );
5548 if ( !FAILED( hr ) ) coInitialized_ = true;
5551 RtApiDs :: ~RtApiDs()
5553 if ( stream_.state != STREAM_CLOSED ) closeStream();
5554 if ( coInitialized_ ) CoUninitialize(); // balanced call.
5557 // The DirectSound default output is always the first device.
5558 unsigned int RtApiDs :: getDefaultOutputDevice( void )
5563 // The DirectSound default input is always the first input device,
5564 // which is the first capture device enumerated.
5565 unsigned int RtApiDs :: getDefaultInputDevice( void )
5570 unsigned int RtApiDs :: getDeviceCount( void )
5572 // Set query flag for previously found devices to false, so that we
5573 // can check for any devices that have disappeared.
5574 for ( unsigned int i=0; i<dsDevices.size(); i++ )
5575 dsDevices[i].found = false;
5577 // Query DirectSound devices.
5578 struct DsProbeData probeInfo;
5579 probeInfo.isInput = false;
5580 probeInfo.dsDevices = &dsDevices;
5581 HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );
5582 if ( FAILED( result ) ) {
5583 errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!";
5584 errorText_ = errorStream_.str();
5585 error( RtAudioError::WARNING );
5588 // Query DirectSoundCapture devices.
5589 probeInfo.isInput = true;
5590 result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );
5591 if ( FAILED( result ) ) {
5592 errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!";
5593 errorText_ = errorStream_.str();
5594 error( RtAudioError::WARNING );
5597 // Clean out any devices that may have disappeared (code update submitted by Eli Zehngut).
5598 for ( unsigned int i=0; i<dsDevices.size(); ) {
5599 if ( dsDevices[i].found == false ) dsDevices.erase( dsDevices.begin() + i );
5603 return static_cast<unsigned int>(dsDevices.size());
5606 RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device )
5608 RtAudio::DeviceInfo info;
5609 info.probed = false;
5611 if ( dsDevices.size() == 0 ) {
5612 // Force a query of all devices
5614 if ( dsDevices.size() == 0 ) {
5615 errorText_ = "RtApiDs::getDeviceInfo: no devices found!";
5616 error( RtAudioError::INVALID_USE );
5621 if ( device >= dsDevices.size() ) {
5622 errorText_ = "RtApiDs::getDeviceInfo: device ID is invalid!";
5623 error( RtAudioError::INVALID_USE );
5628 if ( dsDevices[ device ].validId[0] == false ) goto probeInput;
5630 LPDIRECTSOUND output;
5632 result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );
5633 if ( FAILED( result ) ) {
5634 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!";
5635 errorText_ = errorStream_.str();
5636 error( RtAudioError::WARNING );
5640 outCaps.dwSize = sizeof( outCaps );
5641 result = output->GetCaps( &outCaps );
5642 if ( FAILED( result ) ) {
5644 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting capabilities!";
5645 errorText_ = errorStream_.str();
5646 error( RtAudioError::WARNING );
5650 // Get output channel information.
5651 info.outputChannels = ( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1;
5653 // Get sample rate information.
5654 info.sampleRates.clear();
5655 for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
5656 if ( SAMPLE_RATES[k] >= (unsigned int) outCaps.dwMinSecondarySampleRate &&
5657 SAMPLE_RATES[k] <= (unsigned int) outCaps.dwMaxSecondarySampleRate ) {
5658 info.sampleRates.push_back( SAMPLE_RATES[k] );
5660 if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
5661 info.preferredSampleRate = SAMPLE_RATES[k];
5665 // Get format information.
5666 if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT ) info.nativeFormats |= RTAUDIO_SINT16;
5667 if ( outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) info.nativeFormats |= RTAUDIO_SINT8;
5671 if ( getDefaultOutputDevice() == device )
5672 info.isDefaultOutput = true;
5674 if ( dsDevices[ device ].validId[1] == false ) {
5675 info.name = dsDevices[ device ].name;
5682 LPDIRECTSOUNDCAPTURE input;
5683 result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );
5684 if ( FAILED( result ) ) {
5685 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";
5686 errorText_ = errorStream_.str();
5687 error( RtAudioError::WARNING );
5692 inCaps.dwSize = sizeof( inCaps );
5693 result = input->GetCaps( &inCaps );
5694 if ( FAILED( result ) ) {
5696 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsDevices[ device ].name << ")!";
5697 errorText_ = errorStream_.str();
5698 error( RtAudioError::WARNING );
5702 // Get input channel information.
5703 info.inputChannels = inCaps.dwChannels;
5705 // Get sample rate and format information.
5706 std::vector<unsigned int> rates;
5707 if ( inCaps.dwChannels >= 2 ) {
5708 if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.nativeFormats |= RTAUDIO_SINT16;
5709 if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.nativeFormats |= RTAUDIO_SINT16;
5710 if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.nativeFormats |= RTAUDIO_SINT16;
5711 if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.nativeFormats |= RTAUDIO_SINT16;
5712 if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.nativeFormats |= RTAUDIO_SINT8;
5713 if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.nativeFormats |= RTAUDIO_SINT8;
5714 if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.nativeFormats |= RTAUDIO_SINT8;
5715 if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.nativeFormats |= RTAUDIO_SINT8;
5717 if ( info.nativeFormats & RTAUDIO_SINT16 ) {
5718 if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) rates.push_back( 11025 );
5719 if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) rates.push_back( 22050 );
5720 if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) rates.push_back( 44100 );
5721 if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) rates.push_back( 96000 );
5723 else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
5724 if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) rates.push_back( 11025 );
5725 if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) rates.push_back( 22050 );
5726 if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) rates.push_back( 44100 );
5727 if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) rates.push_back( 96000 );
5730 else if ( inCaps.dwChannels == 1 ) {
5731 if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.nativeFormats |= RTAUDIO_SINT16;
5732 if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.nativeFormats |= RTAUDIO_SINT16;
5733 if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.nativeFormats |= RTAUDIO_SINT16;
5734 if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.nativeFormats |= RTAUDIO_SINT16;
5735 if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.nativeFormats |= RTAUDIO_SINT8;
5736 if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.nativeFormats |= RTAUDIO_SINT8;
5737 if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.nativeFormats |= RTAUDIO_SINT8;
5738 if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.nativeFormats |= RTAUDIO_SINT8;
5740 if ( info.nativeFormats & RTAUDIO_SINT16 ) {
5741 if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) rates.push_back( 11025 );
5742 if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) rates.push_back( 22050 );
5743 if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) rates.push_back( 44100 );
5744 if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) rates.push_back( 96000 );
5746 else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
5747 if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) rates.push_back( 11025 );
5748 if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) rates.push_back( 22050 );
5749 if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) rates.push_back( 44100 );
5750 if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) rates.push_back( 96000 );
5753 else info.inputChannels = 0; // technically, this would be an error
5757 if ( info.inputChannels == 0 ) return info;
5759 // Copy the supported rates to the info structure but avoid duplication.
5761 for ( unsigned int i=0; i<rates.size(); i++ ) {
5763 for ( unsigned int j=0; j<info.sampleRates.size(); j++ ) {
5764 if ( rates[i] == info.sampleRates[j] ) {
5769 if ( found == false ) info.sampleRates.push_back( rates[i] );
5771 std::sort( info.sampleRates.begin(), info.sampleRates.end() );
5773 // If device opens for both playback and capture, we determine the channels.
5774 if ( info.outputChannels > 0 && info.inputChannels > 0 )
5775 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
5777 if ( device == 0 ) info.isDefaultInput = true;
5779 // Copy name and return.
5780 info.name = dsDevices[ device ].name;
5785 bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
5786 unsigned int firstChannel, unsigned int sampleRate,
5787 RtAudioFormat format, unsigned int *bufferSize,
5788 RtAudio::StreamOptions *options )
5790 if ( channels + firstChannel > 2 ) {
5791 errorText_ = "RtApiDs::probeDeviceOpen: DirectSound does not support more than 2 channels per device.";
5795 size_t nDevices = dsDevices.size();
5796 if ( nDevices == 0 ) {
5797 // This should not happen because a check is made before this function is called.
5798 errorText_ = "RtApiDs::probeDeviceOpen: no devices found!";
5802 if ( device >= nDevices ) {
5803 // This should not happen because a check is made before this function is called.
5804 errorText_ = "RtApiDs::probeDeviceOpen: device ID is invalid!";
5808 if ( mode == OUTPUT ) {
5809 if ( dsDevices[ device ].validId[0] == false ) {
5810 errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support output!";
5811 errorText_ = errorStream_.str();
5815 else { // mode == INPUT
5816 if ( dsDevices[ device ].validId[1] == false ) {
5817 errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support input!";
5818 errorText_ = errorStream_.str();
5823 // According to a note in PortAudio, using GetDesktopWindow()
5824 // instead of GetForegroundWindow() is supposed to avoid problems
5825 // that occur when the application's window is not the foreground
5826 // window. Also, if the application window closes before the
5827 // DirectSound buffer, DirectSound can crash. In the past, I had
5828 // problems when using GetDesktopWindow() but it seems fine now
5829 // (January 2010). I'll leave it commented here.
5830 // HWND hWnd = GetForegroundWindow();
5831 HWND hWnd = GetDesktopWindow();
5833 // Check the numberOfBuffers parameter and limit the lowest value to
5834 // two. This is a judgement call and a value of two is probably too
5835 // low for capture, but it should work for playback.
5837 if ( options ) nBuffers = options->numberOfBuffers;
5838 if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) nBuffers = 2;
5839 if ( nBuffers < 2 ) nBuffers = 3;
5841 // Check the lower range of the user-specified buffer size and set
5842 // (arbitrarily) to a lower bound of 32.
5843 if ( *bufferSize < 32 ) *bufferSize = 32;
5845 // Create the wave format structure. The data format setting will
5846 // be determined later.
5847 WAVEFORMATEX waveFormat;
5848 ZeroMemory( &waveFormat, sizeof(WAVEFORMATEX) );
5849 waveFormat.wFormatTag = WAVE_FORMAT_PCM;
5850 waveFormat.nChannels = channels + firstChannel;
5851 waveFormat.nSamplesPerSec = (unsigned long) sampleRate;
5853 // Determine the device buffer size. By default, we'll use the value
5854 // defined above (32K), but we will grow it to make allowances for
5855 // very large software buffer sizes.
5856 DWORD dsBufferSize = MINIMUM_DEVICE_BUFFER_SIZE;
5857 DWORD dsPointerLeadTime = 0;
5859 void *ohandle = 0, *bhandle = 0;
5861 if ( mode == OUTPUT ) {
5863 LPDIRECTSOUND output;
5864 result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );
5865 if ( FAILED( result ) ) {
5866 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!";
5867 errorText_ = errorStream_.str();
5872 outCaps.dwSize = sizeof( outCaps );
5873 result = output->GetCaps( &outCaps );
5874 if ( FAILED( result ) ) {
5876 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting capabilities (" << dsDevices[ device ].name << ")!";
5877 errorText_ = errorStream_.str();
5881 // Check channel information.
5882 if ( channels + firstChannel == 2 && !( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ) {
5883 errorStream_ << "RtApiDs::getDeviceInfo: the output device (" << dsDevices[ device ].name << ") does not support stereo playback.";
5884 errorText_ = errorStream_.str();
5888 // Check format information. Use 16-bit format unless not
5889 // supported or user requests 8-bit.
5890 if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT &&
5891 !( format == RTAUDIO_SINT8 && outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) ) {
5892 waveFormat.wBitsPerSample = 16;
5893 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
5896 waveFormat.wBitsPerSample = 8;
5897 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
5899 stream_.userFormat = format;
5901 // Update wave format structure and buffer information.
5902 waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
5903 waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
5904 dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
5906 // If the user wants an even bigger buffer, increase the device buffer size accordingly.
5907 while ( dsPointerLeadTime * 2U > dsBufferSize )
5910 // Set cooperative level to DSSCL_EXCLUSIVE ... sound stops when window focus changes.
5911 // result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE );
5912 // Set cooperative level to DSSCL_PRIORITY ... sound remains when window focus changes.
5913 result = output->SetCooperativeLevel( hWnd, DSSCL_PRIORITY );
5914 if ( FAILED( result ) ) {
5916 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting cooperative level (" << dsDevices[ device ].name << ")!";
5917 errorText_ = errorStream_.str();
5921 // Even though we will write to the secondary buffer, we need to
5922 // access the primary buffer to set the correct output format
5923 // (since the default is 8-bit, 22 kHz!). Setup the DS primary
5924 // buffer description.
5925 DSBUFFERDESC bufferDescription;
5926 ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
5927 bufferDescription.dwSize = sizeof( DSBUFFERDESC );
5928 bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER;
5930 // Obtain the primary buffer
5931 LPDIRECTSOUNDBUFFER buffer;
5932 result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
5933 if ( FAILED( result ) ) {
5935 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") accessing primary buffer (" << dsDevices[ device ].name << ")!";
5936 errorText_ = errorStream_.str();
5940 // Set the primary DS buffer sound format.
5941 result = buffer->SetFormat( &waveFormat );
5942 if ( FAILED( result ) ) {
5944 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting primary buffer format (" << dsDevices[ device ].name << ")!";
5945 errorText_ = errorStream_.str();
5949 // Setup the secondary DS buffer description.
5950 ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
5951 bufferDescription.dwSize = sizeof( DSBUFFERDESC );
5952 bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
5953 DSBCAPS_GLOBALFOCUS |
5954 DSBCAPS_GETCURRENTPOSITION2 |
5955 DSBCAPS_LOCHARDWARE ); // Force hardware mixing
5956 bufferDescription.dwBufferBytes = dsBufferSize;
5957 bufferDescription.lpwfxFormat = &waveFormat;
5959 // Try to create the secondary DS buffer. If that doesn't work,
5960 // try to use software mixing. Otherwise, there's a problem.
5961 result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
5962 if ( FAILED( result ) ) {
5963 bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
5964 DSBCAPS_GLOBALFOCUS |
5965 DSBCAPS_GETCURRENTPOSITION2 |
5966 DSBCAPS_LOCSOFTWARE ); // Force software mixing
5967 result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
5968 if ( FAILED( result ) ) {
5970 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating secondary buffer (" << dsDevices[ device ].name << ")!";
5971 errorText_ = errorStream_.str();
5976 // Get the buffer size ... might be different from what we specified.
5978 dsbcaps.dwSize = sizeof( DSBCAPS );
5979 result = buffer->GetCaps( &dsbcaps );
5980 if ( FAILED( result ) ) {
5983 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";
5984 errorText_ = errorStream_.str();
5988 dsBufferSize = dsbcaps.dwBufferBytes;
5990 // Lock the DS buffer
5993 result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );
5994 if ( FAILED( result ) ) {
5997 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking buffer (" << dsDevices[ device ].name << ")!";
5998 errorText_ = errorStream_.str();
6002 // Zero the DS buffer
6003 ZeroMemory( audioPtr, dataLen );
6005 // Unlock the DS buffer
6006 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
6007 if ( FAILED( result ) ) {
6010 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking buffer (" << dsDevices[ device ].name << ")!";
6011 errorText_ = errorStream_.str();
6015 ohandle = (void *) output;
6016 bhandle = (void *) buffer;
6019 if ( mode == INPUT ) {
6021 LPDIRECTSOUNDCAPTURE input;
6022 result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );
6023 if ( FAILED( result ) ) {
6024 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";
6025 errorText_ = errorStream_.str();
6030 inCaps.dwSize = sizeof( inCaps );
6031 result = input->GetCaps( &inCaps );
6032 if ( FAILED( result ) ) {
6034 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsDevices[ device ].name << ")!";
6035 errorText_ = errorStream_.str();
6039 // Check channel information.
6040 if ( inCaps.dwChannels < channels + firstChannel ) {
6041 errorText_ = "RtApiDs::getDeviceInfo: the input device does not support requested input channels.";
6045 // Check format information. Use 16-bit format unless user
6047 DWORD deviceFormats;
6048 if ( channels + firstChannel == 2 ) {
6049 deviceFormats = WAVE_FORMAT_1S08 | WAVE_FORMAT_2S08 | WAVE_FORMAT_4S08 | WAVE_FORMAT_96S08;
6050 if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
6051 waveFormat.wBitsPerSample = 8;
6052 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
6054 else { // assume 16-bit is supported
6055 waveFormat.wBitsPerSample = 16;
6056 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
6059 else { // channel == 1
6060 deviceFormats = WAVE_FORMAT_1M08 | WAVE_FORMAT_2M08 | WAVE_FORMAT_4M08 | WAVE_FORMAT_96M08;
6061 if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
6062 waveFormat.wBitsPerSample = 8;
6063 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
6065 else { // assume 16-bit is supported
6066 waveFormat.wBitsPerSample = 16;
6067 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
6070 stream_.userFormat = format;
6072 // Update wave format structure and buffer information.
6073 waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
6074 waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
6075 dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
6077 // If the user wants an even bigger buffer, increase the device buffer size accordingly.
6078 while ( dsPointerLeadTime * 2U > dsBufferSize )
6081 // Setup the secondary DS buffer description.
6082 DSCBUFFERDESC bufferDescription;
6083 ZeroMemory( &bufferDescription, sizeof( DSCBUFFERDESC ) );
6084 bufferDescription.dwSize = sizeof( DSCBUFFERDESC );
6085 bufferDescription.dwFlags = 0;
6086 bufferDescription.dwReserved = 0;
6087 bufferDescription.dwBufferBytes = dsBufferSize;
6088 bufferDescription.lpwfxFormat = &waveFormat;
6090 // Create the capture buffer.
6091 LPDIRECTSOUNDCAPTUREBUFFER buffer;
6092 result = input->CreateCaptureBuffer( &bufferDescription, &buffer, NULL );
6093 if ( FAILED( result ) ) {
6095 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating input buffer (" << dsDevices[ device ].name << ")!";
6096 errorText_ = errorStream_.str();
6100 // Get the buffer size ... might be different from what we specified.
6102 dscbcaps.dwSize = sizeof( DSCBCAPS );
6103 result = buffer->GetCaps( &dscbcaps );
6104 if ( FAILED( result ) ) {
6107 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";
6108 errorText_ = errorStream_.str();
6112 dsBufferSize = dscbcaps.dwBufferBytes;
6114 // NOTE: We could have a problem here if this is a duplex stream
6115 // and the play and capture hardware buffer sizes are different
6116 // (I'm actually not sure if that is a problem or not).
6117 // Currently, we are not verifying that.
6119 // Lock the capture buffer
6122 result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );
6123 if ( FAILED( result ) ) {
6126 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking input buffer (" << dsDevices[ device ].name << ")!";
6127 errorText_ = errorStream_.str();
6132 ZeroMemory( audioPtr, dataLen );
6134 // Unlock the buffer
6135 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
6136 if ( FAILED( result ) ) {
6139 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking input buffer (" << dsDevices[ device ].name << ")!";
6140 errorText_ = errorStream_.str();
6144 ohandle = (void *) input;
6145 bhandle = (void *) buffer;
6148 // Set various stream parameters
6149 DsHandle *handle = 0;
6150 stream_.nDeviceChannels[mode] = channels + firstChannel;
6151 stream_.nUserChannels[mode] = channels;
6152 stream_.bufferSize = *bufferSize;
6153 stream_.channelOffset[mode] = firstChannel;
6154 stream_.deviceInterleaved[mode] = true;
6155 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
6156 else stream_.userInterleaved = true;
6158 // Set flag for buffer conversion
6159 stream_.doConvertBuffer[mode] = false;
6160 if (stream_.nUserChannels[mode] != stream_.nDeviceChannels[mode])
6161 stream_.doConvertBuffer[mode] = true;
6162 if (stream_.userFormat != stream_.deviceFormat[mode])
6163 stream_.doConvertBuffer[mode] = true;
6164 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
6165 stream_.nUserChannels[mode] > 1 )
6166 stream_.doConvertBuffer[mode] = true;
6168 // Allocate necessary internal buffers
6169 long bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
6170 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
6171 if ( stream_.userBuffer[mode] == NULL ) {
6172 errorText_ = "RtApiDs::probeDeviceOpen: error allocating user buffer memory.";
6176 if ( stream_.doConvertBuffer[mode] ) {
6178 bool makeBuffer = true;
6179 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
6180 if ( mode == INPUT ) {
6181 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
6182 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
6183 if ( bufferBytes <= (long) bytesOut ) makeBuffer = false;
6188 bufferBytes *= *bufferSize;
6189 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
6190 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
6191 if ( stream_.deviceBuffer == NULL ) {
6192 errorText_ = "RtApiDs::probeDeviceOpen: error allocating device buffer memory.";
6198 // Allocate our DsHandle structures for the stream.
6199 if ( stream_.apiHandle == 0 ) {
6201 handle = new DsHandle;
6203 catch ( std::bad_alloc& ) {
6204 errorText_ = "RtApiDs::probeDeviceOpen: error allocating AsioHandle memory.";
6208 // Create a manual-reset event.
6209 handle->condition = CreateEvent( NULL, // no security
6210 TRUE, // manual-reset
6211 FALSE, // non-signaled initially
6213 stream_.apiHandle = (void *) handle;
6216 handle = (DsHandle *) stream_.apiHandle;
6217 handle->id[mode] = ohandle;
6218 handle->buffer[mode] = bhandle;
6219 handle->dsBufferSize[mode] = dsBufferSize;
6220 handle->dsPointerLeadTime[mode] = dsPointerLeadTime;
6222 stream_.device[mode] = device;
6223 stream_.state = STREAM_STOPPED;
6224 if ( stream_.mode == OUTPUT && mode == INPUT )
6225 // We had already set up an output stream.
6226 stream_.mode = DUPLEX;
6228 stream_.mode = mode;
6229 stream_.nBuffers = nBuffers;
6230 stream_.sampleRate = sampleRate;
6232 // Setup the buffer conversion information structure.
6233 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
6235 // Setup the callback thread.
6236 if ( stream_.callbackInfo.isRunning == false ) {
6238 stream_.callbackInfo.isRunning = true;
6239 stream_.callbackInfo.object = (void *) this;
6240 stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &callbackHandler,
6241 &stream_.callbackInfo, 0, &threadId );
6242 if ( stream_.callbackInfo.thread == 0 ) {
6243 errorText_ = "RtApiDs::probeDeviceOpen: error creating callback thread!";
6247 // Boost DS thread priority
6248 SetThreadPriority( (HANDLE) stream_.callbackInfo.thread, THREAD_PRIORITY_HIGHEST );
6254 if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
6255 LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
6256 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6257 if ( buffer ) buffer->Release();
6260 if ( handle->buffer[1] ) {
6261 LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
6262 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6263 if ( buffer ) buffer->Release();
6266 CloseHandle( handle->condition );
6268 stream_.apiHandle = 0;
6271 for ( int i=0; i<2; i++ ) {
6272 if ( stream_.userBuffer[i] ) {
6273 free( stream_.userBuffer[i] );
6274 stream_.userBuffer[i] = 0;
6278 if ( stream_.deviceBuffer ) {
6279 free( stream_.deviceBuffer );
6280 stream_.deviceBuffer = 0;
6283 stream_.state = STREAM_CLOSED;
6287 void RtApiDs :: closeStream()
6289 if ( stream_.state == STREAM_CLOSED ) {
6290 errorText_ = "RtApiDs::closeStream(): no open stream to close!";
6291 error( RtAudioError::WARNING );
6295 // Stop the callback thread.
6296 stream_.callbackInfo.isRunning = false;
6297 WaitForSingleObject( (HANDLE) stream_.callbackInfo.thread, INFINITE );
6298 CloseHandle( (HANDLE) stream_.callbackInfo.thread );
6300 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6302 if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
6303 LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
6304 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6311 if ( handle->buffer[1] ) {
6312 LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
6313 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6320 CloseHandle( handle->condition );
6322 stream_.apiHandle = 0;
6325 for ( int i=0; i<2; i++ ) {
6326 if ( stream_.userBuffer[i] ) {
6327 free( stream_.userBuffer[i] );
6328 stream_.userBuffer[i] = 0;
6332 if ( stream_.deviceBuffer ) {
6333 free( stream_.deviceBuffer );
6334 stream_.deviceBuffer = 0;
6337 stream_.mode = UNINITIALIZED;
6338 stream_.state = STREAM_CLOSED;
6341 void RtApiDs :: startStream()
6344 if ( stream_.state == STREAM_RUNNING ) {
6345 errorText_ = "RtApiDs::startStream(): the stream is already running!";
6346 error( RtAudioError::WARNING );
6350 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6352 // Increase scheduler frequency on lesser windows (a side-effect of
6353 // increasing timer accuracy). On greater windows (Win2K or later),
6354 // this is already in effect.
6355 timeBeginPeriod( 1 );
6357 buffersRolling = false;
6358 duplexPrerollBytes = 0;
6360 if ( stream_.mode == DUPLEX ) {
6361 // 0.5 seconds of silence in DUPLEX mode while the devices spin up and synchronize.
6362 duplexPrerollBytes = (int) ( 0.5 * stream_.sampleRate * formatBytes( stream_.deviceFormat[1] ) * stream_.nDeviceChannels[1] );
6366 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
6368 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6369 result = buffer->Play( 0, 0, DSBPLAY_LOOPING );
6370 if ( FAILED( result ) ) {
6371 errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting output buffer!";
6372 errorText_ = errorStream_.str();
6377 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
6379 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6380 result = buffer->Start( DSCBSTART_LOOPING );
6381 if ( FAILED( result ) ) {
6382 errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting input buffer!";
6383 errorText_ = errorStream_.str();
6388 handle->drainCounter = 0;
6389 handle->internalDrain = false;
6390 ResetEvent( handle->condition );
6391 stream_.state = STREAM_RUNNING;
6394 if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );
6397 void RtApiDs :: stopStream()
6400 if ( stream_.state == STREAM_STOPPED ) {
6401 errorText_ = "RtApiDs::stopStream(): the stream is already stopped!";
6402 error( RtAudioError::WARNING );
6409 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6410 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
6411 if ( handle->drainCounter == 0 ) {
6412 handle->drainCounter = 2;
6413 WaitForSingleObject( handle->condition, INFINITE ); // block until signaled
6416 stream_.state = STREAM_STOPPED;
6418 MUTEX_LOCK( &stream_.mutex );
6420 // Stop the buffer and clear memory
6421 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6422 result = buffer->Stop();
6423 if ( FAILED( result ) ) {
6424 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping output buffer!";
6425 errorText_ = errorStream_.str();
6429 // Lock the buffer and clear it so that if we start to play again,
6430 // we won't have old data playing.
6431 result = buffer->Lock( 0, handle->dsBufferSize[0], &audioPtr, &dataLen, NULL, NULL, 0 );
6432 if ( FAILED( result ) ) {
6433 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking output buffer!";
6434 errorText_ = errorStream_.str();
6438 // Zero the DS buffer
6439 ZeroMemory( audioPtr, dataLen );
6441 // Unlock the DS buffer
6442 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
6443 if ( FAILED( result ) ) {
6444 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking output buffer!";
6445 errorText_ = errorStream_.str();
6449 // If we start playing again, we must begin at beginning of buffer.
6450 handle->bufferPointer[0] = 0;
6453 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
6454 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6458 stream_.state = STREAM_STOPPED;
6460 if ( stream_.mode != DUPLEX )
6461 MUTEX_LOCK( &stream_.mutex );
6463 result = buffer->Stop();
6464 if ( FAILED( result ) ) {
6465 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping input buffer!";
6466 errorText_ = errorStream_.str();
6470 // Lock the buffer and clear it so that if we start to play again,
6471 // we won't have old data playing.
6472 result = buffer->Lock( 0, handle->dsBufferSize[1], &audioPtr, &dataLen, NULL, NULL, 0 );
6473 if ( FAILED( result ) ) {
6474 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking input buffer!";
6475 errorText_ = errorStream_.str();
6479 // Zero the DS buffer
6480 ZeroMemory( audioPtr, dataLen );
6482 // Unlock the DS buffer
6483 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
6484 if ( FAILED( result ) ) {
6485 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking input buffer!";
6486 errorText_ = errorStream_.str();
6490 // If we start recording again, we must begin at beginning of buffer.
6491 handle->bufferPointer[1] = 0;
6495 timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows.
6496 MUTEX_UNLOCK( &stream_.mutex );
6498 if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );
6501 void RtApiDs :: abortStream()
6504 if ( stream_.state == STREAM_STOPPED ) {
6505 errorText_ = "RtApiDs::abortStream(): the stream is already stopped!";
6506 error( RtAudioError::WARNING );
6510 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6511 handle->drainCounter = 2;
6516 void RtApiDs :: callbackEvent()
6518 if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) {
6519 Sleep( 50 ); // sleep 50 milliseconds
6523 if ( stream_.state == STREAM_CLOSED ) {
6524 errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!";
6525 error( RtAudioError::WARNING );
6529 CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
6530 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6532 // Check if we were draining the stream and signal is finished.
6533 if ( handle->drainCounter > stream_.nBuffers + 2 ) {
6535 stream_.state = STREAM_STOPPING;
6536 if ( handle->internalDrain == false )
6537 SetEvent( handle->condition );
6543 // Invoke user callback to get fresh output data UNLESS we are
6545 if ( handle->drainCounter == 0 ) {
6546 RtAudioCallback callback = (RtAudioCallback) info->callback;
6547 double streamTime = getStreamTime();
6548 RtAudioStreamStatus status = 0;
6549 if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
6550 status |= RTAUDIO_OUTPUT_UNDERFLOW;
6551 handle->xrun[0] = false;
6553 if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
6554 status |= RTAUDIO_INPUT_OVERFLOW;
6555 handle->xrun[1] = false;
6557 int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
6558 stream_.bufferSize, streamTime, status, info->userData );
6559 if ( cbReturnValue == 2 ) {
6560 stream_.state = STREAM_STOPPING;
6561 handle->drainCounter = 2;
6565 else if ( cbReturnValue == 1 ) {
6566 handle->drainCounter = 1;
6567 handle->internalDrain = true;
6572 DWORD currentWritePointer, safeWritePointer;
6573 DWORD currentReadPointer, safeReadPointer;
6574 UINT nextWritePointer;
6576 LPVOID buffer1 = NULL;
6577 LPVOID buffer2 = NULL;
6578 DWORD bufferSize1 = 0;
6579 DWORD bufferSize2 = 0;
6584 MUTEX_LOCK( &stream_.mutex );
6585 if ( stream_.state == STREAM_STOPPED ) {
6586 MUTEX_UNLOCK( &stream_.mutex );
6590 if ( buffersRolling == false ) {
6591 if ( stream_.mode == DUPLEX ) {
6592 //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
6594 // It takes a while for the devices to get rolling. As a result,
6595 // there's no guarantee that the capture and write device pointers
6596 // will move in lockstep. Wait here for both devices to start
6597 // rolling, and then set our buffer pointers accordingly.
6598 // e.g. Crystal Drivers: the capture buffer starts up 5700 to 9600
6599 // bytes later than the write buffer.
6601 // Stub: a serious risk of having a pre-emptive scheduling round
6602 // take place between the two GetCurrentPosition calls... but I'm
6603 // really not sure how to solve the problem. Temporarily boost to
6604 // Realtime priority, maybe; but I'm not sure what priority the
6605 // DirectSound service threads run at. We *should* be roughly
6606 // within a ms or so of correct.
6608 LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6609 LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6611 DWORD startSafeWritePointer, startSafeReadPointer;
6613 result = dsWriteBuffer->GetCurrentPosition( NULL, &startSafeWritePointer );
6614 if ( FAILED( result ) ) {
6615 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
6616 errorText_ = errorStream_.str();
6617 MUTEX_UNLOCK( &stream_.mutex );
6618 error( RtAudioError::SYSTEM_ERROR );
6621 result = dsCaptureBuffer->GetCurrentPosition( NULL, &startSafeReadPointer );
6622 if ( FAILED( result ) ) {
6623 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
6624 errorText_ = errorStream_.str();
6625 MUTEX_UNLOCK( &stream_.mutex );
6626 error( RtAudioError::SYSTEM_ERROR );
6630 result = dsWriteBuffer->GetCurrentPosition( NULL, &safeWritePointer );
6631 if ( FAILED( result ) ) {
6632 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
6633 errorText_ = errorStream_.str();
6634 MUTEX_UNLOCK( &stream_.mutex );
6635 error( RtAudioError::SYSTEM_ERROR );
6638 result = dsCaptureBuffer->GetCurrentPosition( NULL, &safeReadPointer );
6639 if ( FAILED( result ) ) {
6640 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
6641 errorText_ = errorStream_.str();
6642 MUTEX_UNLOCK( &stream_.mutex );
6643 error( RtAudioError::SYSTEM_ERROR );
6646 if ( safeWritePointer != startSafeWritePointer && safeReadPointer != startSafeReadPointer ) break;
6650 //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
6652 handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];
6653 if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];
6654 handle->bufferPointer[1] = safeReadPointer;
6656 else if ( stream_.mode == OUTPUT ) {
6658 // Set the proper nextWritePosition after initial startup.
6659 LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6660 result = dsWriteBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer );
6661 if ( FAILED( result ) ) {
6662 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
6663 errorText_ = errorStream_.str();
6664 MUTEX_UNLOCK( &stream_.mutex );
6665 error( RtAudioError::SYSTEM_ERROR );
6668 handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];
6669 if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];
6672 buffersRolling = true;
6675 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
6677 LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6679 if ( handle->drainCounter > 1 ) { // write zeros to the output stream
6680 bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
6681 bufferBytes *= formatBytes( stream_.userFormat );
6682 memset( stream_.userBuffer[0], 0, bufferBytes );
6685 // Setup parameters and do buffer conversion if necessary.
6686 if ( stream_.doConvertBuffer[0] ) {
6687 buffer = stream_.deviceBuffer;
6688 convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
6689 bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[0];
6690 bufferBytes *= formatBytes( stream_.deviceFormat[0] );
6693 buffer = stream_.userBuffer[0];
6694 bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
6695 bufferBytes *= formatBytes( stream_.userFormat );
6698 // No byte swapping necessary in DirectSound implementation.
6700 // Ahhh ... windoze. 16-bit data is signed but 8-bit data is
6701 // unsigned. So, we need to convert our signed 8-bit data here to
6703 if ( stream_.deviceFormat[0] == RTAUDIO_SINT8 )
6704 for ( int i=0; i<bufferBytes; i++ ) buffer[i] = (unsigned char) ( buffer[i] + 128 );
6706 DWORD dsBufferSize = handle->dsBufferSize[0];
6707 nextWritePointer = handle->bufferPointer[0];
6709 DWORD endWrite, leadPointer;
6711 // Find out where the read and "safe write" pointers are.
6712 result = dsBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer );
6713 if ( FAILED( result ) ) {
6714 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
6715 errorText_ = errorStream_.str();
6716 MUTEX_UNLOCK( &stream_.mutex );
6717 error( RtAudioError::SYSTEM_ERROR );
6721 // We will copy our output buffer into the region between
6722 // safeWritePointer and leadPointer. If leadPointer is not
6723 // beyond the next endWrite position, wait until it is.
6724 leadPointer = safeWritePointer + handle->dsPointerLeadTime[0];
6725 //std::cout << "safeWritePointer = " << safeWritePointer << ", leadPointer = " << leadPointer << ", nextWritePointer = " << nextWritePointer << std::endl;
6726 if ( leadPointer > dsBufferSize ) leadPointer -= dsBufferSize;
6727 if ( leadPointer < nextWritePointer ) leadPointer += dsBufferSize; // unwrap offset
6728 endWrite = nextWritePointer + bufferBytes;
6730 // Check whether the entire write region is behind the play pointer.
6731 if ( leadPointer >= endWrite ) break;
6733 // If we are here, then we must wait until the leadPointer advances
6734 // beyond the end of our next write region. We use the
6735 // Sleep() function to suspend operation until that happens.
6736 double millis = ( endWrite - leadPointer ) * 1000.0;
6737 millis /= ( formatBytes( stream_.deviceFormat[0]) * stream_.nDeviceChannels[0] * stream_.sampleRate);
6738 if ( millis < 1.0 ) millis = 1.0;
6739 Sleep( (DWORD) millis );
6742 if ( dsPointerBetween( nextWritePointer, safeWritePointer, currentWritePointer, dsBufferSize )
6743 || dsPointerBetween( endWrite, safeWritePointer, currentWritePointer, dsBufferSize ) ) {
6744 // We've strayed into the forbidden zone ... resync the read pointer.
6745 handle->xrun[0] = true;
6746 nextWritePointer = safeWritePointer + handle->dsPointerLeadTime[0] - bufferBytes;
6747 if ( nextWritePointer >= dsBufferSize ) nextWritePointer -= dsBufferSize;
6748 handle->bufferPointer[0] = nextWritePointer;
6749 endWrite = nextWritePointer + bufferBytes;
6752 // Lock free space in the buffer
6753 result = dsBuffer->Lock( nextWritePointer, bufferBytes, &buffer1,
6754 &bufferSize1, &buffer2, &bufferSize2, 0 );
6755 if ( FAILED( result ) ) {
6756 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking buffer during playback!";
6757 errorText_ = errorStream_.str();
6758 MUTEX_UNLOCK( &stream_.mutex );
6759 error( RtAudioError::SYSTEM_ERROR );
6763 // Copy our buffer into the DS buffer
6764 CopyMemory( buffer1, buffer, bufferSize1 );
6765 if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 );
6767 // Update our buffer offset and unlock sound buffer
6768 dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
6769 if ( FAILED( result ) ) {
6770 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!";
6771 errorText_ = errorStream_.str();
6772 MUTEX_UNLOCK( &stream_.mutex );
6773 error( RtAudioError::SYSTEM_ERROR );
6776 nextWritePointer = ( nextWritePointer + bufferSize1 + bufferSize2 ) % dsBufferSize;
6777 handle->bufferPointer[0] = nextWritePointer;
6780 // Don't bother draining input
6781 if ( handle->drainCounter ) {
6782 handle->drainCounter++;
6786 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
6788 // Setup parameters.
6789 if ( stream_.doConvertBuffer[1] ) {
6790 buffer = stream_.deviceBuffer;
6791 bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[1];
6792 bufferBytes *= formatBytes( stream_.deviceFormat[1] );
6795 buffer = stream_.userBuffer[1];
6796 bufferBytes = stream_.bufferSize * stream_.nUserChannels[1];
6797 bufferBytes *= formatBytes( stream_.userFormat );
6800 LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6801 long nextReadPointer = handle->bufferPointer[1];
6802 DWORD dsBufferSize = handle->dsBufferSize[1];
6804 // Find out where the write and "safe read" pointers are.
6805 result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer );
6806 if ( FAILED( result ) ) {
6807 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
6808 errorText_ = errorStream_.str();
6809 MUTEX_UNLOCK( &stream_.mutex );
6810 error( RtAudioError::SYSTEM_ERROR );
6814 if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset
6815 DWORD endRead = nextReadPointer + bufferBytes;
6817 // Handling depends on whether we are INPUT or DUPLEX.
6818 // If we're in INPUT mode then waiting is a good thing. If we're in DUPLEX mode,
6819 // then a wait here will drag the write pointers into the forbidden zone.
6821 // In DUPLEX mode, rather than wait, we will back off the read pointer until
6822 // it's in a safe position. This causes dropouts, but it seems to be the only
6823 // practical way to sync up the read and write pointers reliably, given the
6824 // the very complex relationship between phase and increment of the read and write
6827 // In order to minimize audible dropouts in DUPLEX mode, we will
6828 // provide a pre-roll period of 0.5 seconds in which we return
6829 // zeros from the read buffer while the pointers sync up.
6831 if ( stream_.mode == DUPLEX ) {
6832 if ( safeReadPointer < endRead ) {
6833 if ( duplexPrerollBytes <= 0 ) {
6834 // Pre-roll time over. Be more agressive.
6835 int adjustment = endRead-safeReadPointer;
6837 handle->xrun[1] = true;
6839 // - large adjustments: we've probably run out of CPU cycles, so just resync exactly,
6840 // and perform fine adjustments later.
6841 // - small adjustments: back off by twice as much.
6842 if ( adjustment >= 2*bufferBytes )
6843 nextReadPointer = safeReadPointer-2*bufferBytes;
6845 nextReadPointer = safeReadPointer-bufferBytes-adjustment;
6847 if ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;
6851 // In pre=roll time. Just do it.
6852 nextReadPointer = safeReadPointer - bufferBytes;
6853 while ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;
6855 endRead = nextReadPointer + bufferBytes;
6858 else { // mode == INPUT
6859 while ( safeReadPointer < endRead && stream_.callbackInfo.isRunning ) {
6860 // See comments for playback.
6861 double millis = (endRead - safeReadPointer) * 1000.0;
6862 millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate);
6863 if ( millis < 1.0 ) millis = 1.0;
6864 Sleep( (DWORD) millis );
6866 // Wake up and find out where we are now.
6867 result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer );
6868 if ( FAILED( result ) ) {
6869 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
6870 errorText_ = errorStream_.str();
6871 MUTEX_UNLOCK( &stream_.mutex );
6872 error( RtAudioError::SYSTEM_ERROR );
6876 if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset
6880 // Lock free space in the buffer
6881 result = dsBuffer->Lock( nextReadPointer, bufferBytes, &buffer1,
6882 &bufferSize1, &buffer2, &bufferSize2, 0 );
6883 if ( FAILED( result ) ) {
6884 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking capture buffer!";
6885 errorText_ = errorStream_.str();
6886 MUTEX_UNLOCK( &stream_.mutex );
6887 error( RtAudioError::SYSTEM_ERROR );
6891 if ( duplexPrerollBytes <= 0 ) {
6892 // Copy our buffer into the DS buffer
6893 CopyMemory( buffer, buffer1, bufferSize1 );
6894 if ( buffer2 != NULL ) CopyMemory( buffer+bufferSize1, buffer2, bufferSize2 );
6897 memset( buffer, 0, bufferSize1 );
6898 if ( buffer2 != NULL ) memset( buffer + bufferSize1, 0, bufferSize2 );
6899 duplexPrerollBytes -= bufferSize1 + bufferSize2;
6902 // Update our buffer offset and unlock sound buffer
6903 nextReadPointer = ( nextReadPointer + bufferSize1 + bufferSize2 ) % dsBufferSize;
6904 dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
6905 if ( FAILED( result ) ) {
6906 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking capture buffer!";
6907 errorText_ = errorStream_.str();
6908 MUTEX_UNLOCK( &stream_.mutex );
6909 error( RtAudioError::SYSTEM_ERROR );
6912 handle->bufferPointer[1] = nextReadPointer;
6914 // No byte swapping necessary in DirectSound implementation.
6916 // If necessary, convert 8-bit data from unsigned to signed.
6917 if ( stream_.deviceFormat[1] == RTAUDIO_SINT8 )
6918 for ( int j=0; j<bufferBytes; j++ ) buffer[j] = (signed char) ( buffer[j] - 128 );
6920 // Do buffer conversion if necessary.
6921 if ( stream_.doConvertBuffer[1] )
6922 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
6926 MUTEX_UNLOCK( &stream_.mutex );
6927 RtApi::tickStreamTime();
6930 // Definitions for utility functions and callbacks
6931 // specific to the DirectSound implementation.
6933 static unsigned __stdcall callbackHandler( void *ptr )
6935 CallbackInfo *info = (CallbackInfo *) ptr;
6936 RtApiDs *object = (RtApiDs *) info->object;
6937 bool* isRunning = &info->isRunning;
6939 while ( *isRunning == true ) {
6940 object->callbackEvent();
6947 static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
6948 LPCTSTR description,
6952 struct DsProbeData& probeInfo = *(struct DsProbeData*) lpContext;
6953 std::vector<struct DsDevice>& dsDevices = *probeInfo.dsDevices;
6956 bool validDevice = false;
6957 if ( probeInfo.isInput == true ) {
6959 LPDIRECTSOUNDCAPTURE object;
6961 hr = DirectSoundCaptureCreate( lpguid, &object, NULL );
6962 if ( hr != DS_OK ) return TRUE;
6964 caps.dwSize = sizeof(caps);
6965 hr = object->GetCaps( &caps );
6966 if ( hr == DS_OK ) {
6967 if ( caps.dwChannels > 0 && caps.dwFormats > 0 )
6974 LPDIRECTSOUND object;
6975 hr = DirectSoundCreate( lpguid, &object, NULL );
6976 if ( hr != DS_OK ) return TRUE;
6978 caps.dwSize = sizeof(caps);
6979 hr = object->GetCaps( &caps );
6980 if ( hr == DS_OK ) {
6981 if ( caps.dwFlags & DSCAPS_PRIMARYMONO || caps.dwFlags & DSCAPS_PRIMARYSTEREO )
6987 // If good device, then save its name and guid.
6988 std::string name = convertCharPointerToStdString( description );
6989 //if ( name == "Primary Sound Driver" || name == "Primary Sound Capture Driver" )
6990 if ( lpguid == NULL )
6991 name = "Default Device";
6992 if ( validDevice ) {
6993 for ( unsigned int i=0; i<dsDevices.size(); i++ ) {
6994 if ( dsDevices[i].name == name ) {
6995 dsDevices[i].found = true;
6996 if ( probeInfo.isInput ) {
6997 dsDevices[i].id[1] = lpguid;
6998 dsDevices[i].validId[1] = true;
7001 dsDevices[i].id[0] = lpguid;
7002 dsDevices[i].validId[0] = true;
7010 device.found = true;
7011 if ( probeInfo.isInput ) {
7012 device.id[1] = lpguid;
7013 device.validId[1] = true;
7016 device.id[0] = lpguid;
7017 device.validId[0] = true;
7019 dsDevices.push_back( device );
7025 static const char* getErrorString( int code )
7029 case DSERR_ALLOCATED:
7030 return "Already allocated";
7032 case DSERR_CONTROLUNAVAIL:
7033 return "Control unavailable";
7035 case DSERR_INVALIDPARAM:
7036 return "Invalid parameter";
7038 case DSERR_INVALIDCALL:
7039 return "Invalid call";
7042 return "Generic error";
7044 case DSERR_PRIOLEVELNEEDED:
7045 return "Priority level needed";
7047 case DSERR_OUTOFMEMORY:
7048 return "Out of memory";
7050 case DSERR_BADFORMAT:
7051 return "The sample rate or the channel format is not supported";
7053 case DSERR_UNSUPPORTED:
7054 return "Not supported";
7056 case DSERR_NODRIVER:
7059 case DSERR_ALREADYINITIALIZED:
7060 return "Already initialized";
7062 case DSERR_NOAGGREGATION:
7063 return "No aggregation";
7065 case DSERR_BUFFERLOST:
7066 return "Buffer lost";
7068 case DSERR_OTHERAPPHASPRIO:
7069 return "Another application already has priority";
7071 case DSERR_UNINITIALIZED:
7072 return "Uninitialized";
7075 return "DirectSound unknown error";
7078 //******************** End of __WINDOWS_DS__ *********************//
7082 #if defined(__LINUX_ALSA__)
7084 #include <alsa/asoundlib.h>
7087 // A structure to hold various information related to the ALSA API
7090 snd_pcm_t *handles[2];
7093 pthread_cond_t runnable_cv;
7097 :synchronized(false), runnable(false) { xrun[0] = false; xrun[1] = false; }
7100 static void *alsaCallbackHandler( void * ptr );
7102 RtApiAlsa :: RtApiAlsa()
7104 // Nothing to do here.
7107 RtApiAlsa :: ~RtApiAlsa()
7109 if ( stream_.state != STREAM_CLOSED ) closeStream();
7112 unsigned int RtApiAlsa :: getDeviceCount( void )
7114 unsigned nDevices = 0;
7115 int result, subdevice, card;
7119 // Count cards and devices
7121 snd_card_next( &card );
7122 while ( card >= 0 ) {
7123 sprintf( name, "hw:%d", card );
7124 result = snd_ctl_open( &handle, name, 0 );
7126 errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << ".";
7127 errorText_ = errorStream_.str();
7128 error( RtAudioError::WARNING );
7133 result = snd_ctl_pcm_next_device( handle, &subdevice );
7135 errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
7136 errorText_ = errorStream_.str();
7137 error( RtAudioError::WARNING );
7140 if ( subdevice < 0 )
7145 snd_ctl_close( handle );
7146 snd_card_next( &card );
7149 result = snd_ctl_open( &handle, "default", 0 );
7152 snd_ctl_close( handle );
7158 RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device )
7160 RtAudio::DeviceInfo info;
7161 info.probed = false;
7163 unsigned nDevices = 0;
7164 int result, subdevice, card;
7168 // Count cards and devices
7171 snd_card_next( &card );
7172 while ( card >= 0 ) {
7173 sprintf( name, "hw:%d", card );
7174 result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
7176 errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << ".";
7177 errorText_ = errorStream_.str();
7178 error( RtAudioError::WARNING );
7183 result = snd_ctl_pcm_next_device( chandle, &subdevice );
7185 errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
7186 errorText_ = errorStream_.str();
7187 error( RtAudioError::WARNING );
7190 if ( subdevice < 0 ) break;
7191 if ( nDevices == device ) {
7192 sprintf( name, "hw:%d,%d", card, subdevice );
7198 snd_ctl_close( chandle );
7199 snd_card_next( &card );
7202 result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
7203 if ( result == 0 ) {
7204 if ( nDevices == device ) {
7205 strcpy( name, "default" );
7211 if ( nDevices == 0 ) {
7212 errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!";
7213 error( RtAudioError::INVALID_USE );
7217 if ( device >= nDevices ) {
7218 errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!";
7219 error( RtAudioError::INVALID_USE );
7225 // If a stream is already open, we cannot probe the stream devices.
7226 // Thus, use the saved results.
7227 if ( stream_.state != STREAM_CLOSED &&
7228 ( stream_.device[0] == device || stream_.device[1] == device ) ) {
7229 snd_ctl_close( chandle );
7230 if ( device >= devices_.size() ) {
7231 errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened.";
7232 error( RtAudioError::WARNING );
7235 return devices_[ device ];
7238 int openMode = SND_PCM_ASYNC;
7239 snd_pcm_stream_t stream;
7240 snd_pcm_info_t *pcminfo;
7241 snd_pcm_info_alloca( &pcminfo );
7243 snd_pcm_hw_params_t *params;
7244 snd_pcm_hw_params_alloca( ¶ms );
7246 // First try for playback unless default device (which has subdev -1)
7247 stream = SND_PCM_STREAM_PLAYBACK;
7248 snd_pcm_info_set_stream( pcminfo, stream );
7249 if ( subdevice != -1 ) {
7250 snd_pcm_info_set_device( pcminfo, subdevice );
7251 snd_pcm_info_set_subdevice( pcminfo, 0 );
7253 result = snd_ctl_pcm_info( chandle, pcminfo );
7255 // Device probably doesn't support playback.
7260 result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK );
7262 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
7263 errorText_ = errorStream_.str();
7264 error( RtAudioError::WARNING );
7268 // The device is open ... fill the parameter structure.
7269 result = snd_pcm_hw_params_any( phandle, params );
7271 snd_pcm_close( phandle );
7272 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
7273 errorText_ = errorStream_.str();
7274 error( RtAudioError::WARNING );
7278 // Get output channel information.
7280 result = snd_pcm_hw_params_get_channels_max( params, &value );
7282 snd_pcm_close( phandle );
7283 errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << ".";
7284 errorText_ = errorStream_.str();
7285 error( RtAudioError::WARNING );
7288 info.outputChannels = value;
7289 snd_pcm_close( phandle );
7292 stream = SND_PCM_STREAM_CAPTURE;
7293 snd_pcm_info_set_stream( pcminfo, stream );
7295 // Now try for capture unless default device (with subdev = -1)
7296 if ( subdevice != -1 ) {
7297 result = snd_ctl_pcm_info( chandle, pcminfo );
7298 snd_ctl_close( chandle );
7300 // Device probably doesn't support capture.
7301 if ( info.outputChannels == 0 ) return info;
7302 goto probeParameters;
7306 snd_ctl_close( chandle );
7308 result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
7310 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
7311 errorText_ = errorStream_.str();
7312 error( RtAudioError::WARNING );
7313 if ( info.outputChannels == 0 ) return info;
7314 goto probeParameters;
7317 // The device is open ... fill the parameter structure.
7318 result = snd_pcm_hw_params_any( phandle, params );
7320 snd_pcm_close( phandle );
7321 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
7322 errorText_ = errorStream_.str();
7323 error( RtAudioError::WARNING );
7324 if ( info.outputChannels == 0 ) return info;
7325 goto probeParameters;
7328 result = snd_pcm_hw_params_get_channels_max( params, &value );
7330 snd_pcm_close( phandle );
7331 errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << ".";
7332 errorText_ = errorStream_.str();
7333 error( RtAudioError::WARNING );
7334 if ( info.outputChannels == 0 ) return info;
7335 goto probeParameters;
7337 info.inputChannels = value;
7338 snd_pcm_close( phandle );
7340 // If device opens for both playback and capture, we determine the channels.
7341 if ( info.outputChannels > 0 && info.inputChannels > 0 )
7342 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
7344 // ALSA doesn't provide default devices so we'll use the first available one.
7345 if ( device == 0 && info.outputChannels > 0 )
7346 info.isDefaultOutput = true;
7347 if ( device == 0 && info.inputChannels > 0 )
7348 info.isDefaultInput = true;
7351 // At this point, we just need to figure out the supported data
7352 // formats and sample rates. We'll proceed by opening the device in
7353 // the direction with the maximum number of channels, or playback if
7354 // they are equal. This might limit our sample rate options, but so
7357 if ( info.outputChannels >= info.inputChannels )
7358 stream = SND_PCM_STREAM_PLAYBACK;
7360 stream = SND_PCM_STREAM_CAPTURE;
7361 snd_pcm_info_set_stream( pcminfo, stream );
7363 result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
7365 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
7366 errorText_ = errorStream_.str();
7367 error( RtAudioError::WARNING );
7371 // The device is open ... fill the parameter structure.
7372 result = snd_pcm_hw_params_any( phandle, params );
7374 snd_pcm_close( phandle );
7375 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
7376 errorText_ = errorStream_.str();
7377 error( RtAudioError::WARNING );
7381 // Test our discrete set of sample rate values.
7382 info.sampleRates.clear();
7383 for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {
7384 if ( snd_pcm_hw_params_test_rate( phandle, params, SAMPLE_RATES[i], 0 ) == 0 ) {
7385 info.sampleRates.push_back( SAMPLE_RATES[i] );
7387 if ( !info.preferredSampleRate || ( SAMPLE_RATES[i] <= 48000 && SAMPLE_RATES[i] > info.preferredSampleRate ) )
7388 info.preferredSampleRate = SAMPLE_RATES[i];
7391 if ( info.sampleRates.size() == 0 ) {
7392 snd_pcm_close( phandle );
7393 errorStream_ << "RtApiAlsa::getDeviceInfo: no supported sample rates found for device (" << name << ").";
7394 errorText_ = errorStream_.str();
7395 error( RtAudioError::WARNING );
7399 // Probe the supported data formats ... we don't care about endian-ness just yet
7400 snd_pcm_format_t format;
7401 info.nativeFormats = 0;
7402 format = SND_PCM_FORMAT_S8;
7403 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7404 info.nativeFormats |= RTAUDIO_SINT8;
7405 format = SND_PCM_FORMAT_S16;
7406 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7407 info.nativeFormats |= RTAUDIO_SINT16;
7408 format = SND_PCM_FORMAT_S24;
7409 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7410 info.nativeFormats |= RTAUDIO_SINT24;
7411 format = SND_PCM_FORMAT_S32;
7412 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7413 info.nativeFormats |= RTAUDIO_SINT32;
7414 format = SND_PCM_FORMAT_FLOAT;
7415 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7416 info.nativeFormats |= RTAUDIO_FLOAT32;
7417 format = SND_PCM_FORMAT_FLOAT64;
7418 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7419 info.nativeFormats |= RTAUDIO_FLOAT64;
7421 // Check that we have at least one supported format
7422 if ( info.nativeFormats == 0 ) {
7423 snd_pcm_close( phandle );
7424 errorStream_ << "RtApiAlsa::getDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio.";
7425 errorText_ = errorStream_.str();
7426 error( RtAudioError::WARNING );
7430 // Get the device name
7432 result = snd_card_get_name( card, &cardname );
7433 if ( result >= 0 ) {
7434 sprintf( name, "hw:%s,%d", cardname, subdevice );
7439 // That's all ... close the device and return
7440 snd_pcm_close( phandle );
7445 void RtApiAlsa :: saveDeviceInfo( void )
7449 unsigned int nDevices = getDeviceCount();
7450 devices_.resize( nDevices );
7451 for ( unsigned int i=0; i<nDevices; i++ )
7452 devices_[i] = getDeviceInfo( i );
7455 bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
7456 unsigned int firstChannel, unsigned int sampleRate,
7457 RtAudioFormat format, unsigned int *bufferSize,
7458 RtAudio::StreamOptions *options )
7461 #if defined(__RTAUDIO_DEBUG__)
7463 snd_output_stdio_attach(&out, stderr, 0);
7466 // I'm not using the "plug" interface ... too much inconsistent behavior.
7468 unsigned nDevices = 0;
7469 int result, subdevice, card;
7473 if ( options && options->flags & RTAUDIO_ALSA_USE_DEFAULT )
7474 snprintf(name, sizeof(name), "%s", "default");
7476 // Count cards and devices
7478 snd_card_next( &card );
7479 while ( card >= 0 ) {
7480 sprintf( name, "hw:%d", card );
7481 result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
7483 errorStream_ << "RtApiAlsa::probeDeviceOpen: control open, card = " << card << ", " << snd_strerror( result ) << ".";
7484 errorText_ = errorStream_.str();
7489 result = snd_ctl_pcm_next_device( chandle, &subdevice );
7490 if ( result < 0 ) break;
7491 if ( subdevice < 0 ) break;
7492 if ( nDevices == device ) {
7493 sprintf( name, "hw:%d,%d", card, subdevice );
7494 snd_ctl_close( chandle );
7499 snd_ctl_close( chandle );
7500 snd_card_next( &card );
7503 result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
7504 if ( result == 0 ) {
7505 if ( nDevices == device ) {
7506 strcpy( name, "default" );
7512 if ( nDevices == 0 ) {
7513 // This should not happen because a check is made before this function is called.
7514 errorText_ = "RtApiAlsa::probeDeviceOpen: no devices found!";
7518 if ( device >= nDevices ) {
7519 // This should not happen because a check is made before this function is called.
7520 errorText_ = "RtApiAlsa::probeDeviceOpen: device ID is invalid!";
7527 // The getDeviceInfo() function will not work for a device that is
7528 // already open. Thus, we'll probe the system before opening a
7529 // stream and save the results for use by getDeviceInfo().
7530 if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) // only do once
7531 this->saveDeviceInfo();
7533 snd_pcm_stream_t stream;
7534 if ( mode == OUTPUT )
7535 stream = SND_PCM_STREAM_PLAYBACK;
7537 stream = SND_PCM_STREAM_CAPTURE;
7540 int openMode = SND_PCM_ASYNC;
7541 result = snd_pcm_open( &phandle, name, stream, openMode );
7543 if ( mode == OUTPUT )
7544 errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for output.";
7546 errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for input.";
7547 errorText_ = errorStream_.str();
7551 // Fill the parameter structure.
7552 snd_pcm_hw_params_t *hw_params;
7553 snd_pcm_hw_params_alloca( &hw_params );
7554 result = snd_pcm_hw_params_any( phandle, hw_params );
7556 snd_pcm_close( phandle );
7557 errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") parameters, " << snd_strerror( result ) << ".";
7558 errorText_ = errorStream_.str();
7562 #if defined(__RTAUDIO_DEBUG__)
7563 fprintf( stderr, "\nRtApiAlsa: dump hardware params just after device open:\n\n" );
7564 snd_pcm_hw_params_dump( hw_params, out );
7567 // Set access ... check user preference.
7568 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) {
7569 stream_.userInterleaved = false;
7570 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
7572 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
7573 stream_.deviceInterleaved[mode] = true;
7576 stream_.deviceInterleaved[mode] = false;
7579 stream_.userInterleaved = true;
7580 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
7582 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
7583 stream_.deviceInterleaved[mode] = false;
7586 stream_.deviceInterleaved[mode] = true;
7590 snd_pcm_close( phandle );
7591 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") access, " << snd_strerror( result ) << ".";
7592 errorText_ = errorStream_.str();
7596 // Determine how to set the device format.
7597 stream_.userFormat = format;
7598 snd_pcm_format_t deviceFormat = SND_PCM_FORMAT_UNKNOWN;
7600 if ( format == RTAUDIO_SINT8 )
7601 deviceFormat = SND_PCM_FORMAT_S8;
7602 else if ( format == RTAUDIO_SINT16 )
7603 deviceFormat = SND_PCM_FORMAT_S16;
7604 else if ( format == RTAUDIO_SINT24 )
7605 deviceFormat = SND_PCM_FORMAT_S24;
7606 else if ( format == RTAUDIO_SINT32 )
7607 deviceFormat = SND_PCM_FORMAT_S32;
7608 else if ( format == RTAUDIO_FLOAT32 )
7609 deviceFormat = SND_PCM_FORMAT_FLOAT;
7610 else if ( format == RTAUDIO_FLOAT64 )
7611 deviceFormat = SND_PCM_FORMAT_FLOAT64;
7613 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat) == 0) {
7614 stream_.deviceFormat[mode] = format;
7618 // The user requested format is not natively supported by the device.
7619 deviceFormat = SND_PCM_FORMAT_FLOAT64;
7620 if ( snd_pcm_hw_params_test_format( phandle, hw_params, deviceFormat ) == 0 ) {
7621 stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
7625 deviceFormat = SND_PCM_FORMAT_FLOAT;
7626 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7627 stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
7631 deviceFormat = SND_PCM_FORMAT_S32;
7632 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7633 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
7637 deviceFormat = SND_PCM_FORMAT_S24;
7638 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7639 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
7643 deviceFormat = SND_PCM_FORMAT_S16;
7644 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7645 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
7649 deviceFormat = SND_PCM_FORMAT_S8;
7650 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7651 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
7655 // If we get here, no supported format was found.
7656 snd_pcm_close( phandle );
7657 errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device " << device << " data format not supported by RtAudio.";
7658 errorText_ = errorStream_.str();
7662 result = snd_pcm_hw_params_set_format( phandle, hw_params, deviceFormat );
7664 snd_pcm_close( phandle );
7665 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") data format, " << snd_strerror( result ) << ".";
7666 errorText_ = errorStream_.str();
7670 // Determine whether byte-swaping is necessary.
7671 stream_.doByteSwap[mode] = false;
7672 if ( deviceFormat != SND_PCM_FORMAT_S8 ) {
7673 result = snd_pcm_format_cpu_endian( deviceFormat );
7675 stream_.doByteSwap[mode] = true;
7676 else if (result < 0) {
7677 snd_pcm_close( phandle );
7678 errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") endian-ness, " << snd_strerror( result ) << ".";
7679 errorText_ = errorStream_.str();
7684 // Set the sample rate.
7685 result = snd_pcm_hw_params_set_rate_near( phandle, hw_params, (unsigned int*) &sampleRate, 0 );
7687 snd_pcm_close( phandle );
7688 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting sample rate on device (" << name << "), " << snd_strerror( result ) << ".";
7689 errorText_ = errorStream_.str();
7693 // Determine the number of channels for this device. We support a possible
7694 // minimum device channel number > than the value requested by the user.
7695 stream_.nUserChannels[mode] = channels;
7697 result = snd_pcm_hw_params_get_channels_max( hw_params, &value );
7698 unsigned int deviceChannels = value;
7699 if ( result < 0 || deviceChannels < channels + firstChannel ) {
7700 snd_pcm_close( phandle );
7701 errorStream_ << "RtApiAlsa::probeDeviceOpen: requested channel parameters not supported by device (" << name << "), " << snd_strerror( result ) << ".";
7702 errorText_ = errorStream_.str();
7706 result = snd_pcm_hw_params_get_channels_min( hw_params, &value );
7708 snd_pcm_close( phandle );
7709 errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting minimum channels for device (" << name << "), " << snd_strerror( result ) << ".";
7710 errorText_ = errorStream_.str();
7713 deviceChannels = value;
7714 if ( deviceChannels < channels + firstChannel ) deviceChannels = channels + firstChannel;
7715 stream_.nDeviceChannels[mode] = deviceChannels;
7717 // Set the device channels.
7718 result = snd_pcm_hw_params_set_channels( phandle, hw_params, deviceChannels );
7720 snd_pcm_close( phandle );
7721 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting channels for device (" << name << "), " << snd_strerror( result ) << ".";
7722 errorText_ = errorStream_.str();
7726 // Set the buffer (or period) size.
7728 snd_pcm_uframes_t periodSize = *bufferSize;
7729 result = snd_pcm_hw_params_set_period_size_near( phandle, hw_params, &periodSize, &dir );
7731 snd_pcm_close( phandle );
7732 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting period size for device (" << name << "), " << snd_strerror( result ) << ".";
7733 errorText_ = errorStream_.str();
7736 *bufferSize = periodSize;
7738 // Set the buffer number, which in ALSA is referred to as the "period".
7739 unsigned int periods = 0;
7740 if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2;
7741 if ( options && options->numberOfBuffers > 0 ) periods = options->numberOfBuffers;
7742 if ( periods < 2 ) periods = 4; // a fairly safe default value
7743 result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir );
7745 snd_pcm_close( phandle );
7746 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << ".";
7747 errorText_ = errorStream_.str();
7751 // If attempting to setup a duplex stream, the bufferSize parameter
7752 // MUST be the same in both directions!
7753 if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {
7754 snd_pcm_close( phandle );
7755 errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ").";
7756 errorText_ = errorStream_.str();
7760 stream_.bufferSize = *bufferSize;
7762 // Install the hardware configuration
7763 result = snd_pcm_hw_params( phandle, hw_params );
7765 snd_pcm_close( phandle );
7766 errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing hardware configuration on device (" << name << "), " << snd_strerror( result ) << ".";
7767 errorText_ = errorStream_.str();
7771 #if defined(__RTAUDIO_DEBUG__)
7772 fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n");
7773 snd_pcm_hw_params_dump( hw_params, out );
7776 // Set the software configuration to fill buffers with zeros and prevent device stopping on xruns.
7777 snd_pcm_sw_params_t *sw_params = NULL;
7778 snd_pcm_sw_params_alloca( &sw_params );
7779 snd_pcm_sw_params_current( phandle, sw_params );
7780 snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize );
7781 snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, ULONG_MAX );
7782 snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 );
7784 // The following two settings were suggested by Theo Veenker
7785 //snd_pcm_sw_params_set_avail_min( phandle, sw_params, *bufferSize );
7786 //snd_pcm_sw_params_set_xfer_align( phandle, sw_params, 1 );
7788 // here are two options for a fix
7789 //snd_pcm_sw_params_set_silence_size( phandle, sw_params, ULONG_MAX );
7790 snd_pcm_uframes_t val;
7791 snd_pcm_sw_params_get_boundary( sw_params, &val );
7792 snd_pcm_sw_params_set_silence_size( phandle, sw_params, val );
7794 result = snd_pcm_sw_params( phandle, sw_params );
7796 snd_pcm_close( phandle );
7797 errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing software configuration on device (" << name << "), " << snd_strerror( result ) << ".";
7798 errorText_ = errorStream_.str();
7802 #if defined(__RTAUDIO_DEBUG__)
7803 fprintf(stderr, "\nRtApiAlsa: dump software params after installation:\n\n");
7804 snd_pcm_sw_params_dump( sw_params, out );
7807 // Set flags for buffer conversion
7808 stream_.doConvertBuffer[mode] = false;
7809 if ( stream_.userFormat != stream_.deviceFormat[mode] )
7810 stream_.doConvertBuffer[mode] = true;
7811 if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
7812 stream_.doConvertBuffer[mode] = true;
7813 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
7814 stream_.nUserChannels[mode] > 1 )
7815 stream_.doConvertBuffer[mode] = true;
7817 // Allocate the ApiHandle if necessary and then save.
7818 AlsaHandle *apiInfo = 0;
7819 if ( stream_.apiHandle == 0 ) {
7821 apiInfo = (AlsaHandle *) new AlsaHandle;
7823 catch ( std::bad_alloc& ) {
7824 errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating AlsaHandle memory.";
7828 if ( pthread_cond_init( &apiInfo->runnable_cv, NULL ) ) {
7829 errorText_ = "RtApiAlsa::probeDeviceOpen: error initializing pthread condition variable.";
7833 stream_.apiHandle = (void *) apiInfo;
7834 apiInfo->handles[0] = 0;
7835 apiInfo->handles[1] = 0;
7838 apiInfo = (AlsaHandle *) stream_.apiHandle;
7840 apiInfo->handles[mode] = phandle;
7843 // Allocate necessary internal buffers.
7844 unsigned long bufferBytes;
7845 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
7846 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
7847 if ( stream_.userBuffer[mode] == NULL ) {
7848 errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating user buffer memory.";
7852 if ( stream_.doConvertBuffer[mode] ) {
7854 bool makeBuffer = true;
7855 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
7856 if ( mode == INPUT ) {
7857 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
7858 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
7859 if ( bufferBytes <= bytesOut ) makeBuffer = false;
7864 bufferBytes *= *bufferSize;
7865 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
7866 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
7867 if ( stream_.deviceBuffer == NULL ) {
7868 errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating device buffer memory.";
7874 stream_.sampleRate = sampleRate;
7875 stream_.nBuffers = periods;
7876 stream_.device[mode] = device;
7877 stream_.state = STREAM_STOPPED;
7879 // Setup the buffer conversion information structure.
7880 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
7882 // Setup thread if necessary.
7883 if ( stream_.mode == OUTPUT && mode == INPUT ) {
7884 // We had already set up an output stream.
7885 stream_.mode = DUPLEX;
7886 // Link the streams if possible.
7887 apiInfo->synchronized = false;
7888 if ( snd_pcm_link( apiInfo->handles[0], apiInfo->handles[1] ) == 0 )
7889 apiInfo->synchronized = true;
7891 errorText_ = "RtApiAlsa::probeDeviceOpen: unable to synchronize input and output devices.";
7892 error( RtAudioError::WARNING );
7896 stream_.mode = mode;
7898 // Setup callback thread.
7899 stream_.callbackInfo.object = (void *) this;
7901 // Set the thread attributes for joinable and realtime scheduling
7902 // priority (optional). The higher priority will only take affect
7903 // if the program is run as root or suid. Note, under Linux
7904 // processes with CAP_SYS_NICE privilege, a user can change
7905 // scheduling policy and priority (thus need not be root). See
7906 // POSIX "capabilities".
7907 pthread_attr_t attr;
7908 pthread_attr_init( &attr );
7909 pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
7910 #ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
7911 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
7912 stream_.callbackInfo.doRealtime = true;
7913 struct sched_param param;
7914 int priority = options->priority;
7915 int min = sched_get_priority_min( SCHED_RR );
7916 int max = sched_get_priority_max( SCHED_RR );
7917 if ( priority < min ) priority = min;
7918 else if ( priority > max ) priority = max;
7919 param.sched_priority = priority;
7921 // Set the policy BEFORE the priority. Otherwise it fails.
7922 pthread_attr_setschedpolicy(&attr, SCHED_RR);
7923 pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM);
7924 // This is definitely required. Otherwise it fails.
7925 pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED);
7926 pthread_attr_setschedparam(&attr, ¶m);
7929 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
7931 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
7934 stream_.callbackInfo.isRunning = true;
7935 result = pthread_create( &stream_.callbackInfo.thread, &attr, alsaCallbackHandler, &stream_.callbackInfo );
7936 pthread_attr_destroy( &attr );
7938 // Failed. Try instead with default attributes.
7939 result = pthread_create( &stream_.callbackInfo.thread, NULL, alsaCallbackHandler, &stream_.callbackInfo );
7941 stream_.callbackInfo.isRunning = false;
7942 errorText_ = "RtApiAlsa::error creating callback thread!";
7952 pthread_cond_destroy( &apiInfo->runnable_cv );
7953 if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
7954 if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
7956 stream_.apiHandle = 0;
7959 if ( phandle) snd_pcm_close( phandle );
7961 for ( int i=0; i<2; i++ ) {
7962 if ( stream_.userBuffer[i] ) {
7963 free( stream_.userBuffer[i] );
7964 stream_.userBuffer[i] = 0;
7968 if ( stream_.deviceBuffer ) {
7969 free( stream_.deviceBuffer );
7970 stream_.deviceBuffer = 0;
7973 stream_.state = STREAM_CLOSED;
7977 void RtApiAlsa :: closeStream()
7979 if ( stream_.state == STREAM_CLOSED ) {
7980 errorText_ = "RtApiAlsa::closeStream(): no open stream to close!";
7981 error( RtAudioError::WARNING );
7985 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
7986 stream_.callbackInfo.isRunning = false;
7987 MUTEX_LOCK( &stream_.mutex );
7988 if ( stream_.state == STREAM_STOPPED ) {
7989 apiInfo->runnable = true;
7990 pthread_cond_signal( &apiInfo->runnable_cv );
7992 MUTEX_UNLOCK( &stream_.mutex );
7993 pthread_join( stream_.callbackInfo.thread, NULL );
7995 if ( stream_.state == STREAM_RUNNING ) {
7996 stream_.state = STREAM_STOPPED;
7997 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
7998 snd_pcm_drop( apiInfo->handles[0] );
7999 if ( stream_.mode == INPUT || stream_.mode == DUPLEX )
8000 snd_pcm_drop( apiInfo->handles[1] );
8004 pthread_cond_destroy( &apiInfo->runnable_cv );
8005 if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
8006 if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
8008 stream_.apiHandle = 0;
8011 for ( int i=0; i<2; i++ ) {
8012 if ( stream_.userBuffer[i] ) {
8013 free( stream_.userBuffer[i] );
8014 stream_.userBuffer[i] = 0;
8018 if ( stream_.deviceBuffer ) {
8019 free( stream_.deviceBuffer );
8020 stream_.deviceBuffer = 0;
8023 stream_.mode = UNINITIALIZED;
8024 stream_.state = STREAM_CLOSED;
8027 void RtApiAlsa :: startStream()
8029 // This method calls snd_pcm_prepare if the device isn't already in that state.
8032 if ( stream_.state == STREAM_RUNNING ) {
8033 errorText_ = "RtApiAlsa::startStream(): the stream is already running!";
8034 error( RtAudioError::WARNING );
8038 MUTEX_LOCK( &stream_.mutex );
8041 snd_pcm_state_t state;
8042 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
8043 snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
8044 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
8045 state = snd_pcm_state( handle[0] );
8046 if ( state != SND_PCM_STATE_PREPARED ) {
8047 result = snd_pcm_prepare( handle[0] );
8049 errorStream_ << "RtApiAlsa::startStream: error preparing output pcm device, " << snd_strerror( result ) << ".";
8050 errorText_ = errorStream_.str();
8056 if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
8057 result = snd_pcm_drop(handle[1]); // fix to remove stale data received since device has been open
8058 state = snd_pcm_state( handle[1] );
8059 if ( state != SND_PCM_STATE_PREPARED ) {
8060 result = snd_pcm_prepare( handle[1] );
8062 errorStream_ << "RtApiAlsa::startStream: error preparing input pcm device, " << snd_strerror( result ) << ".";
8063 errorText_ = errorStream_.str();
8069 stream_.state = STREAM_RUNNING;
8072 apiInfo->runnable = true;
8073 pthread_cond_signal( &apiInfo->runnable_cv );
8074 MUTEX_UNLOCK( &stream_.mutex );
8076 if ( result >= 0 ) return;
8077 error( RtAudioError::SYSTEM_ERROR );
8080 void RtApiAlsa :: stopStream()
8083 if ( stream_.state == STREAM_STOPPED ) {
8084 errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!";
8085 error( RtAudioError::WARNING );
8089 stream_.state = STREAM_STOPPED;
8090 MUTEX_LOCK( &stream_.mutex );
8093 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
8094 snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
8095 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
8096 if ( apiInfo->synchronized )
8097 result = snd_pcm_drop( handle[0] );
8099 result = snd_pcm_drain( handle[0] );
8101 errorStream_ << "RtApiAlsa::stopStream: error draining output pcm device, " << snd_strerror( result ) << ".";
8102 errorText_ = errorStream_.str();
8107 if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
8108 result = snd_pcm_drop( handle[1] );
8110 errorStream_ << "RtApiAlsa::stopStream: error stopping input pcm device, " << snd_strerror( result ) << ".";
8111 errorText_ = errorStream_.str();
8117 apiInfo->runnable = false; // fixes high CPU usage when stopped
8118 MUTEX_UNLOCK( &stream_.mutex );
8120 if ( result >= 0 ) return;
8121 error( RtAudioError::SYSTEM_ERROR );
8124 void RtApiAlsa :: abortStream()
8127 if ( stream_.state == STREAM_STOPPED ) {
8128 errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!";
8129 error( RtAudioError::WARNING );
8133 stream_.state = STREAM_STOPPED;
8134 MUTEX_LOCK( &stream_.mutex );
8137 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
8138 snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
8139 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
8140 result = snd_pcm_drop( handle[0] );
8142 errorStream_ << "RtApiAlsa::abortStream: error aborting output pcm device, " << snd_strerror( result ) << ".";
8143 errorText_ = errorStream_.str();
8148 if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
8149 result = snd_pcm_drop( handle[1] );
8151 errorStream_ << "RtApiAlsa::abortStream: error aborting input pcm device, " << snd_strerror( result ) << ".";
8152 errorText_ = errorStream_.str();
8158 apiInfo->runnable = false; // fixes high CPU usage when stopped
8159 MUTEX_UNLOCK( &stream_.mutex );
8161 if ( result >= 0 ) return;
8162 error( RtAudioError::SYSTEM_ERROR );
8165 void RtApiAlsa :: callbackEvent()
8167 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
8168 if ( stream_.state == STREAM_STOPPED ) {
8169 MUTEX_LOCK( &stream_.mutex );
8170 while ( !apiInfo->runnable )
8171 pthread_cond_wait( &apiInfo->runnable_cv, &stream_.mutex );
8173 if ( stream_.state != STREAM_RUNNING ) {
8174 MUTEX_UNLOCK( &stream_.mutex );
8177 MUTEX_UNLOCK( &stream_.mutex );
8180 if ( stream_.state == STREAM_CLOSED ) {
8181 errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!";
8182 error( RtAudioError::WARNING );
8186 int doStopStream = 0;
8187 RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
8188 double streamTime = getStreamTime();
8189 RtAudioStreamStatus status = 0;
8190 if ( stream_.mode != INPUT && apiInfo->xrun[0] == true ) {
8191 status |= RTAUDIO_OUTPUT_UNDERFLOW;
8192 apiInfo->xrun[0] = false;
8194 if ( stream_.mode != OUTPUT && apiInfo->xrun[1] == true ) {
8195 status |= RTAUDIO_INPUT_OVERFLOW;
8196 apiInfo->xrun[1] = false;
8198 doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
8199 stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
8201 if ( doStopStream == 2 ) {
8206 MUTEX_LOCK( &stream_.mutex );
8208 // The state might change while waiting on a mutex.
8209 if ( stream_.state == STREAM_STOPPED ) goto unlock;
8215 snd_pcm_sframes_t frames;
8216 RtAudioFormat format;
8217 handle = (snd_pcm_t **) apiInfo->handles;
8219 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
8221 // Setup parameters.
8222 if ( stream_.doConvertBuffer[1] ) {
8223 buffer = stream_.deviceBuffer;
8224 channels = stream_.nDeviceChannels[1];
8225 format = stream_.deviceFormat[1];
8228 buffer = stream_.userBuffer[1];
8229 channels = stream_.nUserChannels[1];
8230 format = stream_.userFormat;
8233 // Read samples from device in interleaved/non-interleaved format.
8234 if ( stream_.deviceInterleaved[1] )
8235 result = snd_pcm_readi( handle[1], buffer, stream_.bufferSize );
8237 void *bufs[channels];
8238 size_t offset = stream_.bufferSize * formatBytes( format );
8239 for ( int i=0; i<channels; i++ )
8240 bufs[i] = (void *) (buffer + (i * offset));
8241 result = snd_pcm_readn( handle[1], bufs, stream_.bufferSize );
8244 if ( result < (int) stream_.bufferSize ) {
8245 // Either an error or overrun occured.
8246 if ( result == -EPIPE ) {
8247 snd_pcm_state_t state = snd_pcm_state( handle[1] );
8248 if ( state == SND_PCM_STATE_XRUN ) {
8249 apiInfo->xrun[1] = true;
8250 result = snd_pcm_prepare( handle[1] );
8252 errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after overrun, " << snd_strerror( result ) << ".";
8253 errorText_ = errorStream_.str();
8257 errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
8258 errorText_ = errorStream_.str();
8262 errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << ".";
8263 errorText_ = errorStream_.str();
8265 error( RtAudioError::WARNING );
8269 // Do byte swapping if necessary.
8270 if ( stream_.doByteSwap[1] )
8271 byteSwapBuffer( buffer, stream_.bufferSize * channels, format );
8273 // Do buffer conversion if necessary.
8274 if ( stream_.doConvertBuffer[1] )
8275 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
8277 // Check stream latency
8278 result = snd_pcm_delay( handle[1], &frames );
8279 if ( result == 0 && frames > 0 ) stream_.latency[1] = frames;
8284 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
8286 // Setup parameters and do buffer conversion if necessary.
8287 if ( stream_.doConvertBuffer[0] ) {
8288 buffer = stream_.deviceBuffer;
8289 convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
8290 channels = stream_.nDeviceChannels[0];
8291 format = stream_.deviceFormat[0];
8294 buffer = stream_.userBuffer[0];
8295 channels = stream_.nUserChannels[0];
8296 format = stream_.userFormat;
8299 // Do byte swapping if necessary.
8300 if ( stream_.doByteSwap[0] )
8301 byteSwapBuffer(buffer, stream_.bufferSize * channels, format);
8303 // Write samples to device in interleaved/non-interleaved format.
8304 if ( stream_.deviceInterleaved[0] )
8305 result = snd_pcm_writei( handle[0], buffer, stream_.bufferSize );
8307 void *bufs[channels];
8308 size_t offset = stream_.bufferSize * formatBytes( format );
8309 for ( int i=0; i<channels; i++ )
8310 bufs[i] = (void *) (buffer + (i * offset));
8311 result = snd_pcm_writen( handle[0], bufs, stream_.bufferSize );
8314 if ( result < (int) stream_.bufferSize ) {
8315 // Either an error or underrun occured.
8316 if ( result == -EPIPE ) {
8317 snd_pcm_state_t state = snd_pcm_state( handle[0] );
8318 if ( state == SND_PCM_STATE_XRUN ) {
8319 apiInfo->xrun[0] = true;
8320 result = snd_pcm_prepare( handle[0] );
8322 errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after underrun, " << snd_strerror( result ) << ".";
8323 errorText_ = errorStream_.str();
8326 errorText_ = "RtApiAlsa::callbackEvent: audio write error, underrun.";
8329 errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
8330 errorText_ = errorStream_.str();
8334 errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << ".";
8335 errorText_ = errorStream_.str();
8337 error( RtAudioError::WARNING );
8341 // Check stream latency
8342 result = snd_pcm_delay( handle[0], &frames );
8343 if ( result == 0 && frames > 0 ) stream_.latency[0] = frames;
8347 MUTEX_UNLOCK( &stream_.mutex );
8349 RtApi::tickStreamTime();
8350 if ( doStopStream == 1 ) this->stopStream();
8353 static void *alsaCallbackHandler( void *ptr )
8355 CallbackInfo *info = (CallbackInfo *) ptr;
8356 RtApiAlsa *object = (RtApiAlsa *) info->object;
8357 bool *isRunning = &info->isRunning;
8359 #ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
8360 if ( info->doRealtime ) {
8361 std::cerr << "RtAudio alsa: " <<
8362 (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") <<
8363 "running realtime scheduling" << std::endl;
8367 while ( *isRunning == true ) {
8368 pthread_testcancel();
8369 object->callbackEvent();
8372 pthread_exit( NULL );
8375 //******************** End of __LINUX_ALSA__ *********************//
8378 #if defined(__LINUX_PULSE__)
8380 // Code written by Peter Meerwald, pmeerw@pmeerw.net
8381 // and Tristan Matthews.
8383 #include <pulse/error.h>
8384 #include <pulse/simple.h>
8387 static const unsigned int SUPPORTED_SAMPLERATES[] = { 8000, 16000, 22050, 32000,
8388 44100, 48000, 96000, 0};
8390 struct rtaudio_pa_format_mapping_t {
8391 RtAudioFormat rtaudio_format;
8392 pa_sample_format_t pa_format;
8395 static const rtaudio_pa_format_mapping_t supported_sampleformats[] = {
8396 {RTAUDIO_SINT16, PA_SAMPLE_S16LE},
8397 {RTAUDIO_SINT32, PA_SAMPLE_S32LE},
8398 {RTAUDIO_FLOAT32, PA_SAMPLE_FLOAT32LE},
8399 {0, PA_SAMPLE_INVALID}};
8401 struct PulseAudioHandle {
8405 pthread_cond_t runnable_cv;
8407 PulseAudioHandle() : s_play(0), s_rec(0), runnable(false) { }
8410 RtApiPulse::~RtApiPulse()
8412 if ( stream_.state != STREAM_CLOSED )
8416 unsigned int RtApiPulse::getDeviceCount( void )
8421 RtAudio::DeviceInfo RtApiPulse::getDeviceInfo( unsigned int /*device*/ )
8423 RtAudio::DeviceInfo info;
8425 info.name = "PulseAudio";
8426 info.outputChannels = 2;
8427 info.inputChannels = 2;
8428 info.duplexChannels = 2;
8429 info.isDefaultOutput = true;
8430 info.isDefaultInput = true;
8432 for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr )
8433 info.sampleRates.push_back( *sr );
8435 info.preferredSampleRate = 48000;
8436 info.nativeFormats = RTAUDIO_SINT16 | RTAUDIO_SINT32 | RTAUDIO_FLOAT32;
8441 static void *pulseaudio_callback( void * user )
8443 CallbackInfo *cbi = static_cast<CallbackInfo *>( user );
8444 RtApiPulse *context = static_cast<RtApiPulse *>( cbi->object );
8445 volatile bool *isRunning = &cbi->isRunning;
8447 #ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
8448 if (cbi->doRealtime) {
8449 std::cerr << "RtAudio pulse: " <<
8450 (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") <<
8451 "running realtime scheduling" << std::endl;
8455 while ( *isRunning ) {
8456 pthread_testcancel();
8457 context->callbackEvent();
8460 pthread_exit( NULL );
8463 void RtApiPulse::closeStream( void )
8465 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8467 stream_.callbackInfo.isRunning = false;
8469 MUTEX_LOCK( &stream_.mutex );
8470 if ( stream_.state == STREAM_STOPPED ) {
8471 pah->runnable = true;
8472 pthread_cond_signal( &pah->runnable_cv );
8474 MUTEX_UNLOCK( &stream_.mutex );
8476 pthread_join( pah->thread, 0 );
8477 if ( pah->s_play ) {
8478 pa_simple_flush( pah->s_play, NULL );
8479 pa_simple_free( pah->s_play );
8482 pa_simple_free( pah->s_rec );
8484 pthread_cond_destroy( &pah->runnable_cv );
8486 stream_.apiHandle = 0;
8489 if ( stream_.userBuffer[0] ) {
8490 free( stream_.userBuffer[0] );
8491 stream_.userBuffer[0] = 0;
8493 if ( stream_.userBuffer[1] ) {
8494 free( stream_.userBuffer[1] );
8495 stream_.userBuffer[1] = 0;
8498 stream_.state = STREAM_CLOSED;
8499 stream_.mode = UNINITIALIZED;
8502 void RtApiPulse::callbackEvent( void )
8504 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8506 if ( stream_.state == STREAM_STOPPED ) {
8507 MUTEX_LOCK( &stream_.mutex );
8508 while ( !pah->runnable )
8509 pthread_cond_wait( &pah->runnable_cv, &stream_.mutex );
8511 if ( stream_.state != STREAM_RUNNING ) {
8512 MUTEX_UNLOCK( &stream_.mutex );
8515 MUTEX_UNLOCK( &stream_.mutex );
8518 if ( stream_.state == STREAM_CLOSED ) {
8519 errorText_ = "RtApiPulse::callbackEvent(): the stream is closed ... "
8520 "this shouldn't happen!";
8521 error( RtAudioError::WARNING );
8525 RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
8526 double streamTime = getStreamTime();
8527 RtAudioStreamStatus status = 0;
8528 int doStopStream = callback( stream_.userBuffer[OUTPUT], stream_.userBuffer[INPUT],
8529 stream_.bufferSize, streamTime, status,
8530 stream_.callbackInfo.userData );
8532 if ( doStopStream == 2 ) {
8537 MUTEX_LOCK( &stream_.mutex );
8538 void *pulse_in = stream_.doConvertBuffer[INPUT] ? stream_.deviceBuffer : stream_.userBuffer[INPUT];
8539 void *pulse_out = stream_.doConvertBuffer[OUTPUT] ? stream_.deviceBuffer : stream_.userBuffer[OUTPUT];
8541 if ( stream_.state != STREAM_RUNNING )
8546 if (stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
8547 if ( stream_.doConvertBuffer[OUTPUT] ) {
8548 convertBuffer( stream_.deviceBuffer,
8549 stream_.userBuffer[OUTPUT],
8550 stream_.convertInfo[OUTPUT] );
8551 bytes = stream_.nDeviceChannels[OUTPUT] * stream_.bufferSize *
8552 formatBytes( stream_.deviceFormat[OUTPUT] );
8554 bytes = stream_.nUserChannels[OUTPUT] * stream_.bufferSize *
8555 formatBytes( stream_.userFormat );
8557 if ( pa_simple_write( pah->s_play, pulse_out, bytes, &pa_error ) < 0 ) {
8558 errorStream_ << "RtApiPulse::callbackEvent: audio write error, " <<
8559 pa_strerror( pa_error ) << ".";
8560 errorText_ = errorStream_.str();
8561 error( RtAudioError::WARNING );
8565 if ( stream_.mode == INPUT || stream_.mode == DUPLEX) {
8566 if ( stream_.doConvertBuffer[INPUT] )
8567 bytes = stream_.nDeviceChannels[INPUT] * stream_.bufferSize *
8568 formatBytes( stream_.deviceFormat[INPUT] );
8570 bytes = stream_.nUserChannels[INPUT] * stream_.bufferSize *
8571 formatBytes( stream_.userFormat );
8573 if ( pa_simple_read( pah->s_rec, pulse_in, bytes, &pa_error ) < 0 ) {
8574 errorStream_ << "RtApiPulse::callbackEvent: audio read error, " <<
8575 pa_strerror( pa_error ) << ".";
8576 errorText_ = errorStream_.str();
8577 error( RtAudioError::WARNING );
8579 if ( stream_.doConvertBuffer[INPUT] ) {
8580 convertBuffer( stream_.userBuffer[INPUT],
8581 stream_.deviceBuffer,
8582 stream_.convertInfo[INPUT] );
8587 MUTEX_UNLOCK( &stream_.mutex );
8588 RtApi::tickStreamTime();
8590 if ( doStopStream == 1 )
8594 void RtApiPulse::startStream( void )
8596 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8598 if ( stream_.state == STREAM_CLOSED ) {
8599 errorText_ = "RtApiPulse::startStream(): the stream is not open!";
8600 error( RtAudioError::INVALID_USE );
8603 if ( stream_.state == STREAM_RUNNING ) {
8604 errorText_ = "RtApiPulse::startStream(): the stream is already running!";
8605 error( RtAudioError::WARNING );
8609 MUTEX_LOCK( &stream_.mutex );
8611 stream_.state = STREAM_RUNNING;
8613 pah->runnable = true;
8614 pthread_cond_signal( &pah->runnable_cv );
8615 MUTEX_UNLOCK( &stream_.mutex );
8618 void RtApiPulse::stopStream( void )
8620 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8622 if ( stream_.state == STREAM_CLOSED ) {
8623 errorText_ = "RtApiPulse::stopStream(): the stream is not open!";
8624 error( RtAudioError::INVALID_USE );
8627 if ( stream_.state == STREAM_STOPPED ) {
8628 errorText_ = "RtApiPulse::stopStream(): the stream is already stopped!";
8629 error( RtAudioError::WARNING );
8633 stream_.state = STREAM_STOPPED;
8634 MUTEX_LOCK( &stream_.mutex );
8636 if ( pah && pah->s_play ) {
8638 if ( pa_simple_drain( pah->s_play, &pa_error ) < 0 ) {
8639 errorStream_ << "RtApiPulse::stopStream: error draining output device, " <<
8640 pa_strerror( pa_error ) << ".";
8641 errorText_ = errorStream_.str();
8642 MUTEX_UNLOCK( &stream_.mutex );
8643 error( RtAudioError::SYSTEM_ERROR );
8648 stream_.state = STREAM_STOPPED;
8649 MUTEX_UNLOCK( &stream_.mutex );
8652 void RtApiPulse::abortStream( void )
8654 PulseAudioHandle *pah = static_cast<PulseAudioHandle*>( stream_.apiHandle );
8656 if ( stream_.state == STREAM_CLOSED ) {
8657 errorText_ = "RtApiPulse::abortStream(): the stream is not open!";
8658 error( RtAudioError::INVALID_USE );
8661 if ( stream_.state == STREAM_STOPPED ) {
8662 errorText_ = "RtApiPulse::abortStream(): the stream is already stopped!";
8663 error( RtAudioError::WARNING );
8667 stream_.state = STREAM_STOPPED;
8668 MUTEX_LOCK( &stream_.mutex );
8670 if ( pah && pah->s_play ) {
8672 if ( pa_simple_flush( pah->s_play, &pa_error ) < 0 ) {
8673 errorStream_ << "RtApiPulse::abortStream: error flushing output device, " <<
8674 pa_strerror( pa_error ) << ".";
8675 errorText_ = errorStream_.str();
8676 MUTEX_UNLOCK( &stream_.mutex );
8677 error( RtAudioError::SYSTEM_ERROR );
8682 stream_.state = STREAM_STOPPED;
8683 MUTEX_UNLOCK( &stream_.mutex );
8686 bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode,
8687 unsigned int channels, unsigned int firstChannel,
8688 unsigned int sampleRate, RtAudioFormat format,
8689 unsigned int *bufferSize, RtAudio::StreamOptions *options )
8691 PulseAudioHandle *pah = 0;
8692 unsigned long bufferBytes = 0;
8695 if ( device != 0 ) return false;
8696 if ( mode != INPUT && mode != OUTPUT ) return false;
8697 if ( channels != 1 && channels != 2 ) {
8698 errorText_ = "RtApiPulse::probeDeviceOpen: unsupported number of channels.";
8701 ss.channels = channels;
8703 if ( firstChannel != 0 ) return false;
8705 bool sr_found = false;
8706 for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr ) {
8707 if ( sampleRate == *sr ) {
8709 stream_.sampleRate = sampleRate;
8710 ss.rate = sampleRate;
8715 errorText_ = "RtApiPulse::probeDeviceOpen: unsupported sample rate.";
8720 for ( const rtaudio_pa_format_mapping_t *sf = supported_sampleformats;
8721 sf->rtaudio_format && sf->pa_format != PA_SAMPLE_INVALID; ++sf ) {
8722 if ( format == sf->rtaudio_format ) {
8724 stream_.userFormat = sf->rtaudio_format;
8725 stream_.deviceFormat[mode] = stream_.userFormat;
8726 ss.format = sf->pa_format;
8730 if ( !sf_found ) { // Use internal data format conversion.
8731 stream_.userFormat = format;
8732 stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
8733 ss.format = PA_SAMPLE_FLOAT32LE;
8736 // Set other stream parameters.
8737 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
8738 else stream_.userInterleaved = true;
8739 stream_.deviceInterleaved[mode] = true;
8740 stream_.nBuffers = 1;
8741 stream_.doByteSwap[mode] = false;
8742 stream_.nUserChannels[mode] = channels;
8743 stream_.nDeviceChannels[mode] = channels + firstChannel;
8744 stream_.channelOffset[mode] = 0;
8745 std::string streamName = "RtAudio";
8747 // Set flags for buffer conversion.
8748 stream_.doConvertBuffer[mode] = false;
8749 if ( stream_.userFormat != stream_.deviceFormat[mode] )
8750 stream_.doConvertBuffer[mode] = true;
8751 if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
8752 stream_.doConvertBuffer[mode] = true;
8754 // Allocate necessary internal buffers.
8755 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
8756 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
8757 if ( stream_.userBuffer[mode] == NULL ) {
8758 errorText_ = "RtApiPulse::probeDeviceOpen: error allocating user buffer memory.";
8761 stream_.bufferSize = *bufferSize;
8763 if ( stream_.doConvertBuffer[mode] ) {
8765 bool makeBuffer = true;
8766 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
8767 if ( mode == INPUT ) {
8768 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
8769 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
8770 if ( bufferBytes <= bytesOut ) makeBuffer = false;
8775 bufferBytes *= *bufferSize;
8776 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
8777 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
8778 if ( stream_.deviceBuffer == NULL ) {
8779 errorText_ = "RtApiPulse::probeDeviceOpen: error allocating device buffer memory.";
8785 stream_.device[mode] = device;
8787 // Setup the buffer conversion information structure.
8788 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
8790 if ( !stream_.apiHandle ) {
8791 PulseAudioHandle *pah = new PulseAudioHandle;
8793 errorText_ = "RtApiPulse::probeDeviceOpen: error allocating memory for handle.";
8797 stream_.apiHandle = pah;
8798 if ( pthread_cond_init( &pah->runnable_cv, NULL ) != 0 ) {
8799 errorText_ = "RtApiPulse::probeDeviceOpen: error creating condition variable.";
8803 pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8806 if ( options && !options->streamName.empty() ) streamName = options->streamName;
8809 pa_buffer_attr buffer_attr;
8810 buffer_attr.fragsize = bufferBytes;
8811 buffer_attr.maxlength = -1;
8813 pah->s_rec = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_RECORD, NULL, "Record", &ss, NULL, &buffer_attr, &error );
8814 if ( !pah->s_rec ) {
8815 errorText_ = "RtApiPulse::probeDeviceOpen: error connecting input to PulseAudio server.";
8820 pah->s_play = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_PLAYBACK, NULL, "Playback", &ss, NULL, NULL, &error );
8821 if ( !pah->s_play ) {
8822 errorText_ = "RtApiPulse::probeDeviceOpen: error connecting output to PulseAudio server.";
8830 if ( stream_.mode == UNINITIALIZED )
8831 stream_.mode = mode;
8832 else if ( stream_.mode == mode )
8835 stream_.mode = DUPLEX;
8837 if ( !stream_.callbackInfo.isRunning ) {
8838 stream_.callbackInfo.object = this;
8840 stream_.state = STREAM_STOPPED;
8841 // Set the thread attributes for joinable and realtime scheduling
8842 // priority (optional). The higher priority will only take affect
8843 // if the program is run as root or suid. Note, under Linux
8844 // processes with CAP_SYS_NICE privilege, a user can change
8845 // scheduling policy and priority (thus need not be root). See
8846 // POSIX "capabilities".
8847 pthread_attr_t attr;
8848 pthread_attr_init( &attr );
8849 pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
8850 #ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
8851 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
8852 stream_.callbackInfo.doRealtime = true;
8853 struct sched_param param;
8854 int priority = options->priority;
8855 int min = sched_get_priority_min( SCHED_RR );
8856 int max = sched_get_priority_max( SCHED_RR );
8857 if ( priority < min ) priority = min;
8858 else if ( priority > max ) priority = max;
8859 param.sched_priority = priority;
8861 // Set the policy BEFORE the priority. Otherwise it fails.
8862 pthread_attr_setschedpolicy(&attr, SCHED_RR);
8863 pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM);
8864 // This is definitely required. Otherwise it fails.
8865 pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED);
8866 pthread_attr_setschedparam(&attr, ¶m);
8869 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
8871 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
8874 stream_.callbackInfo.isRunning = true;
8875 int result = pthread_create( &pah->thread, &attr, pulseaudio_callback, (void *)&stream_.callbackInfo);
8876 pthread_attr_destroy(&attr);
8878 // Failed. Try instead with default attributes.
8879 result = pthread_create( &pah->thread, NULL, pulseaudio_callback, (void *)&stream_.callbackInfo);
8881 stream_.callbackInfo.isRunning = false;
8882 errorText_ = "RtApiPulse::probeDeviceOpen: error creating thread.";
8891 if ( pah && stream_.callbackInfo.isRunning ) {
8892 pthread_cond_destroy( &pah->runnable_cv );
8894 stream_.apiHandle = 0;
8897 for ( int i=0; i<2; i++ ) {
8898 if ( stream_.userBuffer[i] ) {
8899 free( stream_.userBuffer[i] );
8900 stream_.userBuffer[i] = 0;
8904 if ( stream_.deviceBuffer ) {
8905 free( stream_.deviceBuffer );
8906 stream_.deviceBuffer = 0;
8909 stream_.state = STREAM_CLOSED;
8913 //******************** End of __LINUX_PULSE__ *********************//
8916 #if defined(__LINUX_OSS__)
8919 #include <sys/ioctl.h>
8922 #include <sys/soundcard.h>
8926 static void *ossCallbackHandler(void * ptr);
8928 // A structure to hold various information related to the OSS API
8931 int id[2]; // device ids
8934 pthread_cond_t runnable;
8937 :triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
8940 RtApiOss :: RtApiOss()
8942 // Nothing to do here.
8945 RtApiOss :: ~RtApiOss()
8947 if ( stream_.state != STREAM_CLOSED ) closeStream();
8950 unsigned int RtApiOss :: getDeviceCount( void )
8952 int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
8953 if ( mixerfd == -1 ) {
8954 errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'.";
8955 error( RtAudioError::WARNING );
8959 oss_sysinfo sysinfo;
8960 if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) {
8962 errorText_ = "RtApiOss::getDeviceCount: error getting sysinfo, OSS version >= 4.0 is required.";
8963 error( RtAudioError::WARNING );
8968 return sysinfo.numaudios;
8971 RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device )
8973 RtAudio::DeviceInfo info;
8974 info.probed = false;
8976 int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
8977 if ( mixerfd == -1 ) {
8978 errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'.";
8979 error( RtAudioError::WARNING );
8983 oss_sysinfo sysinfo;
8984 int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
8985 if ( result == -1 ) {
8987 errorText_ = "RtApiOss::getDeviceInfo: error getting sysinfo, OSS version >= 4.0 is required.";
8988 error( RtAudioError::WARNING );
8992 unsigned nDevices = sysinfo.numaudios;
8993 if ( nDevices == 0 ) {
8995 errorText_ = "RtApiOss::getDeviceInfo: no devices found!";
8996 error( RtAudioError::INVALID_USE );
9000 if ( device >= nDevices ) {
9002 errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!";
9003 error( RtAudioError::INVALID_USE );
9007 oss_audioinfo ainfo;
9009 result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
9011 if ( result == -1 ) {
9012 errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
9013 errorText_ = errorStream_.str();
9014 error( RtAudioError::WARNING );
9019 if ( ainfo.caps & PCM_CAP_OUTPUT ) info.outputChannels = ainfo.max_channels;
9020 if ( ainfo.caps & PCM_CAP_INPUT ) info.inputChannels = ainfo.max_channels;
9021 if ( ainfo.caps & PCM_CAP_DUPLEX ) {
9022 if ( info.outputChannels > 0 && info.inputChannels > 0 && ainfo.caps & PCM_CAP_DUPLEX )
9023 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
9026 // Probe data formats ... do for input
9027 unsigned long mask = ainfo.iformats;
9028 if ( mask & AFMT_S16_LE || mask & AFMT_S16_BE )
9029 info.nativeFormats |= RTAUDIO_SINT16;
9030 if ( mask & AFMT_S8 )
9031 info.nativeFormats |= RTAUDIO_SINT8;
9032 if ( mask & AFMT_S32_LE || mask & AFMT_S32_BE )
9033 info.nativeFormats |= RTAUDIO_SINT32;
9035 if ( mask & AFMT_FLOAT )
9036 info.nativeFormats |= RTAUDIO_FLOAT32;
9038 if ( mask & AFMT_S24_LE || mask & AFMT_S24_BE )
9039 info.nativeFormats |= RTAUDIO_SINT24;
9041 // Check that we have at least one supported format
9042 if ( info.nativeFormats == 0 ) {
9043 errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio.";
9044 errorText_ = errorStream_.str();
9045 error( RtAudioError::WARNING );
9049 // Probe the supported sample rates.
9050 info.sampleRates.clear();
9051 if ( ainfo.nrates ) {
9052 for ( unsigned int i=0; i<ainfo.nrates; i++ ) {
9053 for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
9054 if ( ainfo.rates[i] == SAMPLE_RATES[k] ) {
9055 info.sampleRates.push_back( SAMPLE_RATES[k] );
9057 if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
9058 info.preferredSampleRate = SAMPLE_RATES[k];
9066 // Check min and max rate values;
9067 for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
9068 if ( ainfo.min_rate <= (int) SAMPLE_RATES[k] && ainfo.max_rate >= (int) SAMPLE_RATES[k] ) {
9069 info.sampleRates.push_back( SAMPLE_RATES[k] );
9071 if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
9072 info.preferredSampleRate = SAMPLE_RATES[k];
9077 if ( info.sampleRates.size() == 0 ) {
9078 errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ").";
9079 errorText_ = errorStream_.str();
9080 error( RtAudioError::WARNING );
9084 info.name = ainfo.name;
9091 bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
9092 unsigned int firstChannel, unsigned int sampleRate,
9093 RtAudioFormat format, unsigned int *bufferSize,
9094 RtAudio::StreamOptions *options )
9096 int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
9097 if ( mixerfd == -1 ) {
9098 errorText_ = "RtApiOss::probeDeviceOpen: error opening '/dev/mixer'.";
9102 oss_sysinfo sysinfo;
9103 int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
9104 if ( result == -1 ) {
9106 errorText_ = "RtApiOss::probeDeviceOpen: error getting sysinfo, OSS version >= 4.0 is required.";
9110 unsigned nDevices = sysinfo.numaudios;
9111 if ( nDevices == 0 ) {
9112 // This should not happen because a check is made before this function is called.
9114 errorText_ = "RtApiOss::probeDeviceOpen: no devices found!";
9118 if ( device >= nDevices ) {
9119 // This should not happen because a check is made before this function is called.
9121 errorText_ = "RtApiOss::probeDeviceOpen: device ID is invalid!";
9125 oss_audioinfo ainfo;
9127 result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
9129 if ( result == -1 ) {
9130 errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
9131 errorText_ = errorStream_.str();
9135 // Check if device supports input or output
9136 if ( ( mode == OUTPUT && !( ainfo.caps & PCM_CAP_OUTPUT ) ) ||
9137 ( mode == INPUT && !( ainfo.caps & PCM_CAP_INPUT ) ) ) {
9138 if ( mode == OUTPUT )
9139 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support output.";
9141 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support input.";
9142 errorText_ = errorStream_.str();
9147 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9148 if ( mode == OUTPUT )
9150 else { // mode == INPUT
9151 if (stream_.mode == OUTPUT && stream_.device[0] == device) {
9152 // We just set the same device for playback ... close and reopen for duplex (OSS only).
9153 close( handle->id[0] );
9155 if ( !( ainfo.caps & PCM_CAP_DUPLEX ) ) {
9156 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support duplex mode.";
9157 errorText_ = errorStream_.str();
9160 // Check that the number previously set channels is the same.
9161 if ( stream_.nUserChannels[0] != channels ) {
9162 errorStream_ << "RtApiOss::probeDeviceOpen: input/output channels must be equal for OSS duplex device (" << ainfo.name << ").";
9163 errorText_ = errorStream_.str();
9172 // Set exclusive access if specified.
9173 if ( options && options->flags & RTAUDIO_HOG_DEVICE ) flags |= O_EXCL;
9175 // Try to open the device.
9177 fd = open( ainfo.devnode, flags, 0 );
9179 if ( errno == EBUSY )
9180 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") is busy.";
9182 errorStream_ << "RtApiOss::probeDeviceOpen: error opening device (" << ainfo.name << ").";
9183 errorText_ = errorStream_.str();
9187 // For duplex operation, specifically set this mode (this doesn't seem to work).
9189 if ( flags | O_RDWR ) {
9190 result = ioctl( fd, SNDCTL_DSP_SETDUPLEX, NULL );
9191 if ( result == -1) {
9192 errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ").";
9193 errorText_ = errorStream_.str();
9199 // Check the device channel support.
9200 stream_.nUserChannels[mode] = channels;
9201 if ( ainfo.max_channels < (int)(channels + firstChannel) ) {
9203 errorStream_ << "RtApiOss::probeDeviceOpen: the device (" << ainfo.name << ") does not support requested channel parameters.";
9204 errorText_ = errorStream_.str();
9208 // Set the number of channels.
9209 int deviceChannels = channels + firstChannel;
9210 result = ioctl( fd, SNDCTL_DSP_CHANNELS, &deviceChannels );
9211 if ( result == -1 || deviceChannels < (int)(channels + firstChannel) ) {
9213 errorStream_ << "RtApiOss::probeDeviceOpen: error setting channel parameters on device (" << ainfo.name << ").";
9214 errorText_ = errorStream_.str();
9217 stream_.nDeviceChannels[mode] = deviceChannels;
9219 // Get the data format mask
9221 result = ioctl( fd, SNDCTL_DSP_GETFMTS, &mask );
9222 if ( result == -1 ) {
9224 errorStream_ << "RtApiOss::probeDeviceOpen: error getting device (" << ainfo.name << ") data formats.";
9225 errorText_ = errorStream_.str();
9229 // Determine how to set the device format.
9230 stream_.userFormat = format;
9231 int deviceFormat = -1;
9232 stream_.doByteSwap[mode] = false;
9233 if ( format == RTAUDIO_SINT8 ) {
9234 if ( mask & AFMT_S8 ) {
9235 deviceFormat = AFMT_S8;
9236 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
9239 else if ( format == RTAUDIO_SINT16 ) {
9240 if ( mask & AFMT_S16_NE ) {
9241 deviceFormat = AFMT_S16_NE;
9242 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
9244 else if ( mask & AFMT_S16_OE ) {
9245 deviceFormat = AFMT_S16_OE;
9246 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
9247 stream_.doByteSwap[mode] = true;
9250 else if ( format == RTAUDIO_SINT24 ) {
9251 if ( mask & AFMT_S24_NE ) {
9252 deviceFormat = AFMT_S24_NE;
9253 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
9255 else if ( mask & AFMT_S24_OE ) {
9256 deviceFormat = AFMT_S24_OE;
9257 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
9258 stream_.doByteSwap[mode] = true;
9261 else if ( format == RTAUDIO_SINT32 ) {
9262 if ( mask & AFMT_S32_NE ) {
9263 deviceFormat = AFMT_S32_NE;
9264 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
9266 else if ( mask & AFMT_S32_OE ) {
9267 deviceFormat = AFMT_S32_OE;
9268 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
9269 stream_.doByteSwap[mode] = true;
9273 if ( deviceFormat == -1 ) {
9274 // The user requested format is not natively supported by the device.
9275 if ( mask & AFMT_S16_NE ) {
9276 deviceFormat = AFMT_S16_NE;
9277 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
9279 else if ( mask & AFMT_S32_NE ) {
9280 deviceFormat = AFMT_S32_NE;
9281 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
9283 else if ( mask & AFMT_S24_NE ) {
9284 deviceFormat = AFMT_S24_NE;
9285 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
9287 else if ( mask & AFMT_S16_OE ) {
9288 deviceFormat = AFMT_S16_OE;
9289 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
9290 stream_.doByteSwap[mode] = true;
9292 else if ( mask & AFMT_S32_OE ) {
9293 deviceFormat = AFMT_S32_OE;
9294 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
9295 stream_.doByteSwap[mode] = true;
9297 else if ( mask & AFMT_S24_OE ) {
9298 deviceFormat = AFMT_S24_OE;
9299 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
9300 stream_.doByteSwap[mode] = true;
9302 else if ( mask & AFMT_S8) {
9303 deviceFormat = AFMT_S8;
9304 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
9308 if ( stream_.deviceFormat[mode] == 0 ) {
9309 // This really shouldn't happen ...
9311 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") data format not supported by RtAudio.";
9312 errorText_ = errorStream_.str();
9316 // Set the data format.
9317 int temp = deviceFormat;
9318 result = ioctl( fd, SNDCTL_DSP_SETFMT, &deviceFormat );
9319 if ( result == -1 || deviceFormat != temp ) {
9321 errorStream_ << "RtApiOss::probeDeviceOpen: error setting data format on device (" << ainfo.name << ").";
9322 errorText_ = errorStream_.str();
9326 // Attempt to set the buffer size. According to OSS, the minimum
9327 // number of buffers is two. The supposed minimum buffer size is 16
9328 // bytes, so that will be our lower bound. The argument to this
9329 // call is in the form 0xMMMMSSSS (hex), where the buffer size (in
9330 // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM.
9331 // We'll check the actual value used near the end of the setup
9333 int ossBufferBytes = *bufferSize * formatBytes( stream_.deviceFormat[mode] ) * deviceChannels;
9334 if ( ossBufferBytes < 16 ) ossBufferBytes = 16;
9336 if ( options ) buffers = options->numberOfBuffers;
9337 if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) buffers = 2;
9338 if ( buffers < 2 ) buffers = 3;
9339 temp = ((int) buffers << 16) + (int)( log10( (double)ossBufferBytes ) / log10( 2.0 ) );
9340 result = ioctl( fd, SNDCTL_DSP_SETFRAGMENT, &temp );
9341 if ( result == -1 ) {
9343 errorStream_ << "RtApiOss::probeDeviceOpen: error setting buffer size on device (" << ainfo.name << ").";
9344 errorText_ = errorStream_.str();
9347 stream_.nBuffers = buffers;
9349 // Save buffer size (in sample frames).
9350 *bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels );
9351 stream_.bufferSize = *bufferSize;
9353 // Set the sample rate.
9354 int srate = sampleRate;
9355 result = ioctl( fd, SNDCTL_DSP_SPEED, &srate );
9356 if ( result == -1 ) {
9358 errorStream_ << "RtApiOss::probeDeviceOpen: error setting sample rate (" << sampleRate << ") on device (" << ainfo.name << ").";
9359 errorText_ = errorStream_.str();
9363 // Verify the sample rate setup worked.
9364 if ( abs( srate - (int)sampleRate ) > 100 ) {
9366 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ").";
9367 errorText_ = errorStream_.str();
9370 stream_.sampleRate = sampleRate;
9372 if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device) {
9373 // We're doing duplex setup here.
9374 stream_.deviceFormat[0] = stream_.deviceFormat[1];
9375 stream_.nDeviceChannels[0] = deviceChannels;
9378 // Set interleaving parameters.
9379 stream_.userInterleaved = true;
9380 stream_.deviceInterleaved[mode] = true;
9381 if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
9382 stream_.userInterleaved = false;
9384 // Set flags for buffer conversion
9385 stream_.doConvertBuffer[mode] = false;
9386 if ( stream_.userFormat != stream_.deviceFormat[mode] )
9387 stream_.doConvertBuffer[mode] = true;
9388 if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
9389 stream_.doConvertBuffer[mode] = true;
9390 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
9391 stream_.nUserChannels[mode] > 1 )
9392 stream_.doConvertBuffer[mode] = true;
9394 // Allocate the stream handles if necessary and then save.
9395 if ( stream_.apiHandle == 0 ) {
9397 handle = new OssHandle;
9399 catch ( std::bad_alloc& ) {
9400 errorText_ = "RtApiOss::probeDeviceOpen: error allocating OssHandle memory.";
9404 if ( pthread_cond_init( &handle->runnable, NULL ) ) {
9405 errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable.";
9409 stream_.apiHandle = (void *) handle;
9412 handle = (OssHandle *) stream_.apiHandle;
9414 handle->id[mode] = fd;
9416 // Allocate necessary internal buffers.
9417 unsigned long bufferBytes;
9418 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
9419 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
9420 if ( stream_.userBuffer[mode] == NULL ) {
9421 errorText_ = "RtApiOss::probeDeviceOpen: error allocating user buffer memory.";
9425 if ( stream_.doConvertBuffer[mode] ) {
9427 bool makeBuffer = true;
9428 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
9429 if ( mode == INPUT ) {
9430 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
9431 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
9432 if ( bufferBytes <= bytesOut ) makeBuffer = false;
9437 bufferBytes *= *bufferSize;
9438 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
9439 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
9440 if ( stream_.deviceBuffer == NULL ) {
9441 errorText_ = "RtApiOss::probeDeviceOpen: error allocating device buffer memory.";
9447 stream_.device[mode] = device;
9448 stream_.state = STREAM_STOPPED;
9450 // Setup the buffer conversion information structure.
9451 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
9453 // Setup thread if necessary.
9454 if ( stream_.mode == OUTPUT && mode == INPUT ) {
9455 // We had already set up an output stream.
9456 stream_.mode = DUPLEX;
9457 if ( stream_.device[0] == device ) handle->id[0] = fd;
9460 stream_.mode = mode;
9462 // Setup callback thread.
9463 stream_.callbackInfo.object = (void *) this;
9465 // Set the thread attributes for joinable and realtime scheduling
9466 // priority. The higher priority will only take affect if the
9467 // program is run as root or suid.
9468 pthread_attr_t attr;
9469 pthread_attr_init( &attr );
9470 pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
9471 #ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
9472 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
9473 stream_.callbackInfo.doRealtime = true;
9474 struct sched_param param;
9475 int priority = options->priority;
9476 int min = sched_get_priority_min( SCHED_RR );
9477 int max = sched_get_priority_max( SCHED_RR );
9478 if ( priority < min ) priority = min;
9479 else if ( priority > max ) priority = max;
9480 param.sched_priority = priority;
9482 // Set the policy BEFORE the priority. Otherwise it fails.
9483 pthread_attr_setschedpolicy(&attr, SCHED_RR);
9484 pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM);
9485 // This is definitely required. Otherwise it fails.
9486 pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED);
9487 pthread_attr_setschedparam(&attr, ¶m);
9490 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
9492 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
9495 stream_.callbackInfo.isRunning = true;
9496 result = pthread_create( &stream_.callbackInfo.thread, &attr, ossCallbackHandler, &stream_.callbackInfo );
9497 pthread_attr_destroy( &attr );
9499 // Failed. Try instead with default attributes.
9500 result = pthread_create( &stream_.callbackInfo.thread, NULL, ossCallbackHandler, &stream_.callbackInfo );
9502 stream_.callbackInfo.isRunning = false;
9503 errorText_ = "RtApiOss::error creating callback thread!";
9513 pthread_cond_destroy( &handle->runnable );
9514 if ( handle->id[0] ) close( handle->id[0] );
9515 if ( handle->id[1] ) close( handle->id[1] );
9517 stream_.apiHandle = 0;
9520 for ( int i=0; i<2; i++ ) {
9521 if ( stream_.userBuffer[i] ) {
9522 free( stream_.userBuffer[i] );
9523 stream_.userBuffer[i] = 0;
9527 if ( stream_.deviceBuffer ) {
9528 free( stream_.deviceBuffer );
9529 stream_.deviceBuffer = 0;
9532 stream_.state = STREAM_CLOSED;
9536 void RtApiOss :: closeStream()
9538 if ( stream_.state == STREAM_CLOSED ) {
9539 errorText_ = "RtApiOss::closeStream(): no open stream to close!";
9540 error( RtAudioError::WARNING );
9544 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9545 stream_.callbackInfo.isRunning = false;
9546 MUTEX_LOCK( &stream_.mutex );
9547 if ( stream_.state == STREAM_STOPPED )
9548 pthread_cond_signal( &handle->runnable );
9549 MUTEX_UNLOCK( &stream_.mutex );
9550 pthread_join( stream_.callbackInfo.thread, NULL );
9552 if ( stream_.state == STREAM_RUNNING ) {
9553 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
9554 ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
9556 ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
9557 stream_.state = STREAM_STOPPED;
9561 pthread_cond_destroy( &handle->runnable );
9562 if ( handle->id[0] ) close( handle->id[0] );
9563 if ( handle->id[1] ) close( handle->id[1] );
9565 stream_.apiHandle = 0;
9568 for ( int i=0; i<2; i++ ) {
9569 if ( stream_.userBuffer[i] ) {
9570 free( stream_.userBuffer[i] );
9571 stream_.userBuffer[i] = 0;
9575 if ( stream_.deviceBuffer ) {
9576 free( stream_.deviceBuffer );
9577 stream_.deviceBuffer = 0;
9580 stream_.mode = UNINITIALIZED;
9581 stream_.state = STREAM_CLOSED;
9584 void RtApiOss :: startStream()
9587 if ( stream_.state == STREAM_RUNNING ) {
9588 errorText_ = "RtApiOss::startStream(): the stream is already running!";
9589 error( RtAudioError::WARNING );
9593 MUTEX_LOCK( &stream_.mutex );
9595 stream_.state = STREAM_RUNNING;
9597 // No need to do anything else here ... OSS automatically starts
9598 // when fed samples.
9600 MUTEX_UNLOCK( &stream_.mutex );
9602 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9603 pthread_cond_signal( &handle->runnable );
9606 void RtApiOss :: stopStream()
9609 if ( stream_.state == STREAM_STOPPED ) {
9610 errorText_ = "RtApiOss::stopStream(): the stream is already stopped!";
9611 error( RtAudioError::WARNING );
9615 MUTEX_LOCK( &stream_.mutex );
9617 // The state might change while waiting on a mutex.
9618 if ( stream_.state == STREAM_STOPPED ) {
9619 MUTEX_UNLOCK( &stream_.mutex );
9624 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9625 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
9627 // Flush the output with zeros a few times.
9630 RtAudioFormat format;
9632 if ( stream_.doConvertBuffer[0] ) {
9633 buffer = stream_.deviceBuffer;
9634 samples = stream_.bufferSize * stream_.nDeviceChannels[0];
9635 format = stream_.deviceFormat[0];
9638 buffer = stream_.userBuffer[0];
9639 samples = stream_.bufferSize * stream_.nUserChannels[0];
9640 format = stream_.userFormat;
9643 memset( buffer, 0, samples * formatBytes(format) );
9644 for ( unsigned int i=0; i<stream_.nBuffers+1; i++ ) {
9645 result = write( handle->id[0], buffer, samples * formatBytes(format) );
9646 if ( result == -1 ) {
9647 errorText_ = "RtApiOss::stopStream: audio write error.";
9648 error( RtAudioError::WARNING );
9652 result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
9653 if ( result == -1 ) {
9654 errorStream_ << "RtApiOss::stopStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
9655 errorText_ = errorStream_.str();
9658 handle->triggered = false;
9661 if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
9662 result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
9663 if ( result == -1 ) {
9664 errorStream_ << "RtApiOss::stopStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
9665 errorText_ = errorStream_.str();
9671 stream_.state = STREAM_STOPPED;
9672 MUTEX_UNLOCK( &stream_.mutex );
9674 if ( result != -1 ) return;
9675 error( RtAudioError::SYSTEM_ERROR );
9678 void RtApiOss :: abortStream()
9681 if ( stream_.state == STREAM_STOPPED ) {
9682 errorText_ = "RtApiOss::abortStream(): the stream is already stopped!";
9683 error( RtAudioError::WARNING );
9687 MUTEX_LOCK( &stream_.mutex );
9689 // The state might change while waiting on a mutex.
9690 if ( stream_.state == STREAM_STOPPED ) {
9691 MUTEX_UNLOCK( &stream_.mutex );
9696 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9697 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
9698 result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
9699 if ( result == -1 ) {
9700 errorStream_ << "RtApiOss::abortStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
9701 errorText_ = errorStream_.str();
9704 handle->triggered = false;
9707 if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
9708 result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
9709 if ( result == -1 ) {
9710 errorStream_ << "RtApiOss::abortStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
9711 errorText_ = errorStream_.str();
9717 stream_.state = STREAM_STOPPED;
9718 MUTEX_UNLOCK( &stream_.mutex );
9720 if ( result != -1 ) return;
9721 error( RtAudioError::SYSTEM_ERROR );
9724 void RtApiOss :: callbackEvent()
9726 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9727 if ( stream_.state == STREAM_STOPPED ) {
9728 MUTEX_LOCK( &stream_.mutex );
9729 pthread_cond_wait( &handle->runnable, &stream_.mutex );
9730 if ( stream_.state != STREAM_RUNNING ) {
9731 MUTEX_UNLOCK( &stream_.mutex );
9734 MUTEX_UNLOCK( &stream_.mutex );
9737 if ( stream_.state == STREAM_CLOSED ) {
9738 errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!";
9739 error( RtAudioError::WARNING );
9743 // Invoke user callback to get fresh output data.
9744 int doStopStream = 0;
9745 RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
9746 double streamTime = getStreamTime();
9747 RtAudioStreamStatus status = 0;
9748 if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
9749 status |= RTAUDIO_OUTPUT_UNDERFLOW;
9750 handle->xrun[0] = false;
9752 if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
9753 status |= RTAUDIO_INPUT_OVERFLOW;
9754 handle->xrun[1] = false;
9756 doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
9757 stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
9758 if ( doStopStream == 2 ) {
9759 this->abortStream();
9763 MUTEX_LOCK( &stream_.mutex );
9765 // The state might change while waiting on a mutex.
9766 if ( stream_.state == STREAM_STOPPED ) goto unlock;
9771 RtAudioFormat format;
9773 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
9775 // Setup parameters and do buffer conversion if necessary.
9776 if ( stream_.doConvertBuffer[0] ) {
9777 buffer = stream_.deviceBuffer;
9778 convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
9779 samples = stream_.bufferSize * stream_.nDeviceChannels[0];
9780 format = stream_.deviceFormat[0];
9783 buffer = stream_.userBuffer[0];
9784 samples = stream_.bufferSize * stream_.nUserChannels[0];
9785 format = stream_.userFormat;
9788 // Do byte swapping if necessary.
9789 if ( stream_.doByteSwap[0] )
9790 byteSwapBuffer( buffer, samples, format );
9792 if ( stream_.mode == DUPLEX && handle->triggered == false ) {
9794 ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
9795 result = write( handle->id[0], buffer, samples * formatBytes(format) );
9796 trig = PCM_ENABLE_INPUT|PCM_ENABLE_OUTPUT;
9797 ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
9798 handle->triggered = true;
9801 // Write samples to device.
9802 result = write( handle->id[0], buffer, samples * formatBytes(format) );
9804 if ( result == -1 ) {
9805 // We'll assume this is an underrun, though there isn't a
9806 // specific means for determining that.
9807 handle->xrun[0] = true;
9808 errorText_ = "RtApiOss::callbackEvent: audio write error.";
9809 error( RtAudioError::WARNING );
9810 // Continue on to input section.
9814 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
9816 // Setup parameters.
9817 if ( stream_.doConvertBuffer[1] ) {
9818 buffer = stream_.deviceBuffer;
9819 samples = stream_.bufferSize * stream_.nDeviceChannels[1];
9820 format = stream_.deviceFormat[1];
9823 buffer = stream_.userBuffer[1];
9824 samples = stream_.bufferSize * stream_.nUserChannels[1];
9825 format = stream_.userFormat;
9828 // Read samples from device.
9829 result = read( handle->id[1], buffer, samples * formatBytes(format) );
9831 if ( result == -1 ) {
9832 // We'll assume this is an overrun, though there isn't a
9833 // specific means for determining that.
9834 handle->xrun[1] = true;
9835 errorText_ = "RtApiOss::callbackEvent: audio read error.";
9836 error( RtAudioError::WARNING );
9840 // Do byte swapping if necessary.
9841 if ( stream_.doByteSwap[1] )
9842 byteSwapBuffer( buffer, samples, format );
9844 // Do buffer conversion if necessary.
9845 if ( stream_.doConvertBuffer[1] )
9846 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
9850 MUTEX_UNLOCK( &stream_.mutex );
9852 RtApi::tickStreamTime();
9853 if ( doStopStream == 1 ) this->stopStream();
9856 static void *ossCallbackHandler( void *ptr )
9858 CallbackInfo *info = (CallbackInfo *) ptr;
9859 RtApiOss *object = (RtApiOss *) info->object;
9860 bool *isRunning = &info->isRunning;
9862 #ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
9863 if (info->doRealtime) {
9864 std::cerr << "RtAudio oss: " <<
9865 (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") <<
9866 "running realtime scheduling" << std::endl;
9870 while ( *isRunning == true ) {
9871 pthread_testcancel();
9872 object->callbackEvent();
9875 pthread_exit( NULL );
9878 //******************** End of __LINUX_OSS__ *********************//
9882 // *************************************************** //
9884 // Protected common (OS-independent) RtAudio methods.
9886 // *************************************************** //
9888 // This method can be modified to control the behavior of error
9889 // message printing.
9890 void RtApi :: error( RtAudioError::Type type )
9892 errorStream_.str(""); // clear the ostringstream
9894 RtAudioErrorCallback errorCallback = (RtAudioErrorCallback) stream_.callbackInfo.errorCallback;
9895 if ( errorCallback ) {
9896 // abortStream() can generate new error messages. Ignore them. Just keep original one.
9898 if ( firstErrorOccurred_ )
9901 firstErrorOccurred_ = true;
9902 const std::string errorMessage = errorText_;
9904 if ( type != RtAudioError::WARNING && stream_.state != STREAM_STOPPED) {
9905 stream_.callbackInfo.isRunning = false; // exit from the thread
9909 errorCallback( type, errorMessage );
9910 firstErrorOccurred_ = false;
9914 if ( type == RtAudioError::WARNING && showWarnings_ == true )
9915 std::cerr << '\n' << errorText_ << "\n\n";
9916 else if ( type != RtAudioError::WARNING )
9917 throw( RtAudioError( errorText_, type ) );
9920 void RtApi :: verifyStream()
9922 if ( stream_.state == STREAM_CLOSED ) {
9923 errorText_ = "RtApi:: a stream is not open!";
9924 error( RtAudioError::INVALID_USE );
9928 void RtApi :: clearStreamInfo()
9930 stream_.mode = UNINITIALIZED;
9931 stream_.state = STREAM_CLOSED;
9932 stream_.sampleRate = 0;
9933 stream_.bufferSize = 0;
9934 stream_.nBuffers = 0;
9935 stream_.userFormat = 0;
9936 stream_.userInterleaved = true;
9937 stream_.streamTime = 0.0;
9938 stream_.apiHandle = 0;
9939 stream_.deviceBuffer = 0;
9940 stream_.callbackInfo.callback = 0;
9941 stream_.callbackInfo.userData = 0;
9942 stream_.callbackInfo.isRunning = false;
9943 stream_.callbackInfo.errorCallback = 0;
9944 for ( int i=0; i<2; i++ ) {
9945 stream_.device[i] = 11111;
9946 stream_.doConvertBuffer[i] = false;
9947 stream_.deviceInterleaved[i] = true;
9948 stream_.doByteSwap[i] = false;
9949 stream_.nUserChannels[i] = 0;
9950 stream_.nDeviceChannels[i] = 0;
9951 stream_.channelOffset[i] = 0;
9952 stream_.deviceFormat[i] = 0;
9953 stream_.latency[i] = 0;
9954 stream_.userBuffer[i] = 0;
9955 stream_.convertInfo[i].channels = 0;
9956 stream_.convertInfo[i].inJump = 0;
9957 stream_.convertInfo[i].outJump = 0;
9958 stream_.convertInfo[i].inFormat = 0;
9959 stream_.convertInfo[i].outFormat = 0;
9960 stream_.convertInfo[i].inOffset.clear();
9961 stream_.convertInfo[i].outOffset.clear();
9965 unsigned int RtApi :: formatBytes( RtAudioFormat format )
9967 if ( format == RTAUDIO_SINT16 )
9969 else if ( format == RTAUDIO_SINT32 || format == RTAUDIO_FLOAT32 )
9971 else if ( format == RTAUDIO_FLOAT64 )
9973 else if ( format == RTAUDIO_SINT24 )
9975 else if ( format == RTAUDIO_SINT8 )
9978 errorText_ = "RtApi::formatBytes: undefined format.";
9979 error( RtAudioError::WARNING );
9984 void RtApi :: setConvertInfo( StreamMode mode, unsigned int firstChannel )
9986 if ( mode == INPUT ) { // convert device to user buffer
9987 stream_.convertInfo[mode].inJump = stream_.nDeviceChannels[1];
9988 stream_.convertInfo[mode].outJump = stream_.nUserChannels[1];
9989 stream_.convertInfo[mode].inFormat = stream_.deviceFormat[1];
9990 stream_.convertInfo[mode].outFormat = stream_.userFormat;
9992 else { // convert user to device buffer
9993 stream_.convertInfo[mode].inJump = stream_.nUserChannels[0];
9994 stream_.convertInfo[mode].outJump = stream_.nDeviceChannels[0];
9995 stream_.convertInfo[mode].inFormat = stream_.userFormat;
9996 stream_.convertInfo[mode].outFormat = stream_.deviceFormat[0];
9999 if ( stream_.convertInfo[mode].inJump < stream_.convertInfo[mode].outJump )
10000 stream_.convertInfo[mode].channels = stream_.convertInfo[mode].inJump;
10002 stream_.convertInfo[mode].channels = stream_.convertInfo[mode].outJump;
10004 // Set up the interleave/deinterleave offsets.
10005 if ( stream_.deviceInterleaved[mode] != stream_.userInterleaved ) {
10006 if ( ( mode == OUTPUT && stream_.deviceInterleaved[mode] ) ||
10007 ( mode == INPUT && stream_.userInterleaved ) ) {
10008 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
10009 stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
10010 stream_.convertInfo[mode].outOffset.push_back( k );
10011 stream_.convertInfo[mode].inJump = 1;
10015 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
10016 stream_.convertInfo[mode].inOffset.push_back( k );
10017 stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
10018 stream_.convertInfo[mode].outJump = 1;
10022 else { // no (de)interleaving
10023 if ( stream_.userInterleaved ) {
10024 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
10025 stream_.convertInfo[mode].inOffset.push_back( k );
10026 stream_.convertInfo[mode].outOffset.push_back( k );
10030 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
10031 stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
10032 stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
10033 stream_.convertInfo[mode].inJump = 1;
10034 stream_.convertInfo[mode].outJump = 1;
10039 // Add channel offset.
10040 if ( firstChannel > 0 ) {
10041 if ( stream_.deviceInterleaved[mode] ) {
10042 if ( mode == OUTPUT ) {
10043 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
10044 stream_.convertInfo[mode].outOffset[k] += firstChannel;
10047 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
10048 stream_.convertInfo[mode].inOffset[k] += firstChannel;
10052 if ( mode == OUTPUT ) {
10053 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
10054 stream_.convertInfo[mode].outOffset[k] += ( firstChannel * stream_.bufferSize );
10057 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
10058 stream_.convertInfo[mode].inOffset[k] += ( firstChannel * stream_.bufferSize );
10064 void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info )
10066 // This function does format conversion, input/output channel compensation, and
10067 // data interleaving/deinterleaving. 24-bit integers are assumed to occupy
10068 // the lower three bytes of a 32-bit integer.
10070 // Clear our device buffer when in/out duplex device channels are different
10071 if ( outBuffer == stream_.deviceBuffer && stream_.mode == DUPLEX &&
10072 ( stream_.nDeviceChannels[0] < stream_.nDeviceChannels[1] ) )
10073 memset( outBuffer, 0, stream_.bufferSize * info.outJump * formatBytes( info.outFormat ) );
10076 if (info.outFormat == RTAUDIO_FLOAT64) {
10078 Float64 *out = (Float64 *)outBuffer;
10080 if (info.inFormat == RTAUDIO_SINT8) {
10081 signed char *in = (signed char *)inBuffer;
10082 scale = 1.0 / 127.5;
10083 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10084 for (j=0; j<info.channels; j++) {
10085 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
10086 out[info.outOffset[j]] += 0.5;
10087 out[info.outOffset[j]] *= scale;
10090 out += info.outJump;
10093 else if (info.inFormat == RTAUDIO_SINT16) {
10094 Int16 *in = (Int16 *)inBuffer;
10095 scale = 1.0 / 32767.5;
10096 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10097 for (j=0; j<info.channels; j++) {
10098 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
10099 out[info.outOffset[j]] += 0.5;
10100 out[info.outOffset[j]] *= scale;
10103 out += info.outJump;
10106 else if (info.inFormat == RTAUDIO_SINT24) {
10107 Int24 *in = (Int24 *)inBuffer;
10108 scale = 1.0 / 8388607.5;
10109 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10110 for (j=0; j<info.channels; j++) {
10111 out[info.outOffset[j]] = (Float64) (in[info.inOffset[j]].asInt());
10112 out[info.outOffset[j]] += 0.5;
10113 out[info.outOffset[j]] *= scale;
10116 out += info.outJump;
10119 else if (info.inFormat == RTAUDIO_SINT32) {
10120 Int32 *in = (Int32 *)inBuffer;
10121 scale = 1.0 / 2147483647.5;
10122 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10123 for (j=0; j<info.channels; j++) {
10124 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
10125 out[info.outOffset[j]] += 0.5;
10126 out[info.outOffset[j]] *= scale;
10129 out += info.outJump;
10132 else if (info.inFormat == RTAUDIO_FLOAT32) {
10133 Float32 *in = (Float32 *)inBuffer;
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]];
10139 out += info.outJump;
10142 else if (info.inFormat == RTAUDIO_FLOAT64) {
10143 // Channel compensation and/or (de)interleaving only.
10144 Float64 *in = (Float64 *)inBuffer;
10145 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10146 for (j=0; j<info.channels; j++) {
10147 out[info.outOffset[j]] = in[info.inOffset[j]];
10150 out += info.outJump;
10154 else if (info.outFormat == RTAUDIO_FLOAT32) {
10156 Float32 *out = (Float32 *)outBuffer;
10158 if (info.inFormat == RTAUDIO_SINT8) {
10159 signed char *in = (signed char *)inBuffer;
10160 scale = (Float32) ( 1.0 / 127.5 );
10161 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10162 for (j=0; j<info.channels; j++) {
10163 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
10164 out[info.outOffset[j]] += 0.5;
10165 out[info.outOffset[j]] *= scale;
10168 out += info.outJump;
10171 else if (info.inFormat == RTAUDIO_SINT16) {
10172 Int16 *in = (Int16 *)inBuffer;
10173 scale = (Float32) ( 1.0 / 32767.5 );
10174 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10175 for (j=0; j<info.channels; j++) {
10176 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
10177 out[info.outOffset[j]] += 0.5;
10178 out[info.outOffset[j]] *= scale;
10181 out += info.outJump;
10184 else if (info.inFormat == RTAUDIO_SINT24) {
10185 Int24 *in = (Int24 *)inBuffer;
10186 scale = (Float32) ( 1.0 / 8388607.5 );
10187 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10188 for (j=0; j<info.channels; j++) {
10189 out[info.outOffset[j]] = (Float32) (in[info.inOffset[j]].asInt());
10190 out[info.outOffset[j]] += 0.5;
10191 out[info.outOffset[j]] *= scale;
10194 out += info.outJump;
10197 else if (info.inFormat == RTAUDIO_SINT32) {
10198 Int32 *in = (Int32 *)inBuffer;
10199 scale = (Float32) ( 1.0 / 2147483647.5 );
10200 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10201 for (j=0; j<info.channels; j++) {
10202 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
10203 out[info.outOffset[j]] += 0.5;
10204 out[info.outOffset[j]] *= scale;
10207 out += info.outJump;
10210 else if (info.inFormat == RTAUDIO_FLOAT32) {
10211 // Channel compensation and/or (de)interleaving only.
10212 Float32 *in = (Float32 *)inBuffer;
10213 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10214 for (j=0; j<info.channels; j++) {
10215 out[info.outOffset[j]] = in[info.inOffset[j]];
10218 out += info.outJump;
10221 else if (info.inFormat == RTAUDIO_FLOAT64) {
10222 Float64 *in = (Float64 *)inBuffer;
10223 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10224 for (j=0; j<info.channels; j++) {
10225 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
10228 out += info.outJump;
10232 else if (info.outFormat == RTAUDIO_SINT32) {
10233 Int32 *out = (Int32 *)outBuffer;
10234 if (info.inFormat == RTAUDIO_SINT8) {
10235 signed char *in = (signed char *)inBuffer;
10236 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10237 for (j=0; j<info.channels; j++) {
10238 out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
10239 out[info.outOffset[j]] <<= 24;
10242 out += info.outJump;
10245 else if (info.inFormat == RTAUDIO_SINT16) {
10246 Int16 *in = (Int16 *)inBuffer;
10247 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10248 for (j=0; j<info.channels; j++) {
10249 out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
10250 out[info.outOffset[j]] <<= 16;
10253 out += info.outJump;
10256 else if (info.inFormat == RTAUDIO_SINT24) {
10257 Int24 *in = (Int24 *)inBuffer;
10258 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10259 for (j=0; j<info.channels; j++) {
10260 out[info.outOffset[j]] = (Int32) in[info.inOffset[j]].asInt();
10261 out[info.outOffset[j]] <<= 8;
10264 out += info.outJump;
10267 else if (info.inFormat == RTAUDIO_SINT32) {
10268 // Channel compensation and/or (de)interleaving only.
10269 Int32 *in = (Int32 *)inBuffer;
10270 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10271 for (j=0; j<info.channels; j++) {
10272 out[info.outOffset[j]] = in[info.inOffset[j]];
10275 out += info.outJump;
10278 else if (info.inFormat == RTAUDIO_FLOAT32) {
10279 Float32 *in = (Float32 *)inBuffer;
10280 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10281 for (j=0; j<info.channels; j++) {
10282 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
10285 out += info.outJump;
10288 else if (info.inFormat == RTAUDIO_FLOAT64) {
10289 Float64 *in = (Float64 *)inBuffer;
10290 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10291 for (j=0; j<info.channels; j++) {
10292 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
10295 out += info.outJump;
10299 else if (info.outFormat == RTAUDIO_SINT24) {
10300 Int24 *out = (Int24 *)outBuffer;
10301 if (info.inFormat == RTAUDIO_SINT8) {
10302 signed char *in = (signed char *)inBuffer;
10303 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10304 for (j=0; j<info.channels; j++) {
10305 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] << 16);
10306 //out[info.outOffset[j]] <<= 16;
10309 out += info.outJump;
10312 else if (info.inFormat == RTAUDIO_SINT16) {
10313 Int16 *in = (Int16 *)inBuffer;
10314 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10315 for (j=0; j<info.channels; j++) {
10316 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] << 8);
10317 //out[info.outOffset[j]] <<= 8;
10320 out += info.outJump;
10323 else if (info.inFormat == RTAUDIO_SINT24) {
10324 // Channel compensation and/or (de)interleaving only.
10325 Int24 *in = (Int24 *)inBuffer;
10326 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10327 for (j=0; j<info.channels; j++) {
10328 out[info.outOffset[j]] = in[info.inOffset[j]];
10331 out += info.outJump;
10334 else if (info.inFormat == RTAUDIO_SINT32) {
10335 Int32 *in = (Int32 *)inBuffer;
10336 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10337 for (j=0; j<info.channels; j++) {
10338 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] >> 8);
10339 //out[info.outOffset[j]] >>= 8;
10342 out += info.outJump;
10345 else if (info.inFormat == RTAUDIO_FLOAT32) {
10346 Float32 *in = (Float32 *)inBuffer;
10347 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10348 for (j=0; j<info.channels; j++) {
10349 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
10352 out += info.outJump;
10355 else if (info.inFormat == RTAUDIO_FLOAT64) {
10356 Float64 *in = (Float64 *)inBuffer;
10357 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10358 for (j=0; j<info.channels; j++) {
10359 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
10362 out += info.outJump;
10366 else if (info.outFormat == RTAUDIO_SINT16) {
10367 Int16 *out = (Int16 *)outBuffer;
10368 if (info.inFormat == RTAUDIO_SINT8) {
10369 signed char *in = (signed char *)inBuffer;
10370 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10371 for (j=0; j<info.channels; j++) {
10372 out[info.outOffset[j]] = (Int16) in[info.inOffset[j]];
10373 out[info.outOffset[j]] <<= 8;
10376 out += info.outJump;
10379 else if (info.inFormat == RTAUDIO_SINT16) {
10380 // Channel compensation and/or (de)interleaving only.
10381 Int16 *in = (Int16 *)inBuffer;
10382 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10383 for (j=0; j<info.channels; j++) {
10384 out[info.outOffset[j]] = in[info.inOffset[j]];
10387 out += info.outJump;
10390 else if (info.inFormat == RTAUDIO_SINT24) {
10391 Int24 *in = (Int24 *)inBuffer;
10392 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10393 for (j=0; j<info.channels; j++) {
10394 out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]].asInt() >> 8);
10397 out += info.outJump;
10400 else if (info.inFormat == RTAUDIO_SINT32) {
10401 Int32 *in = (Int32 *)inBuffer;
10402 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10403 for (j=0; j<info.channels; j++) {
10404 out[info.outOffset[j]] = (Int16) ((in[info.inOffset[j]] >> 16) & 0x0000ffff);
10407 out += info.outJump;
10410 else if (info.inFormat == RTAUDIO_FLOAT32) {
10411 Float32 *in = (Float32 *)inBuffer;
10412 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10413 for (j=0; j<info.channels; j++) {
10414 out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
10417 out += info.outJump;
10420 else if (info.inFormat == RTAUDIO_FLOAT64) {
10421 Float64 *in = (Float64 *)inBuffer;
10422 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10423 for (j=0; j<info.channels; j++) {
10424 out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
10427 out += info.outJump;
10431 else if (info.outFormat == RTAUDIO_SINT8) {
10432 signed char *out = (signed char *)outBuffer;
10433 if (info.inFormat == RTAUDIO_SINT8) {
10434 // Channel compensation and/or (de)interleaving only.
10435 signed char *in = (signed char *)inBuffer;
10436 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10437 for (j=0; j<info.channels; j++) {
10438 out[info.outOffset[j]] = in[info.inOffset[j]];
10441 out += info.outJump;
10444 if (info.inFormat == RTAUDIO_SINT16) {
10445 Int16 *in = (Int16 *)inBuffer;
10446 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10447 for (j=0; j<info.channels; j++) {
10448 out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 8) & 0x00ff);
10451 out += info.outJump;
10454 else if (info.inFormat == RTAUDIO_SINT24) {
10455 Int24 *in = (Int24 *)inBuffer;
10456 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10457 for (j=0; j<info.channels; j++) {
10458 out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]].asInt() >> 16);
10461 out += info.outJump;
10464 else if (info.inFormat == RTAUDIO_SINT32) {
10465 Int32 *in = (Int32 *)inBuffer;
10466 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10467 for (j=0; j<info.channels; j++) {
10468 out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 24) & 0x000000ff);
10471 out += info.outJump;
10474 else if (info.inFormat == RTAUDIO_FLOAT32) {
10475 Float32 *in = (Float32 *)inBuffer;
10476 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10477 for (j=0; j<info.channels; j++) {
10478 out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
10481 out += info.outJump;
10484 else if (info.inFormat == RTAUDIO_FLOAT64) {
10485 Float64 *in = (Float64 *)inBuffer;
10486 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10487 for (j=0; j<info.channels; j++) {
10488 out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
10491 out += info.outJump;
10497 //static inline uint16_t bswap_16(uint16_t x) { return (x>>8) | (x<<8); }
10498 //static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); }
10499 //static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); }
10501 void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format )
10507 if ( format == RTAUDIO_SINT16 ) {
10508 for ( unsigned int i=0; i<samples; i++ ) {
10509 // Swap 1st and 2nd bytes.
10514 // Increment 2 bytes.
10518 else if ( format == RTAUDIO_SINT32 ||
10519 format == RTAUDIO_FLOAT32 ) {
10520 for ( unsigned int i=0; i<samples; i++ ) {
10521 // Swap 1st and 4th bytes.
10526 // Swap 2nd and 3rd bytes.
10532 // Increment 3 more bytes.
10536 else if ( format == RTAUDIO_SINT24 ) {
10537 for ( unsigned int i=0; i<samples; i++ ) {
10538 // Swap 1st and 3rd bytes.
10543 // Increment 2 more bytes.
10547 else if ( format == RTAUDIO_FLOAT64 ) {
10548 for ( unsigned int i=0; i<samples; i++ ) {
10549 // Swap 1st and 8th bytes
10554 // Swap 2nd and 7th bytes
10560 // Swap 3rd and 6th bytes
10566 // Swap 4th and 5th bytes
10572 // Increment 5 more bytes.
10578 // Indentation settings for Vim and Emacs
10580 // Local Variables:
10581 // c-basic-offset: 2
10582 // indent-tabs-mode: nil
10585 // vim: et sts=2 sw=2