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 void RtAudio :: getCompiledApi( std::vector<RtAudio::Api> &apis )
105 // The order here will control the order of RtAudio's API search in
107 #if defined(__UNIX_JACK__)
108 apis.push_back( UNIX_JACK );
110 #if defined(__LINUX_PULSE__)
111 apis.push_back( LINUX_PULSE );
113 #if defined(__LINUX_ALSA__)
114 apis.push_back( LINUX_ALSA );
116 #if defined(__LINUX_OSS__)
117 apis.push_back( LINUX_OSS );
119 #if defined(__WINDOWS_ASIO__)
120 apis.push_back( WINDOWS_ASIO );
122 #if defined(__WINDOWS_WASAPI__)
123 apis.push_back( WINDOWS_WASAPI );
125 #if defined(__WINDOWS_DS__)
126 apis.push_back( WINDOWS_DS );
128 #if defined(__MACOSX_CORE__)
129 apis.push_back( MACOSX_CORE );
131 #if defined(__RTAUDIO_DUMMY__)
132 apis.push_back( RTAUDIO_DUMMY );
136 void RtAudio :: openRtApi( RtAudio::Api api )
142 #if defined(__UNIX_JACK__)
143 if ( api == UNIX_JACK )
144 rtapi_ = new RtApiJack();
146 #if defined(__LINUX_ALSA__)
147 if ( api == LINUX_ALSA )
148 rtapi_ = new RtApiAlsa();
150 #if defined(__LINUX_PULSE__)
151 if ( api == LINUX_PULSE )
152 rtapi_ = new RtApiPulse();
154 #if defined(__LINUX_OSS__)
155 if ( api == LINUX_OSS )
156 rtapi_ = new RtApiOss();
158 #if defined(__WINDOWS_ASIO__)
159 if ( api == WINDOWS_ASIO )
160 rtapi_ = new RtApiAsio();
162 #if defined(__WINDOWS_WASAPI__)
163 if ( api == WINDOWS_WASAPI )
164 rtapi_ = new RtApiWasapi();
166 #if defined(__WINDOWS_DS__)
167 if ( api == WINDOWS_DS )
168 rtapi_ = new RtApiDs();
170 #if defined(__MACOSX_CORE__)
171 if ( api == MACOSX_CORE )
172 rtapi_ = new RtApiCore();
174 #if defined(__RTAUDIO_DUMMY__)
175 if ( api == RTAUDIO_DUMMY )
176 rtapi_ = new RtApiDummy();
180 RtAudio :: RtAudio( RtAudio::Api api )
184 if ( api != UNSPECIFIED ) {
185 // Attempt to open the specified API.
187 if ( rtapi_ ) return;
189 // No compiled support for specified API value. Issue a debug
190 // warning and continue as if no API was specified.
191 std::cerr << "\nRtAudio: no compiled support for specified API argument!\n" << std::endl;
194 // Iterate through the compiled APIs and return as soon as we find
195 // one with at least one device or we reach the end of the list.
196 std::vector< RtAudio::Api > apis;
197 getCompiledApi( apis );
198 for ( unsigned int i=0; i<apis.size(); i++ ) {
199 openRtApi( apis[i] );
200 if ( rtapi_ && rtapi_->getDeviceCount() ) break;
203 if ( rtapi_ ) return;
205 // It should not be possible to get here because the preprocessor
206 // definition __RTAUDIO_DUMMY__ is automatically defined if no
207 // API-specific definitions are passed to the compiler. But just in
208 // case something weird happens, we'll thow an error.
209 std::string errorText = "\nRtAudio: no compiled API support found ... critical error!!\n\n";
210 throw( RtAudioError( errorText, RtAudioError::UNSPECIFIED ) );
213 RtAudio :: ~RtAudio()
219 void RtAudio :: openStream( RtAudio::StreamParameters *outputParameters,
220 RtAudio::StreamParameters *inputParameters,
221 RtAudioFormat format, unsigned int sampleRate,
222 unsigned int *bufferFrames,
223 RtAudioCallback callback, void *userData,
224 RtAudio::StreamOptions *options,
225 RtAudioErrorCallback errorCallback )
227 return rtapi_->openStream( outputParameters, inputParameters, format,
228 sampleRate, bufferFrames, callback,
229 userData, options, errorCallback );
232 // *************************************************** //
234 // Public RtApi definitions (see end of file for
235 // private or protected utility functions).
237 // *************************************************** //
241 stream_.state = STREAM_CLOSED;
242 stream_.mode = UNINITIALIZED;
243 stream_.apiHandle = 0;
244 stream_.userBuffer[0] = 0;
245 stream_.userBuffer[1] = 0;
246 MUTEX_INITIALIZE( &stream_.mutex );
247 showWarnings_ = true;
248 firstErrorOccurred_ = false;
253 MUTEX_DESTROY( &stream_.mutex );
256 void RtApi :: openStream( RtAudio::StreamParameters *oParams,
257 RtAudio::StreamParameters *iParams,
258 RtAudioFormat format, unsigned int sampleRate,
259 unsigned int *bufferFrames,
260 RtAudioCallback callback, void *userData,
261 RtAudio::StreamOptions *options,
262 RtAudioErrorCallback errorCallback )
264 if ( stream_.state != STREAM_CLOSED ) {
265 errorText_ = "RtApi::openStream: a stream is already open!";
266 error( RtAudioError::INVALID_USE );
270 // Clear stream information potentially left from a previously open stream.
273 if ( oParams && oParams->nChannels < 1 ) {
274 errorText_ = "RtApi::openStream: a non-NULL output StreamParameters structure cannot have an nChannels value less than one.";
275 error( RtAudioError::INVALID_USE );
279 if ( iParams && iParams->nChannels < 1 ) {
280 errorText_ = "RtApi::openStream: a non-NULL input StreamParameters structure cannot have an nChannels value less than one.";
281 error( RtAudioError::INVALID_USE );
285 if ( oParams == NULL && iParams == NULL ) {
286 errorText_ = "RtApi::openStream: input and output StreamParameters structures are both NULL!";
287 error( RtAudioError::INVALID_USE );
291 if ( formatBytes(format) == 0 ) {
292 errorText_ = "RtApi::openStream: 'format' parameter value is undefined.";
293 error( RtAudioError::INVALID_USE );
297 unsigned int nDevices = getDeviceCount();
298 unsigned int oChannels = 0;
300 oChannels = oParams->nChannels;
301 if ( oParams->deviceId >= nDevices ) {
302 errorText_ = "RtApi::openStream: output device parameter value is invalid.";
303 error( RtAudioError::INVALID_USE );
308 unsigned int iChannels = 0;
310 iChannels = iParams->nChannels;
311 if ( iParams->deviceId >= nDevices ) {
312 errorText_ = "RtApi::openStream: input device parameter value is invalid.";
313 error( RtAudioError::INVALID_USE );
320 if ( oChannels > 0 ) {
322 result = probeDeviceOpen( oParams->deviceId, OUTPUT, oChannels, oParams->firstChannel,
323 sampleRate, format, bufferFrames, options );
324 if ( result == false ) {
325 error( RtAudioError::SYSTEM_ERROR );
330 if ( iChannels > 0 ) {
332 result = probeDeviceOpen( iParams->deviceId, INPUT, iChannels, iParams->firstChannel,
333 sampleRate, format, bufferFrames, options );
334 if ( result == false ) {
335 if ( oChannels > 0 ) closeStream();
336 error( RtAudioError::SYSTEM_ERROR );
341 stream_.callbackInfo.callback = (void *) callback;
342 stream_.callbackInfo.userData = userData;
343 stream_.callbackInfo.errorCallback = (void *) errorCallback;
345 if ( options ) options->numberOfBuffers = stream_.nBuffers;
346 stream_.state = STREAM_STOPPED;
349 unsigned int RtApi :: getDefaultInputDevice( void )
351 // Should be implemented in subclasses if possible.
355 unsigned int RtApi :: getDefaultOutputDevice( void )
357 // Should be implemented in subclasses if possible.
361 void RtApi :: closeStream( void )
363 // MUST be implemented in subclasses!
367 bool RtApi :: probeDeviceOpen( unsigned int /*device*/, StreamMode /*mode*/, unsigned int /*channels*/,
368 unsigned int /*firstChannel*/, unsigned int /*sampleRate*/,
369 RtAudioFormat /*format*/, unsigned int * /*bufferSize*/,
370 RtAudio::StreamOptions * /*options*/ )
372 // MUST be implemented in subclasses!
376 void RtApi :: tickStreamTime( void )
378 // Subclasses that do not provide their own implementation of
379 // getStreamTime should call this function once per buffer I/O to
380 // provide basic stream time support.
382 stream_.streamTime += ( stream_.bufferSize * 1.0 / stream_.sampleRate );
384 #if defined( HAVE_GETTIMEOFDAY )
385 gettimeofday( &stream_.lastTickTimestamp, NULL );
389 long RtApi :: getStreamLatency( void )
393 long totalLatency = 0;
394 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
395 totalLatency = stream_.latency[0];
396 if ( stream_.mode == INPUT || stream_.mode == DUPLEX )
397 totalLatency += stream_.latency[1];
402 double RtApi :: getStreamTime( void )
406 #if defined( HAVE_GETTIMEOFDAY )
407 // Return a very accurate estimate of the stream time by
408 // adding in the elapsed time since the last tick.
412 if ( stream_.state != STREAM_RUNNING || stream_.streamTime == 0.0 )
413 return stream_.streamTime;
415 gettimeofday( &now, NULL );
416 then = stream_.lastTickTimestamp;
417 return stream_.streamTime +
418 ((now.tv_sec + 0.000001 * now.tv_usec) -
419 (then.tv_sec + 0.000001 * then.tv_usec));
421 return stream_.streamTime;
425 void RtApi :: setStreamTime( double time )
430 stream_.streamTime = time;
431 #if defined( HAVE_GETTIMEOFDAY )
432 gettimeofday( &stream_.lastTickTimestamp, NULL );
436 unsigned int RtApi :: getStreamSampleRate( void )
440 return stream_.sampleRate;
444 // *************************************************** //
446 // OS/API-specific methods.
448 // *************************************************** //
450 #if defined(__MACOSX_CORE__)
452 // The OS X CoreAudio API is designed to use a separate callback
453 // procedure for each of its audio devices. A single RtAudio duplex
454 // stream using two different devices is supported here, though it
455 // cannot be guaranteed to always behave correctly because we cannot
456 // synchronize these two callbacks.
458 // A property listener is installed for over/underrun information.
459 // However, no functionality is currently provided to allow property
460 // listeners to trigger user handlers because it is unclear what could
461 // be done if a critical stream parameter (buffer size, sample rate,
462 // device disconnect) notification arrived. The listeners entail
463 // quite a bit of extra code and most likely, a user program wouldn't
464 // be prepared for the result anyway. However, we do provide a flag
465 // to the client callback function to inform of an over/underrun.
467 // A structure to hold various information related to the CoreAudio API
470 AudioDeviceID id[2]; // device ids
471 #if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
472 AudioDeviceIOProcID procId[2];
474 UInt32 iStream[2]; // device stream index (or first if using multiple)
475 UInt32 nStreams[2]; // number of streams to use
478 pthread_cond_t condition;
479 int drainCounter; // Tracks callback counts when draining
480 bool internalDrain; // Indicates if stop is initiated from callback or not.
483 :deviceBuffer(0), drainCounter(0), internalDrain(false) { nStreams[0] = 1; nStreams[1] = 1; id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
486 RtApiCore:: RtApiCore()
488 #if defined( AVAILABLE_MAC_OS_X_VERSION_10_6_AND_LATER )
489 // This is a largely undocumented but absolutely necessary
490 // requirement starting with OS-X 10.6. If not called, queries and
491 // updates to various audio device properties are not handled
493 CFRunLoopRef theRunLoop = NULL;
494 AudioObjectPropertyAddress property = { kAudioHardwarePropertyRunLoop,
495 kAudioObjectPropertyScopeGlobal,
496 kAudioObjectPropertyElementMaster };
497 OSStatus result = AudioObjectSetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, sizeof(CFRunLoopRef), &theRunLoop);
498 if ( result != noErr ) {
499 errorText_ = "RtApiCore::RtApiCore: error setting run loop property!";
500 error( RtAudioError::WARNING );
505 RtApiCore :: ~RtApiCore()
507 // The subclass destructor gets called before the base class
508 // destructor, so close an existing stream before deallocating
509 // apiDeviceId memory.
510 if ( stream_.state != STREAM_CLOSED ) closeStream();
513 unsigned int RtApiCore :: getDeviceCount( void )
515 // Find out how many audio devices there are, if any.
517 AudioObjectPropertyAddress propertyAddress = { kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
518 OSStatus result = AudioObjectGetPropertyDataSize( kAudioObjectSystemObject, &propertyAddress, 0, NULL, &dataSize );
519 if ( result != noErr ) {
520 errorText_ = "RtApiCore::getDeviceCount: OS-X error getting device info!";
521 error( RtAudioError::WARNING );
525 return dataSize / sizeof( AudioDeviceID );
528 unsigned int RtApiCore :: getDefaultInputDevice( void )
530 unsigned int nDevices = getDeviceCount();
531 if ( nDevices <= 1 ) return 0;
534 UInt32 dataSize = sizeof( AudioDeviceID );
535 AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultInputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
536 OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id );
537 if ( result != noErr ) {
538 errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device.";
539 error( RtAudioError::WARNING );
543 dataSize *= nDevices;
544 AudioDeviceID deviceList[ nDevices ];
545 property.mSelector = kAudioHardwarePropertyDevices;
546 result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList );
547 if ( result != noErr ) {
548 errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device IDs.";
549 error( RtAudioError::WARNING );
553 for ( unsigned int i=0; i<nDevices; i++ )
554 if ( id == deviceList[i] ) return i;
556 errorText_ = "RtApiCore::getDefaultInputDevice: No default device found!";
557 error( RtAudioError::WARNING );
561 unsigned int RtApiCore :: getDefaultOutputDevice( void )
563 unsigned int nDevices = getDeviceCount();
564 if ( nDevices <= 1 ) return 0;
567 UInt32 dataSize = sizeof( AudioDeviceID );
568 AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultOutputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
569 OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id );
570 if ( result != noErr ) {
571 errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device.";
572 error( RtAudioError::WARNING );
576 dataSize = sizeof( AudioDeviceID ) * nDevices;
577 AudioDeviceID deviceList[ nDevices ];
578 property.mSelector = kAudioHardwarePropertyDevices;
579 result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList );
580 if ( result != noErr ) {
581 errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device IDs.";
582 error( RtAudioError::WARNING );
586 for ( unsigned int i=0; i<nDevices; i++ )
587 if ( id == deviceList[i] ) return i;
589 errorText_ = "RtApiCore::getDefaultOutputDevice: No default device found!";
590 error( RtAudioError::WARNING );
594 RtAudio::DeviceInfo RtApiCore :: getDeviceInfo( unsigned int device )
596 RtAudio::DeviceInfo info;
600 unsigned int nDevices = getDeviceCount();
601 if ( nDevices == 0 ) {
602 errorText_ = "RtApiCore::getDeviceInfo: no devices found!";
603 error( RtAudioError::INVALID_USE );
607 if ( device >= nDevices ) {
608 errorText_ = "RtApiCore::getDeviceInfo: device ID is invalid!";
609 error( RtAudioError::INVALID_USE );
613 AudioDeviceID deviceList[ nDevices ];
614 UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;
615 AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
616 kAudioObjectPropertyScopeGlobal,
617 kAudioObjectPropertyElementMaster };
618 OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property,
619 0, NULL, &dataSize, (void *) &deviceList );
620 if ( result != noErr ) {
621 errorText_ = "RtApiCore::getDeviceInfo: OS-X system error getting device IDs.";
622 error( RtAudioError::WARNING );
626 AudioDeviceID id = deviceList[ device ];
628 // Get the device name.
631 dataSize = sizeof( CFStringRef );
632 property.mSelector = kAudioObjectPropertyManufacturer;
633 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname );
634 if ( result != noErr ) {
635 errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device manufacturer.";
636 errorText_ = errorStream_.str();
637 error( RtAudioError::WARNING );
641 //const char *mname = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() );
642 int length = CFStringGetLength(cfname);
643 char *mname = (char *)malloc(length * 3 + 1);
644 #if defined( UNICODE ) || defined( _UNICODE )
645 CFStringGetCString(cfname, mname, length * 3 + 1, kCFStringEncodingUTF8);
647 CFStringGetCString(cfname, mname, length * 3 + 1, CFStringGetSystemEncoding());
649 info.name.append( (const char *)mname, strlen(mname) );
650 info.name.append( ": " );
654 property.mSelector = kAudioObjectPropertyName;
655 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname );
656 if ( result != noErr ) {
657 errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device name.";
658 errorText_ = errorStream_.str();
659 error( RtAudioError::WARNING );
663 //const char *name = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() );
664 length = CFStringGetLength(cfname);
665 char *name = (char *)malloc(length * 3 + 1);
666 #if defined( UNICODE ) || defined( _UNICODE )
667 CFStringGetCString(cfname, name, length * 3 + 1, kCFStringEncodingUTF8);
669 CFStringGetCString(cfname, name, length * 3 + 1, CFStringGetSystemEncoding());
671 info.name.append( (const char *)name, strlen(name) );
675 // Get the output stream "configuration".
676 AudioBufferList *bufferList = nil;
677 property.mSelector = kAudioDevicePropertyStreamConfiguration;
678 property.mScope = kAudioDevicePropertyScopeOutput;
679 // property.mElement = kAudioObjectPropertyElementWildcard;
681 result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
682 if ( result != noErr || dataSize == 0 ) {
683 errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration info for device (" << device << ").";
684 errorText_ = errorStream_.str();
685 error( RtAudioError::WARNING );
689 // Allocate the AudioBufferList.
690 bufferList = (AudioBufferList *) malloc( dataSize );
691 if ( bufferList == NULL ) {
692 errorText_ = "RtApiCore::getDeviceInfo: memory error allocating output AudioBufferList.";
693 error( RtAudioError::WARNING );
697 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
698 if ( result != noErr || dataSize == 0 ) {
700 errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration for device (" << device << ").";
701 errorText_ = errorStream_.str();
702 error( RtAudioError::WARNING );
706 // Get output channel information.
707 unsigned int i, nStreams = bufferList->mNumberBuffers;
708 for ( i=0; i<nStreams; i++ )
709 info.outputChannels += bufferList->mBuffers[i].mNumberChannels;
712 // Get the input stream "configuration".
713 property.mScope = kAudioDevicePropertyScopeInput;
714 result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
715 if ( result != noErr || dataSize == 0 ) {
716 errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration info for device (" << device << ").";
717 errorText_ = errorStream_.str();
718 error( RtAudioError::WARNING );
722 // Allocate the AudioBufferList.
723 bufferList = (AudioBufferList *) malloc( dataSize );
724 if ( bufferList == NULL ) {
725 errorText_ = "RtApiCore::getDeviceInfo: memory error allocating input AudioBufferList.";
726 error( RtAudioError::WARNING );
730 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
731 if (result != noErr || dataSize == 0) {
733 errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration for device (" << device << ").";
734 errorText_ = errorStream_.str();
735 error( RtAudioError::WARNING );
739 // Get input channel information.
740 nStreams = bufferList->mNumberBuffers;
741 for ( i=0; i<nStreams; i++ )
742 info.inputChannels += bufferList->mBuffers[i].mNumberChannels;
745 // If device opens for both playback and capture, we determine the channels.
746 if ( info.outputChannels > 0 && info.inputChannels > 0 )
747 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
749 // Probe the device sample rates.
750 bool isInput = false;
751 if ( info.outputChannels == 0 ) isInput = true;
753 // Determine the supported sample rates.
754 property.mSelector = kAudioDevicePropertyAvailableNominalSampleRates;
755 if ( isInput == false ) property.mScope = kAudioDevicePropertyScopeOutput;
756 result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
757 if ( result != kAudioHardwareNoError || dataSize == 0 ) {
758 errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rate info.";
759 errorText_ = errorStream_.str();
760 error( RtAudioError::WARNING );
764 UInt32 nRanges = dataSize / sizeof( AudioValueRange );
765 AudioValueRange rangeList[ nRanges ];
766 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &rangeList );
767 if ( result != kAudioHardwareNoError ) {
768 errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rates.";
769 errorText_ = errorStream_.str();
770 error( RtAudioError::WARNING );
774 // The sample rate reporting mechanism is a bit of a mystery. It
775 // seems that it can either return individual rates or a range of
776 // rates. I assume that if the min / max range values are the same,
777 // then that represents a single supported rate and if the min / max
778 // range values are different, the device supports an arbitrary
779 // range of values (though there might be multiple ranges, so we'll
780 // use the most conservative range).
781 Float64 minimumRate = 1.0, maximumRate = 10000000000.0;
782 bool haveValueRange = false;
783 info.sampleRates.clear();
784 for ( UInt32 i=0; i<nRanges; i++ ) {
785 if ( rangeList[i].mMinimum == rangeList[i].mMaximum ) {
786 unsigned int tmpSr = (unsigned int) rangeList[i].mMinimum;
787 info.sampleRates.push_back( tmpSr );
789 if ( !info.preferredSampleRate || ( tmpSr <= 48000 && tmpSr > info.preferredSampleRate ) )
790 info.preferredSampleRate = tmpSr;
793 haveValueRange = true;
794 if ( rangeList[i].mMinimum > minimumRate ) minimumRate = rangeList[i].mMinimum;
795 if ( rangeList[i].mMaximum < maximumRate ) maximumRate = rangeList[i].mMaximum;
799 if ( haveValueRange ) {
800 for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
801 if ( SAMPLE_RATES[k] >= (unsigned int) minimumRate && SAMPLE_RATES[k] <= (unsigned int) maximumRate ) {
802 info.sampleRates.push_back( SAMPLE_RATES[k] );
804 if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
805 info.preferredSampleRate = SAMPLE_RATES[k];
810 // Sort and remove any redundant values
811 std::sort( info.sampleRates.begin(), info.sampleRates.end() );
812 info.sampleRates.erase( unique( info.sampleRates.begin(), info.sampleRates.end() ), info.sampleRates.end() );
814 if ( info.sampleRates.size() == 0 ) {
815 errorStream_ << "RtApiCore::probeDeviceInfo: No supported sample rates found for device (" << device << ").";
816 errorText_ = errorStream_.str();
817 error( RtAudioError::WARNING );
821 // CoreAudio always uses 32-bit floating point data for PCM streams.
822 // Thus, any other "physical" formats supported by the device are of
823 // no interest to the client.
824 info.nativeFormats = RTAUDIO_FLOAT32;
826 if ( info.outputChannels > 0 )
827 if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true;
828 if ( info.inputChannels > 0 )
829 if ( getDefaultInputDevice() == device ) info.isDefaultInput = true;
835 static OSStatus callbackHandler( AudioDeviceID inDevice,
836 const AudioTimeStamp* /*inNow*/,
837 const AudioBufferList* inInputData,
838 const AudioTimeStamp* /*inInputTime*/,
839 AudioBufferList* outOutputData,
840 const AudioTimeStamp* /*inOutputTime*/,
843 CallbackInfo *info = (CallbackInfo *) infoPointer;
845 RtApiCore *object = (RtApiCore *) info->object;
846 if ( object->callbackEvent( inDevice, inInputData, outOutputData ) == false )
847 return kAudioHardwareUnspecifiedError;
849 return kAudioHardwareNoError;
852 static OSStatus xrunListener( AudioObjectID /*inDevice*/,
854 const AudioObjectPropertyAddress properties[],
855 void* handlePointer )
857 CoreHandle *handle = (CoreHandle *) handlePointer;
858 for ( UInt32 i=0; i<nAddresses; i++ ) {
859 if ( properties[i].mSelector == kAudioDeviceProcessorOverload ) {
860 if ( properties[i].mScope == kAudioDevicePropertyScopeInput )
861 handle->xrun[1] = true;
863 handle->xrun[0] = true;
867 return kAudioHardwareNoError;
870 static OSStatus rateListener( AudioObjectID inDevice,
871 UInt32 /*nAddresses*/,
872 const AudioObjectPropertyAddress /*properties*/[],
875 Float64 *rate = (Float64 *) ratePointer;
876 UInt32 dataSize = sizeof( Float64 );
877 AudioObjectPropertyAddress property = { kAudioDevicePropertyNominalSampleRate,
878 kAudioObjectPropertyScopeGlobal,
879 kAudioObjectPropertyElementMaster };
880 AudioObjectGetPropertyData( inDevice, &property, 0, NULL, &dataSize, rate );
881 return kAudioHardwareNoError;
884 bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
885 unsigned int firstChannel, unsigned int sampleRate,
886 RtAudioFormat format, unsigned int *bufferSize,
887 RtAudio::StreamOptions *options )
890 unsigned int nDevices = getDeviceCount();
891 if ( nDevices == 0 ) {
892 // This should not happen because a check is made before this function is called.
893 errorText_ = "RtApiCore::probeDeviceOpen: no devices found!";
897 if ( device >= nDevices ) {
898 // This should not happen because a check is made before this function is called.
899 errorText_ = "RtApiCore::probeDeviceOpen: device ID is invalid!";
903 AudioDeviceID deviceList[ nDevices ];
904 UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;
905 AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
906 kAudioObjectPropertyScopeGlobal,
907 kAudioObjectPropertyElementMaster };
908 OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property,
909 0, NULL, &dataSize, (void *) &deviceList );
910 if ( result != noErr ) {
911 errorText_ = "RtApiCore::probeDeviceOpen: OS-X system error getting device IDs.";
915 AudioDeviceID id = deviceList[ device ];
917 // Setup for stream mode.
918 bool isInput = false;
919 if ( mode == INPUT ) {
921 property.mScope = kAudioDevicePropertyScopeInput;
924 property.mScope = kAudioDevicePropertyScopeOutput;
926 // Get the stream "configuration".
927 AudioBufferList *bufferList = nil;
929 property.mSelector = kAudioDevicePropertyStreamConfiguration;
930 result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
931 if ( result != noErr || dataSize == 0 ) {
932 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration info for device (" << device << ").";
933 errorText_ = errorStream_.str();
937 // Allocate the AudioBufferList.
938 bufferList = (AudioBufferList *) malloc( dataSize );
939 if ( bufferList == NULL ) {
940 errorText_ = "RtApiCore::probeDeviceOpen: memory error allocating AudioBufferList.";
944 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
945 if (result != noErr || dataSize == 0) {
947 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration for device (" << device << ").";
948 errorText_ = errorStream_.str();
952 // Search for one or more streams that contain the desired number of
953 // channels. CoreAudio devices can have an arbitrary number of
954 // streams and each stream can have an arbitrary number of channels.
955 // For each stream, a single buffer of interleaved samples is
956 // provided. RtAudio prefers the use of one stream of interleaved
957 // data or multiple consecutive single-channel streams. However, we
958 // now support multiple consecutive multi-channel streams of
959 // interleaved data as well.
960 UInt32 iStream, offsetCounter = firstChannel;
961 UInt32 nStreams = bufferList->mNumberBuffers;
962 bool monoMode = false;
963 bool foundStream = false;
965 // First check that the device supports the requested number of
967 UInt32 deviceChannels = 0;
968 for ( iStream=0; iStream<nStreams; iStream++ )
969 deviceChannels += bufferList->mBuffers[iStream].mNumberChannels;
971 if ( deviceChannels < ( channels + firstChannel ) ) {
973 errorStream_ << "RtApiCore::probeDeviceOpen: the device (" << device << ") does not support the requested channel count.";
974 errorText_ = errorStream_.str();
978 // Look for a single stream meeting our needs.
979 UInt32 firstStream, streamCount = 1, streamChannels = 0, channelOffset = 0;
980 for ( iStream=0; iStream<nStreams; iStream++ ) {
981 streamChannels = bufferList->mBuffers[iStream].mNumberChannels;
982 if ( streamChannels >= channels + offsetCounter ) {
983 firstStream = iStream;
984 channelOffset = offsetCounter;
988 if ( streamChannels > offsetCounter ) break;
989 offsetCounter -= streamChannels;
992 // If we didn't find a single stream above, then we should be able
993 // to meet the channel specification with multiple streams.
994 if ( foundStream == false ) {
996 offsetCounter = firstChannel;
997 for ( iStream=0; iStream<nStreams; iStream++ ) {
998 streamChannels = bufferList->mBuffers[iStream].mNumberChannels;
999 if ( streamChannels > offsetCounter ) break;
1000 offsetCounter -= streamChannels;
1003 firstStream = iStream;
1004 channelOffset = offsetCounter;
1005 Int32 channelCounter = channels + offsetCounter - streamChannels;
1007 if ( streamChannels > 1 ) monoMode = false;
1008 while ( channelCounter > 0 ) {
1009 streamChannels = bufferList->mBuffers[++iStream].mNumberChannels;
1010 if ( streamChannels > 1 ) monoMode = false;
1011 channelCounter -= streamChannels;
1018 // Determine the buffer size.
1019 AudioValueRange bufferRange;
1020 dataSize = sizeof( AudioValueRange );
1021 property.mSelector = kAudioDevicePropertyBufferFrameSizeRange;
1022 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &bufferRange );
1024 if ( result != noErr ) {
1025 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting buffer size range for device (" << device << ").";
1026 errorText_ = errorStream_.str();
1030 if ( bufferRange.mMinimum > *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMinimum;
1031 else if ( bufferRange.mMaximum < *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMaximum;
1032 if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) *bufferSize = (unsigned long) bufferRange.mMinimum;
1034 // Set the buffer size. For multiple streams, I'm assuming we only
1035 // need to make this setting for the master channel.
1036 UInt32 theSize = (UInt32) *bufferSize;
1037 dataSize = sizeof( UInt32 );
1038 property.mSelector = kAudioDevicePropertyBufferFrameSize;
1039 result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &theSize );
1041 if ( result != noErr ) {
1042 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting the buffer size for device (" << device << ").";
1043 errorText_ = errorStream_.str();
1047 // If attempting to setup a duplex stream, the bufferSize parameter
1048 // MUST be the same in both directions!
1049 *bufferSize = theSize;
1050 if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {
1051 errorStream_ << "RtApiCore::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << device << ").";
1052 errorText_ = errorStream_.str();
1056 stream_.bufferSize = *bufferSize;
1057 stream_.nBuffers = 1;
1059 // Try to set "hog" mode ... it's not clear to me this is working.
1060 if ( options && options->flags & RTAUDIO_HOG_DEVICE ) {
1062 dataSize = sizeof( hog_pid );
1063 property.mSelector = kAudioDevicePropertyHogMode;
1064 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &hog_pid );
1065 if ( result != noErr ) {
1066 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting 'hog' state!";
1067 errorText_ = errorStream_.str();
1071 if ( hog_pid != getpid() ) {
1073 result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &hog_pid );
1074 if ( result != noErr ) {
1075 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting 'hog' state!";
1076 errorText_ = errorStream_.str();
1082 // Check and if necessary, change the sample rate for the device.
1083 Float64 nominalRate;
1084 dataSize = sizeof( Float64 );
1085 property.mSelector = kAudioDevicePropertyNominalSampleRate;
1086 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &nominalRate );
1087 if ( result != noErr ) {
1088 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting current sample rate.";
1089 errorText_ = errorStream_.str();
1093 // Only change the sample rate if off by more than 1 Hz.
1094 if ( fabs( nominalRate - (double)sampleRate ) > 1.0 ) {
1096 // Set a property listener for the sample rate change
1097 Float64 reportedRate = 0.0;
1098 AudioObjectPropertyAddress tmp = { kAudioDevicePropertyNominalSampleRate, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
1099 result = AudioObjectAddPropertyListener( id, &tmp, rateListener, (void *) &reportedRate );
1100 if ( result != noErr ) {
1101 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate property listener for device (" << device << ").";
1102 errorText_ = errorStream_.str();
1106 nominalRate = (Float64) sampleRate;
1107 result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &nominalRate );
1108 if ( result != noErr ) {
1109 AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate );
1110 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate for device (" << device << ").";
1111 errorText_ = errorStream_.str();
1115 // Now wait until the reported nominal rate is what we just set.
1116 UInt32 microCounter = 0;
1117 while ( reportedRate != nominalRate ) {
1118 microCounter += 5000;
1119 if ( microCounter > 5000000 ) break;
1123 // Remove the property listener.
1124 AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate );
1126 if ( microCounter > 5000000 ) {
1127 errorStream_ << "RtApiCore::probeDeviceOpen: timeout waiting for sample rate update for device (" << device << ").";
1128 errorText_ = errorStream_.str();
1133 // Now set the stream format for all streams. Also, check the
1134 // physical format of the device and change that if necessary.
1135 AudioStreamBasicDescription description;
1136 dataSize = sizeof( AudioStreamBasicDescription );
1137 property.mSelector = kAudioStreamPropertyVirtualFormat;
1138 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &description );
1139 if ( result != noErr ) {
1140 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream format for device (" << device << ").";
1141 errorText_ = errorStream_.str();
1145 // Set the sample rate and data format id. However, only make the
1146 // change if the sample rate is not within 1.0 of the desired
1147 // rate and the format is not linear pcm.
1148 bool updateFormat = false;
1149 if ( fabs( description.mSampleRate - (Float64)sampleRate ) > 1.0 ) {
1150 description.mSampleRate = (Float64) sampleRate;
1151 updateFormat = true;
1154 if ( description.mFormatID != kAudioFormatLinearPCM ) {
1155 description.mFormatID = kAudioFormatLinearPCM;
1156 updateFormat = true;
1159 if ( updateFormat ) {
1160 result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &description );
1161 if ( result != noErr ) {
1162 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate or data format for device (" << device << ").";
1163 errorText_ = errorStream_.str();
1168 // Now check the physical format.
1169 property.mSelector = kAudioStreamPropertyPhysicalFormat;
1170 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &description );
1171 if ( result != noErr ) {
1172 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream physical format for device (" << device << ").";
1173 errorText_ = errorStream_.str();
1177 //std::cout << "Current physical stream format:" << std::endl;
1178 //std::cout << " mBitsPerChan = " << description.mBitsPerChannel << std::endl;
1179 //std::cout << " aligned high = " << (description.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (description.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl;
1180 //std::cout << " bytesPerFrame = " << description.mBytesPerFrame << std::endl;
1181 //std::cout << " sample rate = " << description.mSampleRate << std::endl;
1183 if ( description.mFormatID != kAudioFormatLinearPCM || description.mBitsPerChannel < 16 ) {
1184 description.mFormatID = kAudioFormatLinearPCM;
1185 //description.mSampleRate = (Float64) sampleRate;
1186 AudioStreamBasicDescription testDescription = description;
1189 // We'll try higher bit rates first and then work our way down.
1190 std::vector< std::pair<UInt32, UInt32> > physicalFormats;
1191 formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsFloat) & ~kLinearPCMFormatFlagIsSignedInteger;
1192 physicalFormats.push_back( std::pair<Float32, UInt32>( 32, formatFlags ) );
1193 formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat;
1194 physicalFormats.push_back( std::pair<Float32, UInt32>( 32, formatFlags ) );
1195 physicalFormats.push_back( std::pair<Float32, UInt32>( 24, formatFlags ) ); // 24-bit packed
1196 formatFlags &= ~( kAudioFormatFlagIsPacked | kAudioFormatFlagIsAlignedHigh );
1197 physicalFormats.push_back( std::pair<Float32, UInt32>( 24.2, formatFlags ) ); // 24-bit in 4 bytes, aligned low
1198 formatFlags |= kAudioFormatFlagIsAlignedHigh;
1199 physicalFormats.push_back( std::pair<Float32, UInt32>( 24.4, formatFlags ) ); // 24-bit in 4 bytes, aligned high
1200 formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat;
1201 physicalFormats.push_back( std::pair<Float32, UInt32>( 16, formatFlags ) );
1202 physicalFormats.push_back( std::pair<Float32, UInt32>( 8, formatFlags ) );
1204 bool setPhysicalFormat = false;
1205 for( unsigned int i=0; i<physicalFormats.size(); i++ ) {
1206 testDescription = description;
1207 testDescription.mBitsPerChannel = (UInt32) physicalFormats[i].first;
1208 testDescription.mFormatFlags = physicalFormats[i].second;
1209 if ( (24 == (UInt32)physicalFormats[i].first) && ~( physicalFormats[i].second & kAudioFormatFlagIsPacked ) )
1210 testDescription.mBytesPerFrame = 4 * testDescription.mChannelsPerFrame;
1212 testDescription.mBytesPerFrame = testDescription.mBitsPerChannel/8 * testDescription.mChannelsPerFrame;
1213 testDescription.mBytesPerPacket = testDescription.mBytesPerFrame * testDescription.mFramesPerPacket;
1214 result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &testDescription );
1215 if ( result == noErr ) {
1216 setPhysicalFormat = true;
1217 //std::cout << "Updated physical stream format:" << std::endl;
1218 //std::cout << " mBitsPerChan = " << testDescription.mBitsPerChannel << std::endl;
1219 //std::cout << " aligned high = " << (testDescription.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (testDescription.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl;
1220 //std::cout << " bytesPerFrame = " << testDescription.mBytesPerFrame << std::endl;
1221 //std::cout << " sample rate = " << testDescription.mSampleRate << std::endl;
1226 if ( !setPhysicalFormat ) {
1227 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting physical data format for device (" << device << ").";
1228 errorText_ = errorStream_.str();
1231 } // done setting virtual/physical formats.
1233 // Get the stream / device latency.
1235 dataSize = sizeof( UInt32 );
1236 property.mSelector = kAudioDevicePropertyLatency;
1237 if ( AudioObjectHasProperty( id, &property ) == true ) {
1238 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &latency );
1239 if ( result == kAudioHardwareNoError ) stream_.latency[ mode ] = latency;
1241 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting device latency for device (" << device << ").";
1242 errorText_ = errorStream_.str();
1243 error( RtAudioError::WARNING );
1247 // Byte-swapping: According to AudioHardware.h, the stream data will
1248 // always be presented in native-endian format, so we should never
1249 // need to byte swap.
1250 stream_.doByteSwap[mode] = false;
1252 // From the CoreAudio documentation, PCM data must be supplied as
1254 stream_.userFormat = format;
1255 stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
1257 if ( streamCount == 1 )
1258 stream_.nDeviceChannels[mode] = description.mChannelsPerFrame;
1259 else // multiple streams
1260 stream_.nDeviceChannels[mode] = channels;
1261 stream_.nUserChannels[mode] = channels;
1262 stream_.channelOffset[mode] = channelOffset; // offset within a CoreAudio stream
1263 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
1264 else stream_.userInterleaved = true;
1265 stream_.deviceInterleaved[mode] = true;
1266 if ( monoMode == true ) stream_.deviceInterleaved[mode] = false;
1268 // Set flags for buffer conversion.
1269 stream_.doConvertBuffer[mode] = false;
1270 if ( stream_.userFormat != stream_.deviceFormat[mode] )
1271 stream_.doConvertBuffer[mode] = true;
1272 if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
1273 stream_.doConvertBuffer[mode] = true;
1274 if ( streamCount == 1 ) {
1275 if ( stream_.nUserChannels[mode] > 1 &&
1276 stream_.userInterleaved != stream_.deviceInterleaved[mode] )
1277 stream_.doConvertBuffer[mode] = true;
1279 else if ( monoMode && stream_.userInterleaved )
1280 stream_.doConvertBuffer[mode] = true;
1282 // Allocate our CoreHandle structure for the stream.
1283 CoreHandle *handle = 0;
1284 if ( stream_.apiHandle == 0 ) {
1286 handle = new CoreHandle;
1288 catch ( std::bad_alloc& ) {
1289 errorText_ = "RtApiCore::probeDeviceOpen: error allocating CoreHandle memory.";
1293 if ( pthread_cond_init( &handle->condition, NULL ) ) {
1294 errorText_ = "RtApiCore::probeDeviceOpen: error initializing pthread condition variable.";
1297 stream_.apiHandle = (void *) handle;
1300 handle = (CoreHandle *) stream_.apiHandle;
1301 handle->iStream[mode] = firstStream;
1302 handle->nStreams[mode] = streamCount;
1303 handle->id[mode] = id;
1305 // Allocate necessary internal buffers.
1306 unsigned long bufferBytes;
1307 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
1308 // stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
1309 stream_.userBuffer[mode] = (char *) malloc( bufferBytes * sizeof(char) );
1310 memset( stream_.userBuffer[mode], 0, bufferBytes * sizeof(char) );
1311 if ( stream_.userBuffer[mode] == NULL ) {
1312 errorText_ = "RtApiCore::probeDeviceOpen: error allocating user buffer memory.";
1316 // If possible, we will make use of the CoreAudio stream buffers as
1317 // "device buffers". However, we can't do this if using multiple
1319 if ( stream_.doConvertBuffer[mode] && handle->nStreams[mode] > 1 ) {
1321 bool makeBuffer = true;
1322 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
1323 if ( mode == INPUT ) {
1324 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
1325 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
1326 if ( bufferBytes <= bytesOut ) makeBuffer = false;
1331 bufferBytes *= *bufferSize;
1332 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
1333 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
1334 if ( stream_.deviceBuffer == NULL ) {
1335 errorText_ = "RtApiCore::probeDeviceOpen: error allocating device buffer memory.";
1341 stream_.sampleRate = sampleRate;
1342 stream_.device[mode] = device;
1343 stream_.state = STREAM_STOPPED;
1344 stream_.callbackInfo.object = (void *) this;
1346 // Setup the buffer conversion information structure.
1347 if ( stream_.doConvertBuffer[mode] ) {
1348 if ( streamCount > 1 ) setConvertInfo( mode, 0 );
1349 else setConvertInfo( mode, channelOffset );
1352 if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device )
1353 // Only one callback procedure per device.
1354 stream_.mode = DUPLEX;
1356 #if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
1357 result = AudioDeviceCreateIOProcID( id, callbackHandler, (void *) &stream_.callbackInfo, &handle->procId[mode] );
1359 // deprecated in favor of AudioDeviceCreateIOProcID()
1360 result = AudioDeviceAddIOProc( id, callbackHandler, (void *) &stream_.callbackInfo );
1362 if ( result != noErr ) {
1363 errorStream_ << "RtApiCore::probeDeviceOpen: system error setting callback for device (" << device << ").";
1364 errorText_ = errorStream_.str();
1367 if ( stream_.mode == OUTPUT && mode == INPUT )
1368 stream_.mode = DUPLEX;
1370 stream_.mode = mode;
1373 // Setup the device property listener for over/underload.
1374 property.mSelector = kAudioDeviceProcessorOverload;
1375 property.mScope = kAudioObjectPropertyScopeGlobal;
1376 result = AudioObjectAddPropertyListener( id, &property, xrunListener, (void *) handle );
1382 pthread_cond_destroy( &handle->condition );
1384 stream_.apiHandle = 0;
1387 for ( int i=0; i<2; i++ ) {
1388 if ( stream_.userBuffer[i] ) {
1389 free( stream_.userBuffer[i] );
1390 stream_.userBuffer[i] = 0;
1394 if ( stream_.deviceBuffer ) {
1395 free( stream_.deviceBuffer );
1396 stream_.deviceBuffer = 0;
1399 stream_.state = STREAM_CLOSED;
1403 void RtApiCore :: closeStream( void )
1405 if ( stream_.state == STREAM_CLOSED ) {
1406 errorText_ = "RtApiCore::closeStream(): no open stream to close!";
1407 error( RtAudioError::WARNING );
1411 CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
1412 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
1414 AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
1415 kAudioObjectPropertyScopeGlobal,
1416 kAudioObjectPropertyElementMaster };
1418 property.mSelector = kAudioDeviceProcessorOverload;
1419 property.mScope = kAudioObjectPropertyScopeGlobal;
1420 if (AudioObjectRemovePropertyListener( handle->id[0], &property, xrunListener, (void *) handle ) != noErr) {
1421 errorText_ = "RtApiCore::closeStream(): error removing property listener!";
1422 error( RtAudioError::WARNING );
1425 if ( stream_.state == STREAM_RUNNING )
1426 AudioDeviceStop( handle->id[0], callbackHandler );
1427 #if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
1428 AudioDeviceDestroyIOProcID( handle->id[0], handle->procId[0] );
1430 // deprecated in favor of AudioDeviceDestroyIOProcID()
1431 AudioDeviceRemoveIOProc( handle->id[0], callbackHandler );
1435 if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
1437 AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
1438 kAudioObjectPropertyScopeGlobal,
1439 kAudioObjectPropertyElementMaster };
1441 property.mSelector = kAudioDeviceProcessorOverload;
1442 property.mScope = kAudioObjectPropertyScopeGlobal;
1443 if (AudioObjectRemovePropertyListener( handle->id[1], &property, xrunListener, (void *) handle ) != noErr) {
1444 errorText_ = "RtApiCore::closeStream(): error removing property listener!";
1445 error( RtAudioError::WARNING );
1448 if ( stream_.state == STREAM_RUNNING )
1449 AudioDeviceStop( handle->id[1], callbackHandler );
1450 #if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
1451 AudioDeviceDestroyIOProcID( handle->id[1], handle->procId[1] );
1453 // deprecated in favor of AudioDeviceDestroyIOProcID()
1454 AudioDeviceRemoveIOProc( handle->id[1], callbackHandler );
1458 for ( int i=0; i<2; i++ ) {
1459 if ( stream_.userBuffer[i] ) {
1460 free( stream_.userBuffer[i] );
1461 stream_.userBuffer[i] = 0;
1465 if ( stream_.deviceBuffer ) {
1466 free( stream_.deviceBuffer );
1467 stream_.deviceBuffer = 0;
1470 // Destroy pthread condition variable.
1471 pthread_cond_destroy( &handle->condition );
1473 stream_.apiHandle = 0;
1475 stream_.mode = UNINITIALIZED;
1476 stream_.state = STREAM_CLOSED;
1479 void RtApiCore :: startStream( void )
1482 if ( stream_.state == STREAM_RUNNING ) {
1483 errorText_ = "RtApiCore::startStream(): the stream is already running!";
1484 error( RtAudioError::WARNING );
1488 OSStatus result = noErr;
1489 CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
1490 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
1492 result = AudioDeviceStart( handle->id[0], callbackHandler );
1493 if ( result != noErr ) {
1494 errorStream_ << "RtApiCore::startStream: system error (" << getErrorCode( result ) << ") starting callback procedure on device (" << stream_.device[0] << ").";
1495 errorText_ = errorStream_.str();
1500 if ( stream_.mode == INPUT ||
1501 ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
1503 result = AudioDeviceStart( handle->id[1], callbackHandler );
1504 if ( result != noErr ) {
1505 errorStream_ << "RtApiCore::startStream: system error starting input callback procedure on device (" << stream_.device[1] << ").";
1506 errorText_ = errorStream_.str();
1511 handle->drainCounter = 0;
1512 handle->internalDrain = false;
1513 stream_.state = STREAM_RUNNING;
1516 if ( result == noErr ) return;
1517 error( RtAudioError::SYSTEM_ERROR );
1520 void RtApiCore :: stopStream( void )
1523 if ( stream_.state == STREAM_STOPPED ) {
1524 errorText_ = "RtApiCore::stopStream(): the stream is already stopped!";
1525 error( RtAudioError::WARNING );
1529 OSStatus result = noErr;
1530 CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
1531 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
1533 if ( handle->drainCounter == 0 ) {
1534 handle->drainCounter = 2;
1535 pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
1538 result = AudioDeviceStop( handle->id[0], callbackHandler );
1539 if ( result != noErr ) {
1540 errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping callback procedure on device (" << stream_.device[0] << ").";
1541 errorText_ = errorStream_.str();
1546 if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
1548 result = AudioDeviceStop( handle->id[1], callbackHandler );
1549 if ( result != noErr ) {
1550 errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping input callback procedure on device (" << stream_.device[1] << ").";
1551 errorText_ = errorStream_.str();
1556 stream_.state = STREAM_STOPPED;
1559 if ( result == noErr ) return;
1560 error( RtAudioError::SYSTEM_ERROR );
1563 void RtApiCore :: abortStream( void )
1566 if ( stream_.state == STREAM_STOPPED ) {
1567 errorText_ = "RtApiCore::abortStream(): the stream is already stopped!";
1568 error( RtAudioError::WARNING );
1572 CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
1573 handle->drainCounter = 2;
1578 // This function will be called by a spawned thread when the user
1579 // callback function signals that the stream should be stopped or
1580 // aborted. It is better to handle it this way because the
1581 // callbackEvent() function probably should return before the AudioDeviceStop()
1582 // function is called.
1583 static void *coreStopStream( void *ptr )
1585 CallbackInfo *info = (CallbackInfo *) ptr;
1586 RtApiCore *object = (RtApiCore *) info->object;
1588 object->stopStream();
1589 pthread_exit( NULL );
1592 bool RtApiCore :: callbackEvent( AudioDeviceID deviceId,
1593 const AudioBufferList *inBufferList,
1594 const AudioBufferList *outBufferList )
1596 if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;
1597 if ( stream_.state == STREAM_CLOSED ) {
1598 errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";
1599 error( RtAudioError::WARNING );
1603 CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
1604 CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
1606 // Check if we were draining the stream and signal is finished.
1607 if ( handle->drainCounter > 3 ) {
1608 ThreadHandle threadId;
1610 stream_.state = STREAM_STOPPING;
1611 if ( handle->internalDrain == true )
1612 pthread_create( &threadId, NULL, coreStopStream, info );
1613 else // external call to stopStream()
1614 pthread_cond_signal( &handle->condition );
1618 AudioDeviceID outputDevice = handle->id[0];
1620 // Invoke user callback to get fresh output data UNLESS we are
1621 // draining stream or duplex mode AND the input/output devices are
1622 // different AND this function is called for the input device.
1623 if ( handle->drainCounter == 0 && ( stream_.mode != DUPLEX || deviceId == outputDevice ) ) {
1624 RtAudioCallback callback = (RtAudioCallback) info->callback;
1625 double streamTime = getStreamTime();
1626 RtAudioStreamStatus status = 0;
1627 if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
1628 status |= RTAUDIO_OUTPUT_UNDERFLOW;
1629 handle->xrun[0] = false;
1631 if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
1632 status |= RTAUDIO_INPUT_OVERFLOW;
1633 handle->xrun[1] = false;
1636 int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
1637 stream_.bufferSize, streamTime, status, info->userData );
1638 if ( cbReturnValue == 2 ) {
1639 stream_.state = STREAM_STOPPING;
1640 handle->drainCounter = 2;
1644 else if ( cbReturnValue == 1 ) {
1645 handle->drainCounter = 1;
1646 handle->internalDrain = true;
1650 if ( stream_.mode == OUTPUT || ( stream_.mode == DUPLEX && deviceId == outputDevice ) ) {
1652 if ( handle->drainCounter > 1 ) { // write zeros to the output stream
1654 if ( handle->nStreams[0] == 1 ) {
1655 memset( outBufferList->mBuffers[handle->iStream[0]].mData,
1657 outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );
1659 else { // fill multiple streams with zeros
1660 for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) {
1661 memset( outBufferList->mBuffers[handle->iStream[0]+i].mData,
1663 outBufferList->mBuffers[handle->iStream[0]+i].mDataByteSize );
1667 else if ( handle->nStreams[0] == 1 ) {
1668 if ( stream_.doConvertBuffer[0] ) { // convert directly to CoreAudio stream buffer
1669 convertBuffer( (char *) outBufferList->mBuffers[handle->iStream[0]].mData,
1670 stream_.userBuffer[0], stream_.convertInfo[0] );
1672 else { // copy from user buffer
1673 memcpy( outBufferList->mBuffers[handle->iStream[0]].mData,
1674 stream_.userBuffer[0],
1675 outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );
1678 else { // fill multiple streams
1679 Float32 *inBuffer = (Float32 *) stream_.userBuffer[0];
1680 if ( stream_.doConvertBuffer[0] ) {
1681 convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
1682 inBuffer = (Float32 *) stream_.deviceBuffer;
1685 if ( stream_.deviceInterleaved[0] == false ) { // mono mode
1686 UInt32 bufferBytes = outBufferList->mBuffers[handle->iStream[0]].mDataByteSize;
1687 for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
1688 memcpy( outBufferList->mBuffers[handle->iStream[0]+i].mData,
1689 (void *)&inBuffer[i*stream_.bufferSize], bufferBytes );
1692 else { // fill multiple multi-channel streams with interleaved data
1693 UInt32 streamChannels, channelsLeft, inJump, outJump, inOffset;
1696 bool inInterleaved = ( stream_.userInterleaved ) ? true : false;
1697 UInt32 inChannels = stream_.nUserChannels[0];
1698 if ( stream_.doConvertBuffer[0] ) {
1699 inInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode
1700 inChannels = stream_.nDeviceChannels[0];
1703 if ( inInterleaved ) inOffset = 1;
1704 else inOffset = stream_.bufferSize;
1706 channelsLeft = inChannels;
1707 for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) {
1709 out = (Float32 *) outBufferList->mBuffers[handle->iStream[0]+i].mData;
1710 streamChannels = outBufferList->mBuffers[handle->iStream[0]+i].mNumberChannels;
1713 // Account for possible channel offset in first stream
1714 if ( i == 0 && stream_.channelOffset[0] > 0 ) {
1715 streamChannels -= stream_.channelOffset[0];
1716 outJump = stream_.channelOffset[0];
1720 // Account for possible unfilled channels at end of the last stream
1721 if ( streamChannels > channelsLeft ) {
1722 outJump = streamChannels - channelsLeft;
1723 streamChannels = channelsLeft;
1726 // Determine input buffer offsets and skips
1727 if ( inInterleaved ) {
1728 inJump = inChannels;
1729 in += inChannels - channelsLeft;
1733 in += (inChannels - channelsLeft) * inOffset;
1736 for ( unsigned int i=0; i<stream_.bufferSize; i++ ) {
1737 for ( unsigned int j=0; j<streamChannels; j++ ) {
1738 *out++ = in[j*inOffset];
1743 channelsLeft -= streamChannels;
1749 // Don't bother draining input
1750 if ( handle->drainCounter ) {
1751 handle->drainCounter++;
1755 AudioDeviceID inputDevice;
1756 inputDevice = handle->id[1];
1757 if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && deviceId == inputDevice ) ) {
1759 if ( handle->nStreams[1] == 1 ) {
1760 if ( stream_.doConvertBuffer[1] ) { // convert directly from CoreAudio stream buffer
1761 convertBuffer( stream_.userBuffer[1],
1762 (char *) inBufferList->mBuffers[handle->iStream[1]].mData,
1763 stream_.convertInfo[1] );
1765 else { // copy to user buffer
1766 memcpy( stream_.userBuffer[1],
1767 inBufferList->mBuffers[handle->iStream[1]].mData,
1768 inBufferList->mBuffers[handle->iStream[1]].mDataByteSize );
1771 else { // read from multiple streams
1772 Float32 *outBuffer = (Float32 *) stream_.userBuffer[1];
1773 if ( stream_.doConvertBuffer[1] ) outBuffer = (Float32 *) stream_.deviceBuffer;
1775 if ( stream_.deviceInterleaved[1] == false ) { // mono mode
1776 UInt32 bufferBytes = inBufferList->mBuffers[handle->iStream[1]].mDataByteSize;
1777 for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
1778 memcpy( (void *)&outBuffer[i*stream_.bufferSize],
1779 inBufferList->mBuffers[handle->iStream[1]+i].mData, bufferBytes );
1782 else { // read from multiple multi-channel streams
1783 UInt32 streamChannels, channelsLeft, inJump, outJump, outOffset;
1786 bool outInterleaved = ( stream_.userInterleaved ) ? true : false;
1787 UInt32 outChannels = stream_.nUserChannels[1];
1788 if ( stream_.doConvertBuffer[1] ) {
1789 outInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode
1790 outChannels = stream_.nDeviceChannels[1];
1793 if ( outInterleaved ) outOffset = 1;
1794 else outOffset = stream_.bufferSize;
1796 channelsLeft = outChannels;
1797 for ( unsigned int i=0; i<handle->nStreams[1]; i++ ) {
1799 in = (Float32 *) inBufferList->mBuffers[handle->iStream[1]+i].mData;
1800 streamChannels = inBufferList->mBuffers[handle->iStream[1]+i].mNumberChannels;
1803 // Account for possible channel offset in first stream
1804 if ( i == 0 && stream_.channelOffset[1] > 0 ) {
1805 streamChannels -= stream_.channelOffset[1];
1806 inJump = stream_.channelOffset[1];
1810 // Account for possible unread channels at end of the last stream
1811 if ( streamChannels > channelsLeft ) {
1812 inJump = streamChannels - channelsLeft;
1813 streamChannels = channelsLeft;
1816 // Determine output buffer offsets and skips
1817 if ( outInterleaved ) {
1818 outJump = outChannels;
1819 out += outChannels - channelsLeft;
1823 out += (outChannels - channelsLeft) * outOffset;
1826 for ( unsigned int i=0; i<stream_.bufferSize; i++ ) {
1827 for ( unsigned int j=0; j<streamChannels; j++ ) {
1828 out[j*outOffset] = *in++;
1833 channelsLeft -= streamChannels;
1837 if ( stream_.doConvertBuffer[1] ) { // convert from our internal "device" buffer
1838 convertBuffer( stream_.userBuffer[1],
1839 stream_.deviceBuffer,
1840 stream_.convertInfo[1] );
1846 //MUTEX_UNLOCK( &stream_.mutex );
1848 RtApi::tickStreamTime();
1852 const char* RtApiCore :: getErrorCode( OSStatus code )
1856 case kAudioHardwareNotRunningError:
1857 return "kAudioHardwareNotRunningError";
1859 case kAudioHardwareUnspecifiedError:
1860 return "kAudioHardwareUnspecifiedError";
1862 case kAudioHardwareUnknownPropertyError:
1863 return "kAudioHardwareUnknownPropertyError";
1865 case kAudioHardwareBadPropertySizeError:
1866 return "kAudioHardwareBadPropertySizeError";
1868 case kAudioHardwareIllegalOperationError:
1869 return "kAudioHardwareIllegalOperationError";
1871 case kAudioHardwareBadObjectError:
1872 return "kAudioHardwareBadObjectError";
1874 case kAudioHardwareBadDeviceError:
1875 return "kAudioHardwareBadDeviceError";
1877 case kAudioHardwareBadStreamError:
1878 return "kAudioHardwareBadStreamError";
1880 case kAudioHardwareUnsupportedOperationError:
1881 return "kAudioHardwareUnsupportedOperationError";
1883 case kAudioDeviceUnsupportedFormatError:
1884 return "kAudioDeviceUnsupportedFormatError";
1886 case kAudioDevicePermissionsError:
1887 return "kAudioDevicePermissionsError";
1890 return "CoreAudio unknown error";
1894 //******************** End of __MACOSX_CORE__ *********************//
1897 #if defined(__UNIX_JACK__)
1899 // JACK is a low-latency audio server, originally written for the
1900 // GNU/Linux operating system and now also ported to OS-X. It can
1901 // connect a number of different applications to an audio device, as
1902 // well as allowing them to share audio between themselves.
1904 // When using JACK with RtAudio, "devices" refer to JACK clients that
1905 // have ports connected to the server. The JACK server is typically
1906 // started in a terminal as follows:
1908 // .jackd -d alsa -d hw:0
1910 // or through an interface program such as qjackctl. Many of the
1911 // parameters normally set for a stream are fixed by the JACK server
1912 // and can be specified when the JACK server is started. In
1915 // .jackd -d alsa -d hw:0 -r 44100 -p 512 -n 4
1917 // specifies a sample rate of 44100 Hz, a buffer size of 512 sample
1918 // frames, and number of buffers = 4. Once the server is running, it
1919 // is not possible to override these values. If the values are not
1920 // specified in the command-line, the JACK server uses default values.
1922 // The JACK server does not have to be running when an instance of
1923 // RtApiJack is created, though the function getDeviceCount() will
1924 // report 0 devices found until JACK has been started. When no
1925 // devices are available (i.e., the JACK server is not running), a
1926 // stream cannot be opened.
1928 #include <jack/jack.h>
1932 // A structure to hold various information related to the Jack API
1935 jack_client_t *client;
1936 jack_port_t **ports[2];
1937 std::string deviceName[2];
1939 pthread_cond_t condition;
1940 int drainCounter; // Tracks callback counts when draining
1941 bool internalDrain; // Indicates if stop is initiated from callback or not.
1944 :client(0), drainCounter(0), internalDrain(false) { ports[0] = 0; ports[1] = 0; xrun[0] = false; xrun[1] = false; }
1947 #if !defined(__RTAUDIO_DEBUG__)
1948 static void jackSilentError( const char * ) {};
1951 RtApiJack :: RtApiJack()
1952 :shouldAutoconnect_(true) {
1953 // Nothing to do here.
1954 #if !defined(__RTAUDIO_DEBUG__)
1955 // Turn off Jack's internal error reporting.
1956 jack_set_error_function( &jackSilentError );
1960 RtApiJack :: ~RtApiJack()
1962 if ( stream_.state != STREAM_CLOSED ) closeStream();
1965 unsigned int RtApiJack :: getDeviceCount( void )
1967 // See if we can become a jack client.
1968 jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption;
1969 jack_status_t *status = NULL;
1970 jack_client_t *client = jack_client_open( "RtApiJackCount", options, status );
1971 if ( client == 0 ) return 0;
1974 std::string port, previousPort;
1975 unsigned int nChannels = 0, nDevices = 0;
1976 ports = jack_get_ports( client, NULL, JACK_DEFAULT_AUDIO_TYPE, 0 );
1978 // Parse the port names up to the first colon (:).
1981 port = (char *) ports[ nChannels ];
1982 iColon = port.find(":");
1983 if ( iColon != std::string::npos ) {
1984 port = port.substr( 0, iColon + 1 );
1985 if ( port != previousPort ) {
1987 previousPort = port;
1990 } while ( ports[++nChannels] );
1994 jack_client_close( client );
1998 RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device )
2000 RtAudio::DeviceInfo info;
2001 info.probed = false;
2003 jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption
2004 jack_status_t *status = NULL;
2005 jack_client_t *client = jack_client_open( "RtApiJackInfo", options, status );
2006 if ( client == 0 ) {
2007 errorText_ = "RtApiJack::getDeviceInfo: Jack server not found or connection error!";
2008 error( RtAudioError::WARNING );
2013 std::string port, previousPort;
2014 unsigned int nPorts = 0, nDevices = 0;
2015 ports = jack_get_ports( client, NULL, JACK_DEFAULT_AUDIO_TYPE, 0 );
2017 // Parse the port names up to the first colon (:).
2020 port = (char *) ports[ nPorts ];
2021 iColon = port.find(":");
2022 if ( iColon != std::string::npos ) {
2023 port = port.substr( 0, iColon );
2024 if ( port != previousPort ) {
2025 if ( nDevices == device ) info.name = port;
2027 previousPort = port;
2030 } while ( ports[++nPorts] );
2034 if ( device >= nDevices ) {
2035 jack_client_close( client );
2036 errorText_ = "RtApiJack::getDeviceInfo: device ID is invalid!";
2037 error( RtAudioError::INVALID_USE );
2041 // Get the current jack server sample rate.
2042 info.sampleRates.clear();
2044 info.preferredSampleRate = jack_get_sample_rate( client );
2045 info.sampleRates.push_back( info.preferredSampleRate );
2047 // Count the available ports containing the client name as device
2048 // channels. Jack "input ports" equal RtAudio output channels.
2049 unsigned int nChannels = 0;
2050 ports = jack_get_ports( client, info.name.c_str(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput );
2052 while ( ports[ nChannels ] ) nChannels++;
2054 info.outputChannels = nChannels;
2057 // Jack "output ports" equal RtAudio input channels.
2059 ports = jack_get_ports( client, info.name.c_str(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput );
2061 while ( ports[ nChannels ] ) nChannels++;
2063 info.inputChannels = nChannels;
2066 if ( info.outputChannels == 0 && info.inputChannels == 0 ) {
2067 jack_client_close(client);
2068 errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!";
2069 error( RtAudioError::WARNING );
2073 // If device opens for both playback and capture, we determine the channels.
2074 if ( info.outputChannels > 0 && info.inputChannels > 0 )
2075 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
2077 // Jack always uses 32-bit floats.
2078 info.nativeFormats = RTAUDIO_FLOAT32;
2080 // Jack doesn't provide default devices so we'll use the first available one.
2081 if ( device == 0 && info.outputChannels > 0 )
2082 info.isDefaultOutput = true;
2083 if ( device == 0 && info.inputChannels > 0 )
2084 info.isDefaultInput = true;
2086 jack_client_close(client);
2091 static int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer )
2093 CallbackInfo *info = (CallbackInfo *) infoPointer;
2095 RtApiJack *object = (RtApiJack *) info->object;
2096 if ( object->callbackEvent( (unsigned long) nframes ) == false ) return 1;
2101 // This function will be called by a spawned thread when the Jack
2102 // server signals that it is shutting down. It is necessary to handle
2103 // it this way because the jackShutdown() function must return before
2104 // the jack_deactivate() function (in closeStream()) will return.
2105 static void *jackCloseStream( void *ptr )
2107 CallbackInfo *info = (CallbackInfo *) ptr;
2108 RtApiJack *object = (RtApiJack *) info->object;
2110 object->closeStream();
2112 pthread_exit( NULL );
2114 static void jackShutdown( void *infoPointer )
2116 CallbackInfo *info = (CallbackInfo *) infoPointer;
2117 RtApiJack *object = (RtApiJack *) info->object;
2119 // Check current stream state. If stopped, then we'll assume this
2120 // was called as a result of a call to RtApiJack::stopStream (the
2121 // deactivation of a client handle causes this function to be called).
2122 // If not, we'll assume the Jack server is shutting down or some
2123 // other problem occurred and we should close the stream.
2124 if ( object->isStreamRunning() == false ) return;
2126 ThreadHandle threadId;
2127 pthread_create( &threadId, NULL, jackCloseStream, info );
2128 std::cerr << "\nRtApiJack: the Jack server is shutting down this client ... stream stopped and closed!!\n" << std::endl;
2131 static int jackXrun( void *infoPointer )
2133 JackHandle *handle = *((JackHandle **) infoPointer);
2135 if ( handle->ports[0] ) handle->xrun[0] = true;
2136 if ( handle->ports[1] ) handle->xrun[1] = true;
2141 bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
2142 unsigned int firstChannel, unsigned int sampleRate,
2143 RtAudioFormat format, unsigned int *bufferSize,
2144 RtAudio::StreamOptions *options )
2146 JackHandle *handle = (JackHandle *) stream_.apiHandle;
2148 // Look for jack server and try to become a client (only do once per stream).
2149 jack_client_t *client = 0;
2150 if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) {
2151 jack_options_t jackoptions = (jack_options_t) ( JackNoStartServer ); //JackNullOption;
2152 jack_status_t *status = NULL;
2153 if ( options && !options->streamName.empty() )
2154 client = jack_client_open( options->streamName.c_str(), jackoptions, status );
2156 client = jack_client_open( "RtApiJack", jackoptions, status );
2157 if ( client == 0 ) {
2158 errorText_ = "RtApiJack::probeDeviceOpen: Jack server not found or connection error!";
2159 error( RtAudioError::WARNING );
2164 // The handle must have been created on an earlier pass.
2165 client = handle->client;
2169 std::string port, previousPort, deviceName;
2170 unsigned int nPorts = 0, nDevices = 0;
2171 ports = jack_get_ports( client, NULL, JACK_DEFAULT_AUDIO_TYPE, 0 );
2173 // Parse the port names up to the first colon (:).
2176 port = (char *) ports[ nPorts ];
2177 iColon = port.find(":");
2178 if ( iColon != std::string::npos ) {
2179 port = port.substr( 0, iColon );
2180 if ( port != previousPort ) {
2181 if ( nDevices == device ) deviceName = port;
2183 previousPort = port;
2186 } while ( ports[++nPorts] );
2190 if ( device >= nDevices ) {
2191 errorText_ = "RtApiJack::probeDeviceOpen: device ID is invalid!";
2195 unsigned long flag = JackPortIsInput;
2196 if ( mode == INPUT ) flag = JackPortIsOutput;
2198 if ( ! (options && (options->flags & RTAUDIO_JACK_DONT_CONNECT)) ) {
2199 // Count the available ports containing the client name as device
2200 // channels. Jack "input ports" equal RtAudio output channels.
2201 unsigned int nChannels = 0;
2202 ports = jack_get_ports( client, deviceName.c_str(), JACK_DEFAULT_AUDIO_TYPE, flag );
2204 while ( ports[ nChannels ] ) nChannels++;
2207 // Compare the jack ports for specified client to the requested number of channels.
2208 if ( nChannels < (channels + firstChannel) ) {
2209 errorStream_ << "RtApiJack::probeDeviceOpen: requested number of channels (" << channels << ") + offset (" << firstChannel << ") not found for specified device (" << device << ":" << deviceName << ").";
2210 errorText_ = errorStream_.str();
2215 // Check the jack server sample rate.
2216 unsigned int jackRate = jack_get_sample_rate( client );
2217 if ( sampleRate != jackRate ) {
2218 jack_client_close( client );
2219 errorStream_ << "RtApiJack::probeDeviceOpen: the requested sample rate (" << sampleRate << ") is different than the JACK server rate (" << jackRate << ").";
2220 errorText_ = errorStream_.str();
2223 stream_.sampleRate = jackRate;
2225 // Get the latency of the JACK port.
2226 ports = jack_get_ports( client, deviceName.c_str(), JACK_DEFAULT_AUDIO_TYPE, flag );
2227 if ( ports[ firstChannel ] ) {
2229 jack_latency_callback_mode_t cbmode = (mode == INPUT ? JackCaptureLatency : JackPlaybackLatency);
2230 // the range (usually the min and max are equal)
2231 jack_latency_range_t latrange; latrange.min = latrange.max = 0;
2232 // get the latency range
2233 jack_port_get_latency_range( jack_port_by_name( client, ports[firstChannel] ), cbmode, &latrange );
2234 // be optimistic, use the min!
2235 stream_.latency[mode] = latrange.min;
2236 //stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) );
2240 // The jack server always uses 32-bit floating-point data.
2241 stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
2242 stream_.userFormat = format;
2244 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
2245 else stream_.userInterleaved = true;
2247 // Jack always uses non-interleaved buffers.
2248 stream_.deviceInterleaved[mode] = false;
2250 // Jack always provides host byte-ordered data.
2251 stream_.doByteSwap[mode] = false;
2253 // Get the buffer size. The buffer size and number of buffers
2254 // (periods) is set when the jack server is started.
2255 stream_.bufferSize = (int) jack_get_buffer_size( client );
2256 *bufferSize = stream_.bufferSize;
2258 stream_.nDeviceChannels[mode] = channels;
2259 stream_.nUserChannels[mode] = channels;
2261 // Set flags for buffer conversion.
2262 stream_.doConvertBuffer[mode] = false;
2263 if ( stream_.userFormat != stream_.deviceFormat[mode] )
2264 stream_.doConvertBuffer[mode] = true;
2265 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
2266 stream_.nUserChannels[mode] > 1 )
2267 stream_.doConvertBuffer[mode] = true;
2269 // Allocate our JackHandle structure for the stream.
2270 if ( handle == 0 ) {
2272 handle = new JackHandle;
2274 catch ( std::bad_alloc& ) {
2275 errorText_ = "RtApiJack::probeDeviceOpen: error allocating JackHandle memory.";
2279 if ( pthread_cond_init(&handle->condition, NULL) ) {
2280 errorText_ = "RtApiJack::probeDeviceOpen: error initializing pthread condition variable.";
2283 stream_.apiHandle = (void *) handle;
2284 handle->client = client;
2286 handle->deviceName[mode] = deviceName;
2288 // Allocate necessary internal buffers.
2289 unsigned long bufferBytes;
2290 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
2291 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
2292 if ( stream_.userBuffer[mode] == NULL ) {
2293 errorText_ = "RtApiJack::probeDeviceOpen: error allocating user buffer memory.";
2297 if ( stream_.doConvertBuffer[mode] ) {
2299 bool makeBuffer = true;
2300 if ( mode == OUTPUT )
2301 bufferBytes = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
2302 else { // mode == INPUT
2303 bufferBytes = stream_.nDeviceChannels[1] * formatBytes( stream_.deviceFormat[1] );
2304 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
2305 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]);
2306 if ( bufferBytes < bytesOut ) makeBuffer = false;
2311 bufferBytes *= *bufferSize;
2312 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
2313 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
2314 if ( stream_.deviceBuffer == NULL ) {
2315 errorText_ = "RtApiJack::probeDeviceOpen: error allocating device buffer memory.";
2321 // Allocate memory for the Jack ports (channels) identifiers.
2322 handle->ports[mode] = (jack_port_t **) malloc ( sizeof (jack_port_t *) * channels );
2323 if ( handle->ports[mode] == NULL ) {
2324 errorText_ = "RtApiJack::probeDeviceOpen: error allocating port memory.";
2328 stream_.device[mode] = device;
2329 stream_.channelOffset[mode] = firstChannel;
2330 stream_.state = STREAM_STOPPED;
2331 stream_.callbackInfo.object = (void *) this;
2333 if ( stream_.mode == OUTPUT && mode == INPUT )
2334 // We had already set up the stream for output.
2335 stream_.mode = DUPLEX;
2337 stream_.mode = mode;
2338 jack_set_process_callback( handle->client, jackCallbackHandler, (void *) &stream_.callbackInfo );
2339 jack_set_xrun_callback( handle->client, jackXrun, (void *) &stream_.apiHandle );
2340 jack_on_shutdown( handle->client, jackShutdown, (void *) &stream_.callbackInfo );
2343 // Register our ports.
2345 if ( mode == OUTPUT ) {
2346 for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
2347 snprintf( label, 64, "outport %d", i );
2348 handle->ports[0][i] = jack_port_register( handle->client, (const char *)label,
2349 JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0 );
2353 for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
2354 snprintf( label, 64, "inport %d", i );
2355 handle->ports[1][i] = jack_port_register( handle->client, (const char *)label,
2356 JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0 );
2360 // Setup the buffer conversion information structure. We don't use
2361 // buffers to do channel offsets, so we override that parameter
2363 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );
2365 if ( options && options->flags & RTAUDIO_JACK_DONT_CONNECT ) shouldAutoconnect_ = false;
2371 pthread_cond_destroy( &handle->condition );
2372 jack_client_close( handle->client );
2374 if ( handle->ports[0] ) free( handle->ports[0] );
2375 if ( handle->ports[1] ) free( handle->ports[1] );
2378 stream_.apiHandle = 0;
2381 for ( int i=0; i<2; i++ ) {
2382 if ( stream_.userBuffer[i] ) {
2383 free( stream_.userBuffer[i] );
2384 stream_.userBuffer[i] = 0;
2388 if ( stream_.deviceBuffer ) {
2389 free( stream_.deviceBuffer );
2390 stream_.deviceBuffer = 0;
2396 void RtApiJack :: closeStream( void )
2398 if ( stream_.state == STREAM_CLOSED ) {
2399 errorText_ = "RtApiJack::closeStream(): no open stream to close!";
2400 error( RtAudioError::WARNING );
2404 JackHandle *handle = (JackHandle *) stream_.apiHandle;
2407 if ( stream_.state == STREAM_RUNNING )
2408 jack_deactivate( handle->client );
2410 jack_client_close( handle->client );
2414 if ( handle->ports[0] ) free( handle->ports[0] );
2415 if ( handle->ports[1] ) free( handle->ports[1] );
2416 pthread_cond_destroy( &handle->condition );
2418 stream_.apiHandle = 0;
2421 for ( int i=0; i<2; i++ ) {
2422 if ( stream_.userBuffer[i] ) {
2423 free( stream_.userBuffer[i] );
2424 stream_.userBuffer[i] = 0;
2428 if ( stream_.deviceBuffer ) {
2429 free( stream_.deviceBuffer );
2430 stream_.deviceBuffer = 0;
2433 stream_.mode = UNINITIALIZED;
2434 stream_.state = STREAM_CLOSED;
2437 void RtApiJack :: startStream( void )
2440 if ( stream_.state == STREAM_RUNNING ) {
2441 errorText_ = "RtApiJack::startStream(): the stream is already running!";
2442 error( RtAudioError::WARNING );
2446 JackHandle *handle = (JackHandle *) stream_.apiHandle;
2447 int result = jack_activate( handle->client );
2449 errorText_ = "RtApiJack::startStream(): unable to activate JACK client!";
2455 // Get the list of available ports.
2456 if ( shouldAutoconnect_ && (stream_.mode == OUTPUT || stream_.mode == DUPLEX) ) {
2458 ports = jack_get_ports( handle->client, handle->deviceName[0].c_str(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput);
2459 if ( ports == NULL) {
2460 errorText_ = "RtApiJack::startStream(): error determining available JACK input ports!";
2464 // Now make the port connections. Since RtAudio wasn't designed to
2465 // allow the user to select particular channels of a device, we'll
2466 // just open the first "nChannels" ports with offset.
2467 for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
2469 if ( ports[ stream_.channelOffset[0] + i ] )
2470 result = jack_connect( handle->client, jack_port_name( handle->ports[0][i] ), ports[ stream_.channelOffset[0] + i ] );
2473 errorText_ = "RtApiJack::startStream(): error connecting output ports!";
2480 if ( shouldAutoconnect_ && (stream_.mode == INPUT || stream_.mode == DUPLEX) ) {
2482 ports = jack_get_ports( handle->client, handle->deviceName[1].c_str(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput );
2483 if ( ports == NULL) {
2484 errorText_ = "RtApiJack::startStream(): error determining available JACK output ports!";
2488 // Now make the port connections. See note above.
2489 for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
2491 if ( ports[ stream_.channelOffset[1] + i ] )
2492 result = jack_connect( handle->client, ports[ stream_.channelOffset[1] + i ], jack_port_name( handle->ports[1][i] ) );
2495 errorText_ = "RtApiJack::startStream(): error connecting input ports!";
2502 handle->drainCounter = 0;
2503 handle->internalDrain = false;
2504 stream_.state = STREAM_RUNNING;
2507 if ( result == 0 ) return;
2508 error( RtAudioError::SYSTEM_ERROR );
2511 void RtApiJack :: stopStream( void )
2514 if ( stream_.state == STREAM_STOPPED ) {
2515 errorText_ = "RtApiJack::stopStream(): the stream is already stopped!";
2516 error( RtAudioError::WARNING );
2520 JackHandle *handle = (JackHandle *) stream_.apiHandle;
2521 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
2523 if ( handle->drainCounter == 0 ) {
2524 handle->drainCounter = 2;
2525 pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
2529 jack_deactivate( handle->client );
2530 stream_.state = STREAM_STOPPED;
2533 void RtApiJack :: abortStream( void )
2536 if ( stream_.state == STREAM_STOPPED ) {
2537 errorText_ = "RtApiJack::abortStream(): the stream is already stopped!";
2538 error( RtAudioError::WARNING );
2542 JackHandle *handle = (JackHandle *) stream_.apiHandle;
2543 handle->drainCounter = 2;
2548 // This function will be called by a spawned thread when the user
2549 // callback function signals that the stream should be stopped or
2550 // aborted. It is necessary to handle it this way because the
2551 // callbackEvent() function must return before the jack_deactivate()
2552 // function will return.
2553 static void *jackStopStream( void *ptr )
2555 CallbackInfo *info = (CallbackInfo *) ptr;
2556 RtApiJack *object = (RtApiJack *) info->object;
2558 object->stopStream();
2559 pthread_exit( NULL );
2562 bool RtApiJack :: callbackEvent( unsigned long nframes )
2564 if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;
2565 if ( stream_.state == STREAM_CLOSED ) {
2566 errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";
2567 error( RtAudioError::WARNING );
2570 if ( stream_.bufferSize != nframes ) {
2571 errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!";
2572 error( RtAudioError::WARNING );
2576 CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
2577 JackHandle *handle = (JackHandle *) stream_.apiHandle;
2579 // Check if we were draining the stream and signal is finished.
2580 if ( handle->drainCounter > 3 ) {
2581 ThreadHandle threadId;
2583 stream_.state = STREAM_STOPPING;
2584 if ( handle->internalDrain == true )
2585 pthread_create( &threadId, NULL, jackStopStream, info );
2587 pthread_cond_signal( &handle->condition );
2591 // Invoke user callback first, to get fresh output data.
2592 if ( handle->drainCounter == 0 ) {
2593 RtAudioCallback callback = (RtAudioCallback) info->callback;
2594 double streamTime = getStreamTime();
2595 RtAudioStreamStatus status = 0;
2596 if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
2597 status |= RTAUDIO_OUTPUT_UNDERFLOW;
2598 handle->xrun[0] = false;
2600 if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
2601 status |= RTAUDIO_INPUT_OVERFLOW;
2602 handle->xrun[1] = false;
2604 int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
2605 stream_.bufferSize, streamTime, status, info->userData );
2606 if ( cbReturnValue == 2 ) {
2607 stream_.state = STREAM_STOPPING;
2608 handle->drainCounter = 2;
2610 pthread_create( &id, NULL, jackStopStream, info );
2613 else if ( cbReturnValue == 1 ) {
2614 handle->drainCounter = 1;
2615 handle->internalDrain = true;
2619 jack_default_audio_sample_t *jackbuffer;
2620 unsigned long bufferBytes = nframes * sizeof( jack_default_audio_sample_t );
2621 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
2623 if ( handle->drainCounter > 1 ) { // write zeros to the output stream
2625 for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
2626 jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
2627 memset( jackbuffer, 0, bufferBytes );
2631 else if ( stream_.doConvertBuffer[0] ) {
2633 convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
2635 for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
2636 jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
2637 memcpy( jackbuffer, &stream_.deviceBuffer[i*bufferBytes], bufferBytes );
2640 else { // no buffer conversion
2641 for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
2642 jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
2643 memcpy( jackbuffer, &stream_.userBuffer[0][i*bufferBytes], bufferBytes );
2648 // Don't bother draining input
2649 if ( handle->drainCounter ) {
2650 handle->drainCounter++;
2654 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
2656 if ( stream_.doConvertBuffer[1] ) {
2657 for ( unsigned int i=0; i<stream_.nDeviceChannels[1]; i++ ) {
2658 jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );
2659 memcpy( &stream_.deviceBuffer[i*bufferBytes], jackbuffer, bufferBytes );
2661 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
2663 else { // no buffer conversion
2664 for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
2665 jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );
2666 memcpy( &stream_.userBuffer[1][i*bufferBytes], jackbuffer, bufferBytes );
2672 RtApi::tickStreamTime();
2675 //******************** End of __UNIX_JACK__ *********************//
2678 #if defined(__WINDOWS_ASIO__) // ASIO API on Windows
2680 // The ASIO API is designed around a callback scheme, so this
2681 // implementation is similar to that used for OS-X CoreAudio and Linux
2682 // Jack. The primary constraint with ASIO is that it only allows
2683 // access to a single driver at a time. Thus, it is not possible to
2684 // have more than one simultaneous RtAudio stream.
2686 // This implementation also requires a number of external ASIO files
2687 // and a few global variables. The ASIO callback scheme does not
2688 // allow for the passing of user data, so we must create a global
2689 // pointer to our callbackInfo structure.
2691 // On unix systems, we make use of a pthread condition variable.
2692 // Since there is no equivalent in Windows, I hacked something based
2693 // on information found in
2694 // http://www.cs.wustl.edu/~schmidt/win32-cv-1.html.
2696 #include "asiosys.h"
2698 #include "iasiothiscallresolver.h"
2699 #include "asiodrivers.h"
2702 static AsioDrivers drivers;
2703 static ASIOCallbacks asioCallbacks;
2704 static ASIODriverInfo driverInfo;
2705 static CallbackInfo *asioCallbackInfo;
2706 static bool asioXRun;
2709 int drainCounter; // Tracks callback counts when draining
2710 bool internalDrain; // Indicates if stop is initiated from callback or not.
2711 ASIOBufferInfo *bufferInfos;
2715 :drainCounter(0), internalDrain(false), bufferInfos(0) {}
2718 // Function declarations (definitions at end of section)
2719 static const char* getAsioErrorString( ASIOError result );
2720 static void sampleRateChanged( ASIOSampleRate sRate );
2721 static long asioMessages( long selector, long value, void* message, double* opt );
2723 RtApiAsio :: RtApiAsio()
2725 // ASIO cannot run on a multi-threaded appartment. You can call
2726 // CoInitialize beforehand, but it must be for appartment threading
2727 // (in which case, CoInitilialize will return S_FALSE here).
2728 coInitialized_ = false;
2729 HRESULT hr = CoInitialize( NULL );
2731 errorText_ = "RtApiAsio::ASIO requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)";
2732 error( RtAudioError::WARNING );
2734 coInitialized_ = true;
2736 drivers.removeCurrentDriver();
2737 driverInfo.asioVersion = 2;
2739 // See note in DirectSound implementation about GetDesktopWindow().
2740 driverInfo.sysRef = GetForegroundWindow();
2743 RtApiAsio :: ~RtApiAsio()
2745 if ( stream_.state != STREAM_CLOSED ) closeStream();
2746 if ( coInitialized_ ) CoUninitialize();
2749 unsigned int RtApiAsio :: getDeviceCount( void )
2751 return (unsigned int) drivers.asioGetNumDev();
2754 RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device )
2756 RtAudio::DeviceInfo info;
2757 info.probed = false;
2760 unsigned int nDevices = getDeviceCount();
2761 if ( nDevices == 0 ) {
2762 errorText_ = "RtApiAsio::getDeviceInfo: no devices found!";
2763 error( RtAudioError::INVALID_USE );
2767 if ( device >= nDevices ) {
2768 errorText_ = "RtApiAsio::getDeviceInfo: device ID is invalid!";
2769 error( RtAudioError::INVALID_USE );
2773 // If a stream is already open, we cannot probe other devices. Thus, use the saved results.
2774 if ( stream_.state != STREAM_CLOSED ) {
2775 if ( device >= devices_.size() ) {
2776 errorText_ = "RtApiAsio::getDeviceInfo: device ID was not present before stream was opened.";
2777 error( RtAudioError::WARNING );
2780 return devices_[ device ];
2783 char driverName[32];
2784 ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 );
2785 if ( result != ASE_OK ) {
2786 errorStream_ << "RtApiAsio::getDeviceInfo: unable to get driver name (" << getAsioErrorString( result ) << ").";
2787 errorText_ = errorStream_.str();
2788 error( RtAudioError::WARNING );
2792 info.name = driverName;
2794 if ( !drivers.loadDriver( driverName ) ) {
2795 errorStream_ << "RtApiAsio::getDeviceInfo: unable to load driver (" << driverName << ").";
2796 errorText_ = errorStream_.str();
2797 error( RtAudioError::WARNING );
2801 result = ASIOInit( &driverInfo );
2802 if ( result != ASE_OK ) {
2803 errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
2804 errorText_ = errorStream_.str();
2805 error( RtAudioError::WARNING );
2809 // Determine the device channel information.
2810 long inputChannels, outputChannels;
2811 result = ASIOGetChannels( &inputChannels, &outputChannels );
2812 if ( result != ASE_OK ) {
2813 drivers.removeCurrentDriver();
2814 errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
2815 errorText_ = errorStream_.str();
2816 error( RtAudioError::WARNING );
2820 info.outputChannels = outputChannels;
2821 info.inputChannels = inputChannels;
2822 if ( info.outputChannels > 0 && info.inputChannels > 0 )
2823 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
2825 // Determine the supported sample rates.
2826 info.sampleRates.clear();
2827 for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {
2828 result = ASIOCanSampleRate( (ASIOSampleRate) SAMPLE_RATES[i] );
2829 if ( result == ASE_OK ) {
2830 info.sampleRates.push_back( SAMPLE_RATES[i] );
2832 if ( !info.preferredSampleRate || ( SAMPLE_RATES[i] <= 48000 && SAMPLE_RATES[i] > info.preferredSampleRate ) )
2833 info.preferredSampleRate = SAMPLE_RATES[i];
2837 // Determine supported data types ... just check first channel and assume rest are the same.
2838 ASIOChannelInfo channelInfo;
2839 channelInfo.channel = 0;
2840 channelInfo.isInput = true;
2841 if ( info.inputChannels <= 0 ) channelInfo.isInput = false;
2842 result = ASIOGetChannelInfo( &channelInfo );
2843 if ( result != ASE_OK ) {
2844 drivers.removeCurrentDriver();
2845 errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting driver channel info (" << driverName << ").";
2846 errorText_ = errorStream_.str();
2847 error( RtAudioError::WARNING );
2851 info.nativeFormats = 0;
2852 if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB )
2853 info.nativeFormats |= RTAUDIO_SINT16;
2854 else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB )
2855 info.nativeFormats |= RTAUDIO_SINT32;
2856 else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB )
2857 info.nativeFormats |= RTAUDIO_FLOAT32;
2858 else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB )
2859 info.nativeFormats |= RTAUDIO_FLOAT64;
2860 else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB )
2861 info.nativeFormats |= RTAUDIO_SINT24;
2863 if ( info.outputChannels > 0 )
2864 if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true;
2865 if ( info.inputChannels > 0 )
2866 if ( getDefaultInputDevice() == device ) info.isDefaultInput = true;
2869 drivers.removeCurrentDriver();
2873 static void bufferSwitch( long index, ASIOBool /*processNow*/ )
2875 RtApiAsio *object = (RtApiAsio *) asioCallbackInfo->object;
2876 object->callbackEvent( index );
2879 void RtApiAsio :: saveDeviceInfo( void )
2883 unsigned int nDevices = getDeviceCount();
2884 devices_.resize( nDevices );
2885 for ( unsigned int i=0; i<nDevices; i++ )
2886 devices_[i] = getDeviceInfo( i );
2889 bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
2890 unsigned int firstChannel, unsigned int sampleRate,
2891 RtAudioFormat format, unsigned int *bufferSize,
2892 RtAudio::StreamOptions *options )
2893 {////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
2895 bool isDuplexInput = mode == INPUT && stream_.mode == OUTPUT;
2897 // For ASIO, a duplex stream MUST use the same driver.
2898 if ( isDuplexInput && stream_.device[0] != device ) {
2899 errorText_ = "RtApiAsio::probeDeviceOpen: an ASIO duplex stream must use the same device for input and output!";
2903 char driverName[32];
2904 ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 );
2905 if ( result != ASE_OK ) {
2906 errorStream_ << "RtApiAsio::probeDeviceOpen: unable to get driver name (" << getAsioErrorString( result ) << ").";
2907 errorText_ = errorStream_.str();
2911 // Only load the driver once for duplex stream.
2912 if ( !isDuplexInput ) {
2913 // The getDeviceInfo() function will not work when a stream is open
2914 // because ASIO does not allow multiple devices to run at the same
2915 // time. Thus, we'll probe the system before opening a stream and
2916 // save the results for use by getDeviceInfo().
2917 this->saveDeviceInfo();
2919 if ( !drivers.loadDriver( driverName ) ) {
2920 errorStream_ << "RtApiAsio::probeDeviceOpen: unable to load driver (" << driverName << ").";
2921 errorText_ = errorStream_.str();
2925 result = ASIOInit( &driverInfo );
2926 if ( result != ASE_OK ) {
2927 errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
2928 errorText_ = errorStream_.str();
2933 // keep them before any "goto error", they are used for error cleanup + goto device boundary checks
2934 bool buffersAllocated = false;
2935 AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
2936 unsigned int nChannels;
2939 // Check the device channel count.
2940 long inputChannels, outputChannels;
2941 result = ASIOGetChannels( &inputChannels, &outputChannels );
2942 if ( result != ASE_OK ) {
2943 errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
2944 errorText_ = errorStream_.str();
2948 if ( ( mode == OUTPUT && (channels+firstChannel) > (unsigned int) outputChannels) ||
2949 ( mode == INPUT && (channels+firstChannel) > (unsigned int) inputChannels) ) {
2950 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested channel count (" << channels << ") + offset (" << firstChannel << ").";
2951 errorText_ = errorStream_.str();
2954 stream_.nDeviceChannels[mode] = channels;
2955 stream_.nUserChannels[mode] = channels;
2956 stream_.channelOffset[mode] = firstChannel;
2958 // Verify the sample rate is supported.
2959 result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate );
2960 if ( result != ASE_OK ) {
2961 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested sample rate (" << sampleRate << ").";
2962 errorText_ = errorStream_.str();
2966 // Get the current sample rate
2967 ASIOSampleRate currentRate;
2968 result = ASIOGetSampleRate( ¤tRate );
2969 if ( result != ASE_OK ) {
2970 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error getting sample rate.";
2971 errorText_ = errorStream_.str();
2975 // Set the sample rate only if necessary
2976 if ( currentRate != sampleRate ) {
2977 result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate );
2978 if ( result != ASE_OK ) {
2979 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ").";
2980 errorText_ = errorStream_.str();
2985 // Determine the driver data type.
2986 ASIOChannelInfo channelInfo;
2987 channelInfo.channel = 0;
2988 if ( mode == OUTPUT ) channelInfo.isInput = false;
2989 else channelInfo.isInput = true;
2990 result = ASIOGetChannelInfo( &channelInfo );
2991 if ( result != ASE_OK ) {
2992 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting data format.";
2993 errorText_ = errorStream_.str();
2997 // Assuming WINDOWS host is always little-endian.
2998 stream_.doByteSwap[mode] = false;
2999 stream_.userFormat = format;
3000 stream_.deviceFormat[mode] = 0;
3001 if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) {
3002 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
3003 if ( channelInfo.type == ASIOSTInt16MSB ) stream_.doByteSwap[mode] = true;
3005 else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) {
3006 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
3007 if ( channelInfo.type == ASIOSTInt32MSB ) stream_.doByteSwap[mode] = true;
3009 else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) {
3010 stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
3011 if ( channelInfo.type == ASIOSTFloat32MSB ) stream_.doByteSwap[mode] = true;
3013 else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) {
3014 stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
3015 if ( channelInfo.type == ASIOSTFloat64MSB ) stream_.doByteSwap[mode] = true;
3017 else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB ) {
3018 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
3019 if ( channelInfo.type == ASIOSTInt24MSB ) stream_.doByteSwap[mode] = true;
3022 if ( stream_.deviceFormat[mode] == 0 ) {
3023 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") data format not supported by RtAudio.";
3024 errorText_ = errorStream_.str();
3028 // Set the buffer size. For a duplex stream, this will end up
3029 // setting the buffer size based on the input constraints, which
3031 long minSize, maxSize, preferSize, granularity;
3032 result = ASIOGetBufferSize( &minSize, &maxSize, &preferSize, &granularity );
3033 if ( result != ASE_OK ) {
3034 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting buffer size.";
3035 errorText_ = errorStream_.str();
3039 if ( isDuplexInput ) {
3040 // When this is the duplex input (output was opened before), then we have to use the same
3041 // buffersize as the output, because it might use the preferred buffer size, which most
3042 // likely wasn't passed as input to this. The buffer sizes have to be identically anyway,
3043 // So instead of throwing an error, make them equal. The caller uses the reference
3044 // to the "bufferSize" param as usual to set up processing buffers.
3046 *bufferSize = stream_.bufferSize;
3049 if ( *bufferSize == 0 ) *bufferSize = preferSize;
3050 else if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;
3051 else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;
3052 else if ( granularity == -1 ) {
3053 // Make sure bufferSize is a power of two.
3054 int log2_of_min_size = 0;
3055 int log2_of_max_size = 0;
3057 for ( unsigned int i = 0; i < sizeof(long) * 8; i++ ) {
3058 if ( minSize & ((long)1 << i) ) log2_of_min_size = i;
3059 if ( maxSize & ((long)1 << i) ) log2_of_max_size = i;
3062 long min_delta = std::abs( (long)*bufferSize - ((long)1 << log2_of_min_size) );
3063 int min_delta_num = log2_of_min_size;
3065 for (int i = log2_of_min_size + 1; i <= log2_of_max_size; i++) {
3066 long current_delta = std::abs( (long)*bufferSize - ((long)1 << i) );
3067 if (current_delta < min_delta) {
3068 min_delta = current_delta;
3073 *bufferSize = ( (unsigned int)1 << min_delta_num );
3074 if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;
3075 else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;
3077 else if ( granularity != 0 ) {
3078 // Set to an even multiple of granularity, rounding up.
3079 *bufferSize = (*bufferSize + granularity-1) / granularity * granularity;
3084 // we don't use it anymore, see above!
3085 // Just left it here for the case...
3086 if ( isDuplexInput && stream_.bufferSize != *bufferSize ) {
3087 errorText_ = "RtApiAsio::probeDeviceOpen: input/output buffersize discrepancy!";
3092 stream_.bufferSize = *bufferSize;
3093 stream_.nBuffers = 2;
3095 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
3096 else stream_.userInterleaved = true;
3098 // ASIO always uses non-interleaved buffers.
3099 stream_.deviceInterleaved[mode] = false;
3101 // Allocate, if necessary, our AsioHandle structure for the stream.
3102 if ( handle == 0 ) {
3104 handle = new AsioHandle;
3106 catch ( std::bad_alloc& ) {
3107 errorText_ = "RtApiAsio::probeDeviceOpen: error allocating AsioHandle memory.";
3110 handle->bufferInfos = 0;
3112 // Create a manual-reset event.
3113 handle->condition = CreateEvent( NULL, // no security
3114 TRUE, // manual-reset
3115 FALSE, // non-signaled initially
3117 stream_.apiHandle = (void *) handle;
3120 // Create the ASIO internal buffers. Since RtAudio sets up input
3121 // and output separately, we'll have to dispose of previously
3122 // created output buffers for a duplex stream.
3123 if ( mode == INPUT && stream_.mode == OUTPUT ) {
3124 ASIODisposeBuffers();
3125 if ( handle->bufferInfos ) free( handle->bufferInfos );
3128 // Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure.
3130 nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
3131 handle->bufferInfos = (ASIOBufferInfo *) malloc( nChannels * sizeof(ASIOBufferInfo) );
3132 if ( handle->bufferInfos == NULL ) {
3133 errorStream_ << "RtApiAsio::probeDeviceOpen: error allocating bufferInfo memory for driver (" << driverName << ").";
3134 errorText_ = errorStream_.str();
3138 ASIOBufferInfo *infos;
3139 infos = handle->bufferInfos;
3140 for ( i=0; i<stream_.nDeviceChannels[0]; i++, infos++ ) {
3141 infos->isInput = ASIOFalse;
3142 infos->channelNum = i + stream_.channelOffset[0];
3143 infos->buffers[0] = infos->buffers[1] = 0;
3145 for ( i=0; i<stream_.nDeviceChannels[1]; i++, infos++ ) {
3146 infos->isInput = ASIOTrue;
3147 infos->channelNum = i + stream_.channelOffset[1];
3148 infos->buffers[0] = infos->buffers[1] = 0;
3151 // prepare for callbacks
3152 stream_.sampleRate = sampleRate;
3153 stream_.device[mode] = device;
3154 stream_.mode = isDuplexInput ? DUPLEX : mode;
3156 // store this class instance before registering callbacks, that are going to use it
3157 asioCallbackInfo = &stream_.callbackInfo;
3158 stream_.callbackInfo.object = (void *) this;
3160 // Set up the ASIO callback structure and create the ASIO data buffers.
3161 asioCallbacks.bufferSwitch = &bufferSwitch;
3162 asioCallbacks.sampleRateDidChange = &sampleRateChanged;
3163 asioCallbacks.asioMessage = &asioMessages;
3164 asioCallbacks.bufferSwitchTimeInfo = NULL;
3165 result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks );
3166 if ( result != ASE_OK ) {
3167 // Standard method failed. This can happen with strict/misbehaving drivers that return valid buffer size ranges
3168 // but only accept the preferred buffer size as parameter for ASIOCreateBuffers. eg. Creatives ASIO driver
3169 // in that case, let's be naïve and try that instead
3170 *bufferSize = preferSize;
3171 stream_.bufferSize = *bufferSize;
3172 result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks );
3175 if ( result != ASE_OK ) {
3176 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") creating buffers.";
3177 errorText_ = errorStream_.str();
3180 buffersAllocated = true;
3181 stream_.state = STREAM_STOPPED;
3183 // Set flags for buffer conversion.
3184 stream_.doConvertBuffer[mode] = false;
3185 if ( stream_.userFormat != stream_.deviceFormat[mode] )
3186 stream_.doConvertBuffer[mode] = true;
3187 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
3188 stream_.nUserChannels[mode] > 1 )
3189 stream_.doConvertBuffer[mode] = true;
3191 // Allocate necessary internal buffers
3192 unsigned long bufferBytes;
3193 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
3194 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
3195 if ( stream_.userBuffer[mode] == NULL ) {
3196 errorText_ = "RtApiAsio::probeDeviceOpen: error allocating user buffer memory.";
3200 if ( stream_.doConvertBuffer[mode] ) {
3202 bool makeBuffer = true;
3203 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
3204 if ( isDuplexInput && stream_.deviceBuffer ) {
3205 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
3206 if ( bufferBytes <= bytesOut ) makeBuffer = false;
3210 bufferBytes *= *bufferSize;
3211 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
3212 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
3213 if ( stream_.deviceBuffer == NULL ) {
3214 errorText_ = "RtApiAsio::probeDeviceOpen: error allocating device buffer memory.";
3220 // Determine device latencies
3221 long inputLatency, outputLatency;
3222 result = ASIOGetLatencies( &inputLatency, &outputLatency );
3223 if ( result != ASE_OK ) {
3224 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting latency.";
3225 errorText_ = errorStream_.str();
3226 error( RtAudioError::WARNING); // warn but don't fail
3229 stream_.latency[0] = outputLatency;
3230 stream_.latency[1] = inputLatency;
3233 // Setup the buffer conversion information structure. We don't use
3234 // buffers to do channel offsets, so we override that parameter
3236 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );
3241 if ( !isDuplexInput ) {
3242 // the cleanup for error in the duplex input, is done by RtApi::openStream
3243 // So we clean up for single channel only
3245 if ( buffersAllocated )
3246 ASIODisposeBuffers();
3248 drivers.removeCurrentDriver();
3251 CloseHandle( handle->condition );
3252 if ( handle->bufferInfos )
3253 free( handle->bufferInfos );
3256 stream_.apiHandle = 0;
3260 if ( stream_.userBuffer[mode] ) {
3261 free( stream_.userBuffer[mode] );
3262 stream_.userBuffer[mode] = 0;
3265 if ( stream_.deviceBuffer ) {
3266 free( stream_.deviceBuffer );
3267 stream_.deviceBuffer = 0;
3272 }////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
3274 void RtApiAsio :: closeStream()
3276 if ( stream_.state == STREAM_CLOSED ) {
3277 errorText_ = "RtApiAsio::closeStream(): no open stream to close!";
3278 error( RtAudioError::WARNING );
3282 if ( stream_.state == STREAM_RUNNING ) {
3283 stream_.state = STREAM_STOPPED;
3286 ASIODisposeBuffers();
3287 drivers.removeCurrentDriver();
3289 AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
3291 CloseHandle( handle->condition );
3292 if ( handle->bufferInfos )
3293 free( handle->bufferInfos );
3295 stream_.apiHandle = 0;
3298 for ( int i=0; i<2; i++ ) {
3299 if ( stream_.userBuffer[i] ) {
3300 free( stream_.userBuffer[i] );
3301 stream_.userBuffer[i] = 0;
3305 if ( stream_.deviceBuffer ) {
3306 free( stream_.deviceBuffer );
3307 stream_.deviceBuffer = 0;
3310 stream_.mode = UNINITIALIZED;
3311 stream_.state = STREAM_CLOSED;
3314 bool stopThreadCalled = false;
3316 void RtApiAsio :: startStream()
3319 if ( stream_.state == STREAM_RUNNING ) {
3320 errorText_ = "RtApiAsio::startStream(): the stream is already running!";
3321 error( RtAudioError::WARNING );
3325 AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
3326 ASIOError result = ASIOStart();
3327 if ( result != ASE_OK ) {
3328 errorStream_ << "RtApiAsio::startStream: error (" << getAsioErrorString( result ) << ") starting device.";
3329 errorText_ = errorStream_.str();
3333 handle->drainCounter = 0;
3334 handle->internalDrain = false;
3335 ResetEvent( handle->condition );
3336 stream_.state = STREAM_RUNNING;
3340 stopThreadCalled = false;
3342 if ( result == ASE_OK ) return;
3343 error( RtAudioError::SYSTEM_ERROR );
3346 void RtApiAsio :: stopStream()
3349 if ( stream_.state == STREAM_STOPPED ) {
3350 errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!";
3351 error( RtAudioError::WARNING );
3355 AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
3356 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
3357 if ( handle->drainCounter == 0 ) {
3358 handle->drainCounter = 2;
3359 WaitForSingleObject( handle->condition, INFINITE ); // block until signaled
3363 stream_.state = STREAM_STOPPED;
3365 ASIOError result = ASIOStop();
3366 if ( result != ASE_OK ) {
3367 errorStream_ << "RtApiAsio::stopStream: error (" << getAsioErrorString( result ) << ") stopping device.";
3368 errorText_ = errorStream_.str();
3371 if ( result == ASE_OK ) return;
3372 error( RtAudioError::SYSTEM_ERROR );
3375 void RtApiAsio :: abortStream()
3378 if ( stream_.state == STREAM_STOPPED ) {
3379 errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!";
3380 error( RtAudioError::WARNING );
3384 // The following lines were commented-out because some behavior was
3385 // noted where the device buffers need to be zeroed to avoid
3386 // continuing sound, even when the device buffers are completely
3387 // disposed. So now, calling abort is the same as calling stop.
3388 // AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
3389 // handle->drainCounter = 2;
3393 // This function will be called by a spawned thread when the user
3394 // callback function signals that the stream should be stopped or
3395 // aborted. It is necessary to handle it this way because the
3396 // callbackEvent() function must return before the ASIOStop()
3397 // function will return.
3398 static unsigned __stdcall asioStopStream( void *ptr )
3400 CallbackInfo *info = (CallbackInfo *) ptr;
3401 RtApiAsio *object = (RtApiAsio *) info->object;
3403 object->stopStream();
3408 bool RtApiAsio :: callbackEvent( long bufferIndex )
3410 if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;
3411 if ( stream_.state == STREAM_CLOSED ) {
3412 errorText_ = "RtApiAsio::callbackEvent(): the stream is closed ... this shouldn't happen!";
3413 error( RtAudioError::WARNING );
3417 CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
3418 AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
3420 // Check if we were draining the stream and signal if finished.
3421 if ( handle->drainCounter > 3 ) {
3423 stream_.state = STREAM_STOPPING;
3424 if ( handle->internalDrain == false )
3425 SetEvent( handle->condition );
3426 else { // spawn a thread to stop the stream
3428 stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream,
3429 &stream_.callbackInfo, 0, &threadId );
3434 // Invoke user callback to get fresh output data UNLESS we are
3436 if ( handle->drainCounter == 0 ) {
3437 RtAudioCallback callback = (RtAudioCallback) info->callback;
3438 double streamTime = getStreamTime();
3439 RtAudioStreamStatus status = 0;
3440 if ( stream_.mode != INPUT && asioXRun == true ) {
3441 status |= RTAUDIO_OUTPUT_UNDERFLOW;
3444 if ( stream_.mode != OUTPUT && asioXRun == true ) {
3445 status |= RTAUDIO_INPUT_OVERFLOW;
3448 int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
3449 stream_.bufferSize, streamTime, status, info->userData );
3450 if ( cbReturnValue == 2 ) {
3451 stream_.state = STREAM_STOPPING;
3452 handle->drainCounter = 2;
3454 stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream,
3455 &stream_.callbackInfo, 0, &threadId );
3458 else if ( cbReturnValue == 1 ) {
3459 handle->drainCounter = 1;
3460 handle->internalDrain = true;
3464 unsigned int nChannels, bufferBytes, i, j;
3465 nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
3466 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
3468 bufferBytes = stream_.bufferSize * formatBytes( stream_.deviceFormat[0] );
3470 if ( handle->drainCounter > 1 ) { // write zeros to the output stream
3472 for ( i=0, j=0; i<nChannels; i++ ) {
3473 if ( handle->bufferInfos[i].isInput != ASIOTrue )
3474 memset( handle->bufferInfos[i].buffers[bufferIndex], 0, bufferBytes );
3478 else if ( stream_.doConvertBuffer[0] ) {
3480 convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
3481 if ( stream_.doByteSwap[0] )
3482 byteSwapBuffer( stream_.deviceBuffer,
3483 stream_.bufferSize * stream_.nDeviceChannels[0],
3484 stream_.deviceFormat[0] );
3486 for ( i=0, j=0; i<nChannels; i++ ) {
3487 if ( handle->bufferInfos[i].isInput != ASIOTrue )
3488 memcpy( handle->bufferInfos[i].buffers[bufferIndex],
3489 &stream_.deviceBuffer[j++*bufferBytes], bufferBytes );
3495 if ( stream_.doByteSwap[0] )
3496 byteSwapBuffer( stream_.userBuffer[0],
3497 stream_.bufferSize * stream_.nUserChannels[0],
3498 stream_.userFormat );
3500 for ( i=0, j=0; i<nChannels; i++ ) {
3501 if ( handle->bufferInfos[i].isInput != ASIOTrue )
3502 memcpy( handle->bufferInfos[i].buffers[bufferIndex],
3503 &stream_.userBuffer[0][bufferBytes*j++], bufferBytes );
3509 // Don't bother draining input
3510 if ( handle->drainCounter ) {
3511 handle->drainCounter++;
3515 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
3517 bufferBytes = stream_.bufferSize * formatBytes(stream_.deviceFormat[1]);
3519 if (stream_.doConvertBuffer[1]) {
3521 // Always interleave ASIO input data.
3522 for ( i=0, j=0; i<nChannels; i++ ) {
3523 if ( handle->bufferInfos[i].isInput == ASIOTrue )
3524 memcpy( &stream_.deviceBuffer[j++*bufferBytes],
3525 handle->bufferInfos[i].buffers[bufferIndex],
3529 if ( stream_.doByteSwap[1] )
3530 byteSwapBuffer( stream_.deviceBuffer,
3531 stream_.bufferSize * stream_.nDeviceChannels[1],
3532 stream_.deviceFormat[1] );
3533 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
3537 for ( i=0, j=0; i<nChannels; i++ ) {
3538 if ( handle->bufferInfos[i].isInput == ASIOTrue ) {
3539 memcpy( &stream_.userBuffer[1][bufferBytes*j++],
3540 handle->bufferInfos[i].buffers[bufferIndex],
3545 if ( stream_.doByteSwap[1] )
3546 byteSwapBuffer( stream_.userBuffer[1],
3547 stream_.bufferSize * stream_.nUserChannels[1],
3548 stream_.userFormat );
3553 // The following call was suggested by Malte Clasen. While the API
3554 // documentation indicates it should not be required, some device
3555 // drivers apparently do not function correctly without it.
3558 RtApi::tickStreamTime();
3562 static void sampleRateChanged( ASIOSampleRate sRate )
3564 // The ASIO documentation says that this usually only happens during
3565 // external sync. Audio processing is not stopped by the driver,
3566 // actual sample rate might not have even changed, maybe only the
3567 // sample rate status of an AES/EBU or S/PDIF digital input at the
3570 RtApi *object = (RtApi *) asioCallbackInfo->object;
3572 object->stopStream();
3574 catch ( RtAudioError &exception ) {
3575 std::cerr << "\nRtApiAsio: sampleRateChanged() error (" << exception.getMessage() << ")!\n" << std::endl;
3579 std::cerr << "\nRtApiAsio: driver reports sample rate changed to " << sRate << " ... stream stopped!!!\n" << std::endl;
3582 static long asioMessages( long selector, long value, void* /*message*/, double* /*opt*/ )
3586 switch( selector ) {
3587 case kAsioSelectorSupported:
3588 if ( value == kAsioResetRequest
3589 || value == kAsioEngineVersion
3590 || value == kAsioResyncRequest
3591 || value == kAsioLatenciesChanged
3592 // The following three were added for ASIO 2.0, you don't
3593 // necessarily have to support them.
3594 || value == kAsioSupportsTimeInfo
3595 || value == kAsioSupportsTimeCode
3596 || value == kAsioSupportsInputMonitor)
3599 case kAsioResetRequest:
3600 // Defer the task and perform the reset of the driver during the
3601 // next "safe" situation. You cannot reset the driver right now,
3602 // as this code is called from the driver. Reset the driver is
3603 // done by completely destruct is. I.e. ASIOStop(),
3604 // ASIODisposeBuffers(), Destruction Afterwards you initialize the
3606 std::cerr << "\nRtApiAsio: driver reset requested!!!" << std::endl;
3609 case kAsioResyncRequest:
3610 // This informs the application that the driver encountered some
3611 // non-fatal data loss. It is used for synchronization purposes
3612 // of different media. Added mainly to work around the Win16Mutex
3613 // problems in Windows 95/98 with the Windows Multimedia system,
3614 // which could lose data because the Mutex was held too long by
3615 // another thread. However a driver can issue it in other
3617 // std::cerr << "\nRtApiAsio: driver resync requested!!!" << std::endl;
3621 case kAsioLatenciesChanged:
3622 // This will inform the host application that the drivers were
3623 // latencies changed. Beware, it this does not mean that the
3624 // buffer sizes have changed! You might need to update internal
3626 std::cerr << "\nRtApiAsio: driver latency may have changed!!!" << std::endl;
3629 case kAsioEngineVersion:
3630 // Return the supported ASIO version of the host application. If
3631 // a host application does not implement this selector, ASIO 1.0
3632 // is assumed by the driver.
3635 case kAsioSupportsTimeInfo:
3636 // Informs the driver whether the
3637 // asioCallbacks.bufferSwitchTimeInfo() callback is supported.
3638 // For compatibility with ASIO 1.0 drivers the host application
3639 // should always support the "old" bufferSwitch method, too.
3642 case kAsioSupportsTimeCode:
3643 // Informs the driver whether application is interested in time
3644 // code info. If an application does not need to know about time
3645 // code, the driver has less work to do.
3652 static const char* getAsioErrorString( ASIOError result )
3660 static const Messages m[] =
3662 { ASE_NotPresent, "Hardware input or output is not present or available." },
3663 { ASE_HWMalfunction, "Hardware is malfunctioning." },
3664 { ASE_InvalidParameter, "Invalid input parameter." },
3665 { ASE_InvalidMode, "Invalid mode." },
3666 { ASE_SPNotAdvancing, "Sample position not advancing." },
3667 { ASE_NoClock, "Sample clock or rate cannot be determined or is not present." },
3668 { ASE_NoMemory, "Not enough memory to complete the request." }
3671 for ( unsigned int i = 0; i < sizeof(m)/sizeof(m[0]); ++i )
3672 if ( m[i].value == result ) return m[i].message;
3674 return "Unknown error.";
3677 //******************** End of __WINDOWS_ASIO__ *********************//
3681 #if defined(__WINDOWS_WASAPI__) // Windows WASAPI API
3683 // Authored by Marcus Tomlinson <themarcustomlinson@gmail.com>, April 2014
3684 // - Introduces support for the Windows WASAPI API
3685 // - Aims to deliver bit streams to and from hardware at the lowest possible latency, via the absolute minimum buffer sizes required
3686 // - Provides flexible stream configuration to an otherwise strict and inflexible WASAPI interface
3687 // - Includes automatic internal conversion of sample rate and buffer size between hardware and the user
3692 #include <audioclient.h>
3694 #include <mmdeviceapi.h>
3695 #include <functiondiscoverykeys_devpkey.h>
3698 #include <mferror.h>
3700 #include <Wmcodecdsp.h>
3702 #pragma comment( lib, "mfplat.lib" )
3703 #pragma comment( lib, "mfuuid.lib" )
3704 #pragma comment( lib, "wmcodecdspuuid" )
3706 //=============================================================================
3708 #define SAFE_RELEASE( objectPtr )\
3711 objectPtr->Release();\
3715 typedef HANDLE ( __stdcall *TAvSetMmThreadCharacteristicsPtr )( LPCWSTR TaskName, LPDWORD TaskIndex );
3717 //-----------------------------------------------------------------------------
3719 // WASAPI dictates stream sample rate, format, channel count, and in some cases, buffer size.
3720 // Therefore we must perform all necessary conversions to user buffers in order to satisfy these
3721 // requirements. WasapiBuffer ring buffers are used between HwIn->UserIn and UserOut->HwOut to
3722 // provide intermediate storage for read / write synchronization.
3736 // sets the length of the internal ring buffer
3737 void setBufferSize( unsigned int bufferSize, unsigned int formatBytes ) {
3740 buffer_ = ( char* ) calloc( bufferSize, formatBytes );
3742 bufferSize_ = bufferSize;
3747 // attempt to push a buffer into the ring buffer at the current "in" index
3748 bool pushBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format )
3750 if ( !buffer || // incoming buffer is NULL
3751 bufferSize == 0 || // incoming buffer has no data
3752 bufferSize > bufferSize_ ) // incoming buffer too large
3757 unsigned int relOutIndex = outIndex_;
3758 unsigned int inIndexEnd = inIndex_ + bufferSize;
3759 if ( relOutIndex < inIndex_ && inIndexEnd >= bufferSize_ ) {
3760 relOutIndex += bufferSize_;
3763 // "in" index can end on the "out" index but cannot begin at it
3764 if ( inIndex_ <= relOutIndex && inIndexEnd > relOutIndex ) {
3765 return false; // not enough space between "in" index and "out" index
3768 // copy buffer from external to internal
3769 int fromZeroSize = inIndex_ + bufferSize - bufferSize_;
3770 fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize;
3771 int fromInSize = bufferSize - fromZeroSize;
3776 memcpy( &( ( char* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( char ) );
3777 memcpy( buffer_, &( ( char* ) buffer )[fromInSize], fromZeroSize * sizeof( char ) );
3779 case RTAUDIO_SINT16:
3780 memcpy( &( ( short* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( short ) );
3781 memcpy( buffer_, &( ( short* ) buffer )[fromInSize], fromZeroSize * sizeof( short ) );
3783 case RTAUDIO_SINT24:
3784 memcpy( &( ( S24* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( S24 ) );
3785 memcpy( buffer_, &( ( S24* ) buffer )[fromInSize], fromZeroSize * sizeof( S24 ) );
3787 case RTAUDIO_SINT32:
3788 memcpy( &( ( int* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( int ) );
3789 memcpy( buffer_, &( ( int* ) buffer )[fromInSize], fromZeroSize * sizeof( int ) );
3791 case RTAUDIO_FLOAT32:
3792 memcpy( &( ( float* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( float ) );
3793 memcpy( buffer_, &( ( float* ) buffer )[fromInSize], fromZeroSize * sizeof( float ) );
3795 case RTAUDIO_FLOAT64:
3796 memcpy( &( ( double* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( double ) );
3797 memcpy( buffer_, &( ( double* ) buffer )[fromInSize], fromZeroSize * sizeof( double ) );
3801 // update "in" index
3802 inIndex_ += bufferSize;
3803 inIndex_ %= bufferSize_;
3808 // attempt to pull a buffer from the ring buffer from the current "out" index
3809 bool pullBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format )
3811 if ( !buffer || // incoming buffer is NULL
3812 bufferSize == 0 || // incoming buffer has no data
3813 bufferSize > bufferSize_ ) // incoming buffer too large
3818 unsigned int relInIndex = inIndex_;
3819 unsigned int outIndexEnd = outIndex_ + bufferSize;
3820 if ( relInIndex < outIndex_ && outIndexEnd >= bufferSize_ ) {
3821 relInIndex += bufferSize_;
3824 // "out" index can begin at and end on the "in" index
3825 if ( outIndex_ < relInIndex && outIndexEnd > relInIndex ) {
3826 return false; // not enough space between "out" index and "in" index
3829 // copy buffer from internal to external
3830 int fromZeroSize = outIndex_ + bufferSize - bufferSize_;
3831 fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize;
3832 int fromOutSize = bufferSize - fromZeroSize;
3837 memcpy( buffer, &( ( char* ) buffer_ )[outIndex_], fromOutSize * sizeof( char ) );
3838 memcpy( &( ( char* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( char ) );
3840 case RTAUDIO_SINT16:
3841 memcpy( buffer, &( ( short* ) buffer_ )[outIndex_], fromOutSize * sizeof( short ) );
3842 memcpy( &( ( short* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( short ) );
3844 case RTAUDIO_SINT24:
3845 memcpy( buffer, &( ( S24* ) buffer_ )[outIndex_], fromOutSize * sizeof( S24 ) );
3846 memcpy( &( ( S24* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( S24 ) );
3848 case RTAUDIO_SINT32:
3849 memcpy( buffer, &( ( int* ) buffer_ )[outIndex_], fromOutSize * sizeof( int ) );
3850 memcpy( &( ( int* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( int ) );
3852 case RTAUDIO_FLOAT32:
3853 memcpy( buffer, &( ( float* ) buffer_ )[outIndex_], fromOutSize * sizeof( float ) );
3854 memcpy( &( ( float* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( float ) );
3856 case RTAUDIO_FLOAT64:
3857 memcpy( buffer, &( ( double* ) buffer_ )[outIndex_], fromOutSize * sizeof( double ) );
3858 memcpy( &( ( double* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( double ) );
3862 // update "out" index
3863 outIndex_ += bufferSize;
3864 outIndex_ %= bufferSize_;
3871 unsigned int bufferSize_;
3872 unsigned int inIndex_;
3873 unsigned int outIndex_;
3876 //-----------------------------------------------------------------------------
3878 // In order to satisfy WASAPI's buffer requirements, we need a means of converting sample rate
3879 // between HW and the user. The WasapiResampler class is used to perform this conversion between
3880 // HwIn->UserIn and UserOut->HwOut during the stream callback loop.
3881 class WasapiResampler
3884 WasapiResampler( bool isFloat, unsigned int bitsPerSample, unsigned int channelCount,
3885 unsigned int inSampleRate, unsigned int outSampleRate )
3886 : _bytesPerSample( bitsPerSample / 8 )
3887 , _channelCount( channelCount )
3888 , _sampleRatio( ( float ) outSampleRate / inSampleRate )
3889 , _transformUnk( NULL )
3890 , _transform( NULL )
3891 , _resamplerProps( NULL )
3892 , _mediaType( NULL )
3893 , _inputMediaType( NULL )
3894 , _outputMediaType( NULL )
3896 // 1. Initialization
3898 MFStartup( MF_VERSION, MFSTARTUP_NOSOCKET );
3900 // 2. Create Resampler Transform Object
3902 CoCreateInstance( CLSID_CResamplerMediaObject, NULL, CLSCTX_INPROC_SERVER,
3903 IID_IUnknown, ( void** ) &_transformUnk );
3905 _transformUnk->QueryInterface( IID_PPV_ARGS( &_transform ) );
3907 _transformUnk->QueryInterface( IID_PPV_ARGS( &_resamplerProps ) );
3908 _resamplerProps->SetHalfFilterLength( 60 ); // best conversion quality
3910 // 3. Specify input / output format
3912 MFCreateMediaType( &_mediaType );
3913 _mediaType->SetGUID( MF_MT_MAJOR_TYPE, MFMediaType_Audio );
3914 _mediaType->SetGUID( MF_MT_SUBTYPE, isFloat ? MFAudioFormat_Float : MFAudioFormat_PCM );
3915 _mediaType->SetUINT32( MF_MT_AUDIO_NUM_CHANNELS, channelCount );
3916 _mediaType->SetUINT32( MF_MT_AUDIO_SAMPLES_PER_SECOND, inSampleRate );
3917 _mediaType->SetUINT32( MF_MT_AUDIO_BLOCK_ALIGNMENT, _bytesPerSample * channelCount );
3918 _mediaType->SetUINT32( MF_MT_AUDIO_AVG_BYTES_PER_SECOND, _bytesPerSample * channelCount * inSampleRate );
3919 _mediaType->SetUINT32( MF_MT_AUDIO_BITS_PER_SAMPLE, bitsPerSample );
3920 _mediaType->SetUINT32( MF_MT_ALL_SAMPLES_INDEPENDENT, TRUE );
3922 MFCreateMediaType( &_inputMediaType );
3923 _mediaType->CopyAllItems( _inputMediaType );
3925 _transform->SetInputType( 0, _inputMediaType, 0 );
3927 MFCreateMediaType( &_outputMediaType );
3928 _mediaType->CopyAllItems( _outputMediaType );
3930 _outputMediaType->SetUINT32( MF_MT_AUDIO_SAMPLES_PER_SECOND, outSampleRate );
3931 _outputMediaType->SetUINT32( MF_MT_AUDIO_AVG_BYTES_PER_SECOND, _bytesPerSample * channelCount * outSampleRate );
3933 _transform->SetOutputType( 0, _outputMediaType, 0 );
3935 // 4. Send stream start messages to Resampler
3937 _transform->ProcessMessage( MFT_MESSAGE_COMMAND_FLUSH, NULL );
3938 _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_BEGIN_STREAMING, NULL );
3939 _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_START_OF_STREAM, NULL );
3944 // 8. Send stream stop messages to Resampler
3946 _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_END_OF_STREAM, NULL );
3947 _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_END_STREAMING, NULL );
3953 SAFE_RELEASE( _transformUnk );
3954 SAFE_RELEASE( _transform );
3955 SAFE_RELEASE( _resamplerProps );
3956 SAFE_RELEASE( _mediaType );
3957 SAFE_RELEASE( _inputMediaType );
3958 SAFE_RELEASE( _outputMediaType );
3961 void Convert( char* outBuffer, const char* inBuffer, unsigned int inSampleCount, unsigned int& outSampleCount )
3963 unsigned int inputBufferSize = _bytesPerSample * _channelCount * inSampleCount;
3964 if ( _sampleRatio == 1 )
3966 // no sample rate conversion required
3967 memcpy( outBuffer, inBuffer, inputBufferSize );
3968 outSampleCount = inSampleCount;
3972 unsigned int outputBufferSize = ( unsigned int ) ceilf( inputBufferSize * _sampleRatio ) + ( _bytesPerSample * _channelCount );
3974 IMFMediaBuffer* rInBuffer;
3975 IMFSample* rInSample;
3976 BYTE* rInByteBuffer = NULL;
3978 // 5. Create Sample object from input data
3980 MFCreateMemoryBuffer( inputBufferSize, &rInBuffer );
3982 rInBuffer->Lock( &rInByteBuffer, NULL, NULL );
3983 memcpy( rInByteBuffer, inBuffer, inputBufferSize );
3984 rInBuffer->Unlock();
3985 rInByteBuffer = NULL;
3987 rInBuffer->SetCurrentLength( inputBufferSize );
3989 MFCreateSample( &rInSample );
3990 rInSample->AddBuffer( rInBuffer );
3992 // 6. Pass input data to Resampler
3994 _transform->ProcessInput( 0, rInSample, 0 );
3996 SAFE_RELEASE( rInBuffer );
3997 SAFE_RELEASE( rInSample );
3999 // 7. Perform sample rate conversion
4001 IMFMediaBuffer* rOutBuffer = NULL;
4002 BYTE* rOutByteBuffer = NULL;
4004 MFT_OUTPUT_DATA_BUFFER rOutDataBuffer;
4006 DWORD rBytes = outputBufferSize; // maximum bytes accepted per ProcessOutput
4008 // 7.1 Create Sample object for output data
4010 memset( &rOutDataBuffer, 0, sizeof rOutDataBuffer );
4011 MFCreateSample( &( rOutDataBuffer.pSample ) );
4012 MFCreateMemoryBuffer( rBytes, &rOutBuffer );
4013 rOutDataBuffer.pSample->AddBuffer( rOutBuffer );
4014 rOutDataBuffer.dwStreamID = 0;
4015 rOutDataBuffer.dwStatus = 0;
4016 rOutDataBuffer.pEvents = NULL;
4018 // 7.2 Get output data from Resampler
4020 if ( _transform->ProcessOutput( 0, 1, &rOutDataBuffer, &rStatus ) == MF_E_TRANSFORM_NEED_MORE_INPUT )
4023 SAFE_RELEASE( rOutBuffer );
4024 SAFE_RELEASE( rOutDataBuffer.pSample );
4028 // 7.3 Write output data to outBuffer
4030 SAFE_RELEASE( rOutBuffer );
4031 rOutDataBuffer.pSample->ConvertToContiguousBuffer( &rOutBuffer );
4032 rOutBuffer->GetCurrentLength( &rBytes );
4034 rOutBuffer->Lock( &rOutByteBuffer, NULL, NULL );
4035 memcpy( outBuffer, rOutByteBuffer, rBytes );
4036 rOutBuffer->Unlock();
4037 rOutByteBuffer = NULL;
4039 outSampleCount = rBytes / _bytesPerSample / _channelCount;
4040 SAFE_RELEASE( rOutBuffer );
4041 SAFE_RELEASE( rOutDataBuffer.pSample );
4045 unsigned int _bytesPerSample;
4046 unsigned int _channelCount;
4049 IUnknown* _transformUnk;
4050 IMFTransform* _transform;
4051 IWMResamplerProps* _resamplerProps;
4052 IMFMediaType* _mediaType;
4053 IMFMediaType* _inputMediaType;
4054 IMFMediaType* _outputMediaType;
4057 //-----------------------------------------------------------------------------
4059 // A structure to hold various information related to the WASAPI implementation.
4062 IAudioClient* captureAudioClient;
4063 IAudioClient* renderAudioClient;
4064 IAudioCaptureClient* captureClient;
4065 IAudioRenderClient* renderClient;
4066 HANDLE captureEvent;
4070 : captureAudioClient( NULL ),
4071 renderAudioClient( NULL ),
4072 captureClient( NULL ),
4073 renderClient( NULL ),
4074 captureEvent( NULL ),
4075 renderEvent( NULL ) {}
4078 //=============================================================================
4080 RtApiWasapi::RtApiWasapi()
4081 : coInitialized_( false ), deviceEnumerator_( NULL )
4083 // WASAPI can run either apartment or multi-threaded
4084 HRESULT hr = CoInitialize( NULL );
4085 if ( !FAILED( hr ) )
4086 coInitialized_ = true;
4088 // Instantiate device enumerator
4089 hr = CoCreateInstance( __uuidof( MMDeviceEnumerator ), NULL,
4090 CLSCTX_ALL, __uuidof( IMMDeviceEnumerator ),
4091 ( void** ) &deviceEnumerator_ );
4093 if ( FAILED( hr ) ) {
4094 errorText_ = "RtApiWasapi::RtApiWasapi: Unable to instantiate device enumerator";
4095 error( RtAudioError::DRIVER_ERROR );
4099 //-----------------------------------------------------------------------------
4101 RtApiWasapi::~RtApiWasapi()
4103 if ( stream_.state != STREAM_CLOSED )
4106 SAFE_RELEASE( deviceEnumerator_ );
4108 // If this object previously called CoInitialize()
4109 if ( coInitialized_ )
4113 //=============================================================================
4115 unsigned int RtApiWasapi::getDeviceCount( void )
4117 unsigned int captureDeviceCount = 0;
4118 unsigned int renderDeviceCount = 0;
4120 IMMDeviceCollection* captureDevices = NULL;
4121 IMMDeviceCollection* renderDevices = NULL;
4123 // Count capture devices
4125 HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
4126 if ( FAILED( hr ) ) {
4127 errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device collection.";
4131 hr = captureDevices->GetCount( &captureDeviceCount );
4132 if ( FAILED( hr ) ) {
4133 errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device count.";
4137 // Count render devices
4138 hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
4139 if ( FAILED( hr ) ) {
4140 errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device collection.";
4144 hr = renderDevices->GetCount( &renderDeviceCount );
4145 if ( FAILED( hr ) ) {
4146 errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device count.";
4151 // release all references
4152 SAFE_RELEASE( captureDevices );
4153 SAFE_RELEASE( renderDevices );
4155 if ( errorText_.empty() )
4156 return captureDeviceCount + renderDeviceCount;
4158 error( RtAudioError::DRIVER_ERROR );
4162 //-----------------------------------------------------------------------------
4164 RtAudio::DeviceInfo RtApiWasapi::getDeviceInfo( unsigned int device )
4166 RtAudio::DeviceInfo info;
4167 unsigned int captureDeviceCount = 0;
4168 unsigned int renderDeviceCount = 0;
4169 std::string defaultDeviceName;
4170 bool isCaptureDevice = false;
4172 PROPVARIANT deviceNameProp;
4173 PROPVARIANT defaultDeviceNameProp;
4175 IMMDeviceCollection* captureDevices = NULL;
4176 IMMDeviceCollection* renderDevices = NULL;
4177 IMMDevice* devicePtr = NULL;
4178 IMMDevice* defaultDevicePtr = NULL;
4179 IAudioClient* audioClient = NULL;
4180 IPropertyStore* devicePropStore = NULL;
4181 IPropertyStore* defaultDevicePropStore = NULL;
4183 WAVEFORMATEX* deviceFormat = NULL;
4184 WAVEFORMATEX* closestMatchFormat = NULL;
4187 info.probed = false;
4189 // Count capture devices
4191 RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
4192 HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
4193 if ( FAILED( hr ) ) {
4194 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device collection.";
4198 hr = captureDevices->GetCount( &captureDeviceCount );
4199 if ( FAILED( hr ) ) {
4200 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device count.";
4204 // Count render devices
4205 hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
4206 if ( FAILED( hr ) ) {
4207 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device collection.";
4211 hr = renderDevices->GetCount( &renderDeviceCount );
4212 if ( FAILED( hr ) ) {
4213 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device count.";
4217 // validate device index
4218 if ( device >= captureDeviceCount + renderDeviceCount ) {
4219 errorText_ = "RtApiWasapi::getDeviceInfo: Invalid device index.";
4220 errorType = RtAudioError::INVALID_USE;
4224 // determine whether index falls within capture or render devices
4225 if ( device >= renderDeviceCount ) {
4226 hr = captureDevices->Item( device - renderDeviceCount, &devicePtr );
4227 if ( FAILED( hr ) ) {
4228 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device handle.";
4231 isCaptureDevice = true;
4234 hr = renderDevices->Item( device, &devicePtr );
4235 if ( FAILED( hr ) ) {
4236 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device handle.";
4239 isCaptureDevice = false;
4242 // get default device name
4243 if ( isCaptureDevice ) {
4244 hr = deviceEnumerator_->GetDefaultAudioEndpoint( eCapture, eConsole, &defaultDevicePtr );
4245 if ( FAILED( hr ) ) {
4246 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default capture device handle.";
4251 hr = deviceEnumerator_->GetDefaultAudioEndpoint( eRender, eConsole, &defaultDevicePtr );
4252 if ( FAILED( hr ) ) {
4253 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default render device handle.";
4258 hr = defaultDevicePtr->OpenPropertyStore( STGM_READ, &defaultDevicePropStore );
4259 if ( FAILED( hr ) ) {
4260 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open default device property store.";
4263 PropVariantInit( &defaultDeviceNameProp );
4265 hr = defaultDevicePropStore->GetValue( PKEY_Device_FriendlyName, &defaultDeviceNameProp );
4266 if ( FAILED( hr ) ) {
4267 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default device property: PKEY_Device_FriendlyName.";
4271 defaultDeviceName = convertCharPointerToStdString(defaultDeviceNameProp.pwszVal);
4274 hr = devicePtr->OpenPropertyStore( STGM_READ, &devicePropStore );
4275 if ( FAILED( hr ) ) {
4276 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open device property store.";
4280 PropVariantInit( &deviceNameProp );
4282 hr = devicePropStore->GetValue( PKEY_Device_FriendlyName, &deviceNameProp );
4283 if ( FAILED( hr ) ) {
4284 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device property: PKEY_Device_FriendlyName.";
4288 info.name =convertCharPointerToStdString(deviceNameProp.pwszVal);
4291 if ( isCaptureDevice ) {
4292 info.isDefaultInput = info.name == defaultDeviceName;
4293 info.isDefaultOutput = false;
4296 info.isDefaultInput = false;
4297 info.isDefaultOutput = info.name == defaultDeviceName;
4301 hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, NULL, ( void** ) &audioClient );
4302 if ( FAILED( hr ) ) {
4303 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device audio client.";
4307 hr = audioClient->GetMixFormat( &deviceFormat );
4308 if ( FAILED( hr ) ) {
4309 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device mix format.";
4313 if ( isCaptureDevice ) {
4314 info.inputChannels = deviceFormat->nChannels;
4315 info.outputChannels = 0;
4316 info.duplexChannels = 0;
4319 info.inputChannels = 0;
4320 info.outputChannels = deviceFormat->nChannels;
4321 info.duplexChannels = 0;
4325 info.sampleRates.clear();
4327 // allow support for all sample rates as we have a built-in sample rate converter
4328 for ( unsigned int i = 0; i < MAX_SAMPLE_RATES; i++ ) {
4329 info.sampleRates.push_back( SAMPLE_RATES[i] );
4331 info.preferredSampleRate = deviceFormat->nSamplesPerSec;
4334 info.nativeFormats = 0;
4336 if ( deviceFormat->wFormatTag == WAVE_FORMAT_IEEE_FLOAT ||
4337 ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
4338 ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_IEEE_FLOAT ) )
4340 if ( deviceFormat->wBitsPerSample == 32 ) {
4341 info.nativeFormats |= RTAUDIO_FLOAT32;
4343 else if ( deviceFormat->wBitsPerSample == 64 ) {
4344 info.nativeFormats |= RTAUDIO_FLOAT64;
4347 else if ( deviceFormat->wFormatTag == WAVE_FORMAT_PCM ||
4348 ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
4349 ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_PCM ) )
4351 if ( deviceFormat->wBitsPerSample == 8 ) {
4352 info.nativeFormats |= RTAUDIO_SINT8;
4354 else if ( deviceFormat->wBitsPerSample == 16 ) {
4355 info.nativeFormats |= RTAUDIO_SINT16;
4357 else if ( deviceFormat->wBitsPerSample == 24 ) {
4358 info.nativeFormats |= RTAUDIO_SINT24;
4360 else if ( deviceFormat->wBitsPerSample == 32 ) {
4361 info.nativeFormats |= RTAUDIO_SINT32;
4369 // release all references
4370 PropVariantClear( &deviceNameProp );
4371 PropVariantClear( &defaultDeviceNameProp );
4373 SAFE_RELEASE( captureDevices );
4374 SAFE_RELEASE( renderDevices );
4375 SAFE_RELEASE( devicePtr );
4376 SAFE_RELEASE( defaultDevicePtr );
4377 SAFE_RELEASE( audioClient );
4378 SAFE_RELEASE( devicePropStore );
4379 SAFE_RELEASE( defaultDevicePropStore );
4381 CoTaskMemFree( deviceFormat );
4382 CoTaskMemFree( closestMatchFormat );
4384 if ( !errorText_.empty() )
4389 //-----------------------------------------------------------------------------
4391 unsigned int RtApiWasapi::getDefaultOutputDevice( void )
4393 for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {
4394 if ( getDeviceInfo( i ).isDefaultOutput ) {
4402 //-----------------------------------------------------------------------------
4404 unsigned int RtApiWasapi::getDefaultInputDevice( void )
4406 for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {
4407 if ( getDeviceInfo( i ).isDefaultInput ) {
4415 //-----------------------------------------------------------------------------
4417 void RtApiWasapi::closeStream( void )
4419 if ( stream_.state == STREAM_CLOSED ) {
4420 errorText_ = "RtApiWasapi::closeStream: No open stream to close.";
4421 error( RtAudioError::WARNING );
4425 if ( stream_.state != STREAM_STOPPED )
4428 // clean up stream memory
4429 SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient )
4430 SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient )
4432 SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureClient )
4433 SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderClient )
4435 if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent )
4436 CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent );
4438 if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent )
4439 CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent );
4441 delete ( WasapiHandle* ) stream_.apiHandle;
4442 stream_.apiHandle = NULL;
4444 for ( int i = 0; i < 2; i++ ) {
4445 if ( stream_.userBuffer[i] ) {
4446 free( stream_.userBuffer[i] );
4447 stream_.userBuffer[i] = 0;
4451 if ( stream_.deviceBuffer ) {
4452 free( stream_.deviceBuffer );
4453 stream_.deviceBuffer = 0;
4456 // update stream state
4457 stream_.state = STREAM_CLOSED;
4460 //-----------------------------------------------------------------------------
4462 void RtApiWasapi::startStream( void )
4466 if ( stream_.state == STREAM_RUNNING ) {
4467 errorText_ = "RtApiWasapi::startStream: The stream is already running.";
4468 error( RtAudioError::WARNING );
4472 // update stream state
4473 stream_.state = STREAM_RUNNING;
4475 // create WASAPI stream thread
4476 stream_.callbackInfo.thread = ( ThreadHandle ) CreateThread( NULL, 0, runWasapiThread, this, CREATE_SUSPENDED, NULL );
4478 if ( !stream_.callbackInfo.thread ) {
4479 errorText_ = "RtApiWasapi::startStream: Unable to instantiate callback thread.";
4480 error( RtAudioError::THREAD_ERROR );
4483 SetThreadPriority( ( void* ) stream_.callbackInfo.thread, stream_.callbackInfo.priority );
4484 ResumeThread( ( void* ) stream_.callbackInfo.thread );
4488 //-----------------------------------------------------------------------------
4490 void RtApiWasapi::stopStream( void )
4494 if ( stream_.state == STREAM_STOPPED ) {
4495 errorText_ = "RtApiWasapi::stopStream: The stream is already stopped.";
4496 error( RtAudioError::WARNING );
4500 // inform stream thread by setting stream state to STREAM_STOPPING
4501 stream_.state = STREAM_STOPPING;
4503 // wait until stream thread is stopped
4504 while( stream_.state != STREAM_STOPPED ) {
4508 // Wait for the last buffer to play before stopping.
4509 Sleep( 1000 * stream_.bufferSize / stream_.sampleRate );
4511 // stop capture client if applicable
4512 if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) {
4513 HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop();
4514 if ( FAILED( hr ) ) {
4515 errorText_ = "RtApiWasapi::stopStream: Unable to stop capture stream.";
4516 error( RtAudioError::DRIVER_ERROR );
4521 // stop render client if applicable
4522 if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) {
4523 HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop();
4524 if ( FAILED( hr ) ) {
4525 errorText_ = "RtApiWasapi::stopStream: Unable to stop render stream.";
4526 error( RtAudioError::DRIVER_ERROR );
4531 // close thread handle
4532 if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
4533 errorText_ = "RtApiWasapi::stopStream: Unable to close callback thread.";
4534 error( RtAudioError::THREAD_ERROR );
4538 stream_.callbackInfo.thread = (ThreadHandle) NULL;
4541 //-----------------------------------------------------------------------------
4543 void RtApiWasapi::abortStream( void )
4547 if ( stream_.state == STREAM_STOPPED ) {
4548 errorText_ = "RtApiWasapi::abortStream: The stream is already stopped.";
4549 error( RtAudioError::WARNING );
4553 // inform stream thread by setting stream state to STREAM_STOPPING
4554 stream_.state = STREAM_STOPPING;
4556 // wait until stream thread is stopped
4557 while ( stream_.state != STREAM_STOPPED ) {
4561 // stop capture client if applicable
4562 if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) {
4563 HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop();
4564 if ( FAILED( hr ) ) {
4565 errorText_ = "RtApiWasapi::abortStream: Unable to stop capture stream.";
4566 error( RtAudioError::DRIVER_ERROR );
4571 // stop render client if applicable
4572 if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) {
4573 HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop();
4574 if ( FAILED( hr ) ) {
4575 errorText_ = "RtApiWasapi::abortStream: Unable to stop render stream.";
4576 error( RtAudioError::DRIVER_ERROR );
4581 // close thread handle
4582 if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
4583 errorText_ = "RtApiWasapi::abortStream: Unable to close callback thread.";
4584 error( RtAudioError::THREAD_ERROR );
4588 stream_.callbackInfo.thread = (ThreadHandle) NULL;
4591 //-----------------------------------------------------------------------------
4593 bool RtApiWasapi::probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
4594 unsigned int firstChannel, unsigned int sampleRate,
4595 RtAudioFormat format, unsigned int* bufferSize,
4596 RtAudio::StreamOptions* options )
4598 bool methodResult = FAILURE;
4599 unsigned int captureDeviceCount = 0;
4600 unsigned int renderDeviceCount = 0;
4602 IMMDeviceCollection* captureDevices = NULL;
4603 IMMDeviceCollection* renderDevices = NULL;
4604 IMMDevice* devicePtr = NULL;
4605 WAVEFORMATEX* deviceFormat = NULL;
4606 unsigned int bufferBytes;
4607 stream_.state = STREAM_STOPPED;
4609 // create API Handle if not already created
4610 if ( !stream_.apiHandle )
4611 stream_.apiHandle = ( void* ) new WasapiHandle();
4613 // Count capture devices
4615 RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
4616 HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
4617 if ( FAILED( hr ) ) {
4618 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device collection.";
4622 hr = captureDevices->GetCount( &captureDeviceCount );
4623 if ( FAILED( hr ) ) {
4624 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device count.";
4628 // Count render devices
4629 hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
4630 if ( FAILED( hr ) ) {
4631 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device collection.";
4635 hr = renderDevices->GetCount( &renderDeviceCount );
4636 if ( FAILED( hr ) ) {
4637 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device count.";
4641 // validate device index
4642 if ( device >= captureDeviceCount + renderDeviceCount ) {
4643 errorType = RtAudioError::INVALID_USE;
4644 errorText_ = "RtApiWasapi::probeDeviceOpen: Invalid device index.";
4648 // determine whether index falls within capture or render devices
4649 if ( device >= renderDeviceCount ) {
4650 if ( mode != INPUT ) {
4651 errorType = RtAudioError::INVALID_USE;
4652 errorText_ = "RtApiWasapi::probeDeviceOpen: Capture device selected as output device.";
4656 // retrieve captureAudioClient from devicePtr
4657 IAudioClient*& captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
4659 hr = captureDevices->Item( device - renderDeviceCount, &devicePtr );
4660 if ( FAILED( hr ) ) {
4661 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device handle.";
4665 hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
4666 NULL, ( void** ) &captureAudioClient );
4667 if ( FAILED( hr ) ) {
4668 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client.";
4672 hr = captureAudioClient->GetMixFormat( &deviceFormat );
4673 if ( FAILED( hr ) ) {
4674 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format.";
4678 stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
4679 captureAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
4682 if ( mode != OUTPUT ) {
4683 errorType = RtAudioError::INVALID_USE;
4684 errorText_ = "RtApiWasapi::probeDeviceOpen: Render device selected as input device.";
4688 // retrieve renderAudioClient from devicePtr
4689 IAudioClient*& renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
4691 hr = renderDevices->Item( device, &devicePtr );
4692 if ( FAILED( hr ) ) {
4693 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device handle.";
4697 hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
4698 NULL, ( void** ) &renderAudioClient );
4699 if ( FAILED( hr ) ) {
4700 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client.";
4704 hr = renderAudioClient->GetMixFormat( &deviceFormat );
4705 if ( FAILED( hr ) ) {
4706 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format.";
4710 stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
4711 renderAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
4715 if ( ( stream_.mode == OUTPUT && mode == INPUT ) ||
4716 ( stream_.mode == INPUT && mode == OUTPUT ) ) {
4717 stream_.mode = DUPLEX;
4720 stream_.mode = mode;
4723 stream_.device[mode] = device;
4724 stream_.doByteSwap[mode] = false;
4725 stream_.sampleRate = sampleRate;
4726 stream_.bufferSize = *bufferSize;
4727 stream_.nBuffers = 1;
4728 stream_.nUserChannels[mode] = channels;
4729 stream_.channelOffset[mode] = firstChannel;
4730 stream_.userFormat = format;
4731 stream_.deviceFormat[mode] = getDeviceInfo( device ).nativeFormats;
4733 if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
4734 stream_.userInterleaved = false;
4736 stream_.userInterleaved = true;
4737 stream_.deviceInterleaved[mode] = true;
4739 // Set flags for buffer conversion.
4740 stream_.doConvertBuffer[mode] = false;
4741 if ( stream_.userFormat != stream_.deviceFormat[mode] ||
4742 stream_.nUserChannels != stream_.nDeviceChannels )
4743 stream_.doConvertBuffer[mode] = true;
4744 else if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
4745 stream_.nUserChannels[mode] > 1 )
4746 stream_.doConvertBuffer[mode] = true;
4748 if ( stream_.doConvertBuffer[mode] )
4749 setConvertInfo( mode, 0 );
4751 // Allocate necessary internal buffers
4752 bufferBytes = stream_.nUserChannels[mode] * stream_.bufferSize * formatBytes( stream_.userFormat );
4754 stream_.userBuffer[mode] = ( char* ) calloc( bufferBytes, 1 );
4755 if ( !stream_.userBuffer[mode] ) {
4756 errorType = RtAudioError::MEMORY_ERROR;
4757 errorText_ = "RtApiWasapi::probeDeviceOpen: Error allocating user buffer memory.";
4761 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME )
4762 stream_.callbackInfo.priority = 15;
4764 stream_.callbackInfo.priority = 0;
4766 ///! TODO: RTAUDIO_MINIMIZE_LATENCY // Provide stream buffers directly to callback
4767 ///! TODO: RTAUDIO_HOG_DEVICE // Exclusive mode
4769 methodResult = SUCCESS;
4773 SAFE_RELEASE( captureDevices );
4774 SAFE_RELEASE( renderDevices );
4775 SAFE_RELEASE( devicePtr );
4776 CoTaskMemFree( deviceFormat );
4778 // if method failed, close the stream
4779 if ( methodResult == FAILURE )
4782 if ( !errorText_.empty() )
4784 return methodResult;
4787 //=============================================================================
4789 DWORD WINAPI RtApiWasapi::runWasapiThread( void* wasapiPtr )
4792 ( ( RtApiWasapi* ) wasapiPtr )->wasapiThread();
4797 DWORD WINAPI RtApiWasapi::stopWasapiThread( void* wasapiPtr )
4800 ( ( RtApiWasapi* ) wasapiPtr )->stopStream();
4805 DWORD WINAPI RtApiWasapi::abortWasapiThread( void* wasapiPtr )
4808 ( ( RtApiWasapi* ) wasapiPtr )->abortStream();
4813 //-----------------------------------------------------------------------------
4815 void RtApiWasapi::wasapiThread()
4817 // as this is a new thread, we must CoInitialize it
4818 CoInitialize( NULL );
4822 IAudioClient* captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
4823 IAudioClient* renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
4824 IAudioCaptureClient* captureClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureClient;
4825 IAudioRenderClient* renderClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderClient;
4826 HANDLE captureEvent = ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent;
4827 HANDLE renderEvent = ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent;
4829 WAVEFORMATEX* captureFormat = NULL;
4830 WAVEFORMATEX* renderFormat = NULL;
4831 float captureSrRatio = 0.0f;
4832 float renderSrRatio = 0.0f;
4833 WasapiBuffer captureBuffer;
4834 WasapiBuffer renderBuffer;
4835 WasapiResampler* captureResampler = NULL;
4836 WasapiResampler* renderResampler = NULL;
4838 // declare local stream variables
4839 RtAudioCallback callback = ( RtAudioCallback ) stream_.callbackInfo.callback;
4840 BYTE* streamBuffer = NULL;
4841 unsigned long captureFlags = 0;
4842 unsigned int bufferFrameCount = 0;
4843 unsigned int numFramesPadding = 0;
4844 unsigned int convBufferSize = 0;
4845 bool callbackPushed = true;
4846 bool callbackPulled = false;
4847 bool callbackStopped = false;
4848 int callbackResult = 0;
4850 // convBuffer is used to store converted buffers between WASAPI and the user
4851 char* convBuffer = NULL;
4852 unsigned int convBuffSize = 0;
4853 unsigned int deviceBuffSize = 0;
4856 RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
4858 // Attempt to assign "Pro Audio" characteristic to thread
4859 HMODULE AvrtDll = LoadLibrary( (LPCTSTR) "AVRT.dll" );
4861 DWORD taskIndex = 0;
4862 TAvSetMmThreadCharacteristicsPtr AvSetMmThreadCharacteristicsPtr = ( TAvSetMmThreadCharacteristicsPtr ) GetProcAddress( AvrtDll, "AvSetMmThreadCharacteristicsW" );
4863 AvSetMmThreadCharacteristicsPtr( L"Pro Audio", &taskIndex );
4864 FreeLibrary( AvrtDll );
4867 // start capture stream if applicable
4868 if ( captureAudioClient ) {
4869 hr = captureAudioClient->GetMixFormat( &captureFormat );
4870 if ( FAILED( hr ) ) {
4871 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
4875 // init captureResampler
4876 captureResampler = new WasapiResampler( stream_.deviceFormat[INPUT] == RTAUDIO_FLOAT32 || stream_.deviceFormat[INPUT] == RTAUDIO_FLOAT64,
4877 formatBytes( stream_.deviceFormat[INPUT] ) * 8, stream_.nDeviceChannels[INPUT],
4878 captureFormat->nSamplesPerSec, stream_.sampleRate );
4880 captureSrRatio = ( ( float ) captureFormat->nSamplesPerSec / stream_.sampleRate );
4882 // initialize capture stream according to desire buffer size
4883 float desiredBufferSize = stream_.bufferSize * captureSrRatio;
4884 REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / captureFormat->nSamplesPerSec );
4886 if ( !captureClient ) {
4887 hr = captureAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
4888 AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
4889 desiredBufferPeriod,
4890 desiredBufferPeriod,
4893 if ( FAILED( hr ) ) {
4894 errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize capture audio client.";
4898 hr = captureAudioClient->GetService( __uuidof( IAudioCaptureClient ),
4899 ( void** ) &captureClient );
4900 if ( FAILED( hr ) ) {
4901 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture client handle.";
4905 // configure captureEvent to trigger on every available capture buffer
4906 captureEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
4907 if ( !captureEvent ) {
4908 errorType = RtAudioError::SYSTEM_ERROR;
4909 errorText_ = "RtApiWasapi::wasapiThread: Unable to create capture event.";
4913 hr = captureAudioClient->SetEventHandle( captureEvent );
4914 if ( FAILED( hr ) ) {
4915 errorText_ = "RtApiWasapi::wasapiThread: Unable to set capture event handle.";
4919 ( ( WasapiHandle* ) stream_.apiHandle )->captureClient = captureClient;
4920 ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent = captureEvent;
4923 unsigned int inBufferSize = 0;
4924 hr = captureAudioClient->GetBufferSize( &inBufferSize );
4925 if ( FAILED( hr ) ) {
4926 errorText_ = "RtApiWasapi::wasapiThread: Unable to get capture buffer size.";
4930 // scale outBufferSize according to stream->user sample rate ratio
4931 unsigned int outBufferSize = ( unsigned int ) ceilf( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT];
4932 inBufferSize *= stream_.nDeviceChannels[INPUT];
4934 // set captureBuffer size
4935 captureBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[INPUT] ) );
4937 // reset the capture stream
4938 hr = captureAudioClient->Reset();
4939 if ( FAILED( hr ) ) {
4940 errorText_ = "RtApiWasapi::wasapiThread: Unable to reset capture stream.";
4944 // start the capture stream
4945 hr = captureAudioClient->Start();
4946 if ( FAILED( hr ) ) {
4947 errorText_ = "RtApiWasapi::wasapiThread: Unable to start capture stream.";
4952 // start render stream if applicable
4953 if ( renderAudioClient ) {
4954 hr = renderAudioClient->GetMixFormat( &renderFormat );
4955 if ( FAILED( hr ) ) {
4956 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
4960 // init renderResampler
4961 renderResampler = new WasapiResampler( stream_.deviceFormat[OUTPUT] == RTAUDIO_FLOAT32 || stream_.deviceFormat[OUTPUT] == RTAUDIO_FLOAT64,
4962 formatBytes( stream_.deviceFormat[OUTPUT] ) * 8, stream_.nDeviceChannels[OUTPUT],
4963 stream_.sampleRate, renderFormat->nSamplesPerSec );
4965 renderSrRatio = ( ( float ) renderFormat->nSamplesPerSec / stream_.sampleRate );
4967 // initialize render stream according to desire buffer size
4968 float desiredBufferSize = stream_.bufferSize * renderSrRatio;
4969 REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / renderFormat->nSamplesPerSec );
4971 if ( !renderClient ) {
4972 hr = renderAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
4973 AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
4974 desiredBufferPeriod,
4975 desiredBufferPeriod,
4978 if ( FAILED( hr ) ) {
4979 errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize render audio client.";
4983 hr = renderAudioClient->GetService( __uuidof( IAudioRenderClient ),
4984 ( void** ) &renderClient );
4985 if ( FAILED( hr ) ) {
4986 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render client handle.";
4990 // configure renderEvent to trigger on every available render buffer
4991 renderEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
4992 if ( !renderEvent ) {
4993 errorType = RtAudioError::SYSTEM_ERROR;
4994 errorText_ = "RtApiWasapi::wasapiThread: Unable to create render event.";
4998 hr = renderAudioClient->SetEventHandle( renderEvent );
4999 if ( FAILED( hr ) ) {
5000 errorText_ = "RtApiWasapi::wasapiThread: Unable to set render event handle.";
5004 ( ( WasapiHandle* ) stream_.apiHandle )->renderClient = renderClient;
5005 ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent = renderEvent;
5008 unsigned int outBufferSize = 0;
5009 hr = renderAudioClient->GetBufferSize( &outBufferSize );
5010 if ( FAILED( hr ) ) {
5011 errorText_ = "RtApiWasapi::wasapiThread: Unable to get render buffer size.";
5015 // scale inBufferSize according to user->stream sample rate ratio
5016 unsigned int inBufferSize = ( unsigned int ) ceilf( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT];
5017 outBufferSize *= stream_.nDeviceChannels[OUTPUT];
5019 // set renderBuffer size
5020 renderBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[OUTPUT] ) );
5022 // reset the render stream
5023 hr = renderAudioClient->Reset();
5024 if ( FAILED( hr ) ) {
5025 errorText_ = "RtApiWasapi::wasapiThread: Unable to reset render stream.";
5029 // start the render stream
5030 hr = renderAudioClient->Start();
5031 if ( FAILED( hr ) ) {
5032 errorText_ = "RtApiWasapi::wasapiThread: Unable to start render stream.";
5037 // malloc buffer memory
5038 if ( stream_.mode == INPUT )
5040 using namespace std; // for ceilf
5041 convBuffSize = ( size_t ) ( ceilf( stream_.bufferSize * captureSrRatio ) ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
5042 deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
5044 else if ( stream_.mode == OUTPUT )
5046 convBuffSize = ( size_t ) ( ceilf( stream_.bufferSize * renderSrRatio ) ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
5047 deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
5049 else if ( stream_.mode == DUPLEX )
5051 convBuffSize = std::max( ( size_t ) ( ceilf( stream_.bufferSize * captureSrRatio ) ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
5052 ( size_t ) ( ceilf( stream_.bufferSize * renderSrRatio ) ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
5053 deviceBuffSize = std::max( stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
5054 stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
5057 convBuffSize *= 2; // allow overflow for *SrRatio remainders
5058 convBuffer = ( char* ) malloc( convBuffSize );
5059 stream_.deviceBuffer = ( char* ) malloc( deviceBuffSize );
5060 if ( !convBuffer || !stream_.deviceBuffer ) {
5061 errorType = RtAudioError::MEMORY_ERROR;
5062 errorText_ = "RtApiWasapi::wasapiThread: Error allocating device buffer memory.";
5066 // stream process loop
5067 while ( stream_.state != STREAM_STOPPING ) {
5068 if ( !callbackPulled ) {
5071 // 1. Pull callback buffer from inputBuffer
5072 // 2. If 1. was successful: Convert callback buffer to user sample rate and channel count
5073 // Convert callback buffer to user format
5075 if ( captureAudioClient )
5077 int samplesToPull = ( unsigned int ) floorf( stream_.bufferSize * captureSrRatio );
5078 if ( captureSrRatio != 1 )
5080 // account for remainders
5085 while ( convBufferSize < stream_.bufferSize )
5087 // Pull callback buffer from inputBuffer
5088 callbackPulled = captureBuffer.pullBuffer( convBuffer,
5089 samplesToPull * stream_.nDeviceChannels[INPUT],
5090 stream_.deviceFormat[INPUT] );
5092 if ( !callbackPulled )
5097 // Convert callback buffer to user sample rate
5098 unsigned int deviceBufferOffset = convBufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.userFormat );
5099 unsigned int convSamples = 0;
5101 captureResampler->Convert( stream_.deviceBuffer + deviceBufferOffset,
5106 convBufferSize += convSamples;
5107 samplesToPull = 1; // now pull one sample at a time until we have stream_.bufferSize samples
5110 if ( callbackPulled )
5112 if ( stream_.doConvertBuffer[INPUT] ) {
5113 // Convert callback buffer to user format
5114 convertBuffer( stream_.userBuffer[INPUT],
5115 stream_.deviceBuffer,
5116 stream_.convertInfo[INPUT] );
5119 // no further conversion, simple copy deviceBuffer to userBuffer
5120 memcpy( stream_.userBuffer[INPUT],
5121 stream_.deviceBuffer,
5122 stream_.bufferSize * stream_.nUserChannels[INPUT] * formatBytes( stream_.userFormat ) );
5127 // if there is no capture stream, set callbackPulled flag
5128 callbackPulled = true;
5133 // 1. Execute user callback method
5134 // 2. Handle return value from callback
5136 // if callback has not requested the stream to stop
5137 if ( callbackPulled && !callbackStopped ) {
5138 // Execute user callback method
5139 callbackResult = callback( stream_.userBuffer[OUTPUT],
5140 stream_.userBuffer[INPUT],
5143 captureFlags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY ? RTAUDIO_INPUT_OVERFLOW : 0,
5144 stream_.callbackInfo.userData );
5146 // Handle return value from callback
5147 if ( callbackResult == 1 ) {
5148 // instantiate a thread to stop this thread
5149 HANDLE threadHandle = CreateThread( NULL, 0, stopWasapiThread, this, 0, NULL );
5150 if ( !threadHandle ) {
5151 errorType = RtAudioError::THREAD_ERROR;
5152 errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream stop thread.";
5155 else if ( !CloseHandle( threadHandle ) ) {
5156 errorType = RtAudioError::THREAD_ERROR;
5157 errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream stop thread handle.";
5161 callbackStopped = true;
5163 else if ( callbackResult == 2 ) {
5164 // instantiate a thread to stop this thread
5165 HANDLE threadHandle = CreateThread( NULL, 0, abortWasapiThread, this, 0, NULL );
5166 if ( !threadHandle ) {
5167 errorType = RtAudioError::THREAD_ERROR;
5168 errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream abort thread.";
5171 else if ( !CloseHandle( threadHandle ) ) {
5172 errorType = RtAudioError::THREAD_ERROR;
5173 errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream abort thread handle.";
5177 callbackStopped = true;
5184 // 1. Convert callback buffer to stream format
5185 // 2. Convert callback buffer to stream sample rate and channel count
5186 // 3. Push callback buffer into outputBuffer
5188 if ( renderAudioClient && callbackPulled )
5190 // if the last call to renderBuffer.PushBuffer() was successful
5191 if ( callbackPushed || convBufferSize == 0 )
5193 if ( stream_.doConvertBuffer[OUTPUT] )
5195 // Convert callback buffer to stream format
5196 convertBuffer( stream_.deviceBuffer,
5197 stream_.userBuffer[OUTPUT],
5198 stream_.convertInfo[OUTPUT] );
5202 // Convert callback buffer to stream sample rate
5203 renderResampler->Convert( convBuffer,
5204 stream_.deviceBuffer,
5209 // Push callback buffer into outputBuffer
5210 callbackPushed = renderBuffer.pushBuffer( convBuffer,
5211 convBufferSize * stream_.nDeviceChannels[OUTPUT],
5212 stream_.deviceFormat[OUTPUT] );
5215 // if there is no render stream, set callbackPushed flag
5216 callbackPushed = true;
5221 // 1. Get capture buffer from stream
5222 // 2. Push capture buffer into inputBuffer
5223 // 3. If 2. was successful: Release capture buffer
5225 if ( captureAudioClient ) {
5226 // if the callback input buffer was not pulled from captureBuffer, wait for next capture event
5227 if ( !callbackPulled ) {
5228 WaitForSingleObject( captureEvent, INFINITE );
5231 // Get capture buffer from stream
5232 hr = captureClient->GetBuffer( &streamBuffer,
5234 &captureFlags, NULL, NULL );
5235 if ( FAILED( hr ) ) {
5236 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture buffer.";
5240 if ( bufferFrameCount != 0 ) {
5241 // Push capture buffer into inputBuffer
5242 if ( captureBuffer.pushBuffer( ( char* ) streamBuffer,
5243 bufferFrameCount * stream_.nDeviceChannels[INPUT],
5244 stream_.deviceFormat[INPUT] ) )
5246 // Release capture buffer
5247 hr = captureClient->ReleaseBuffer( bufferFrameCount );
5248 if ( FAILED( hr ) ) {
5249 errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
5255 // Inform WASAPI that capture was unsuccessful
5256 hr = captureClient->ReleaseBuffer( 0 );
5257 if ( FAILED( hr ) ) {
5258 errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
5265 // Inform WASAPI that capture was unsuccessful
5266 hr = captureClient->ReleaseBuffer( 0 );
5267 if ( FAILED( hr ) ) {
5268 errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
5276 // 1. Get render buffer from stream
5277 // 2. Pull next buffer from outputBuffer
5278 // 3. If 2. was successful: Fill render buffer with next buffer
5279 // Release render buffer
5281 if ( renderAudioClient ) {
5282 // if the callback output buffer was not pushed to renderBuffer, wait for next render event
5283 if ( callbackPulled && !callbackPushed ) {
5284 WaitForSingleObject( renderEvent, INFINITE );
5287 // Get render buffer from stream
5288 hr = renderAudioClient->GetBufferSize( &bufferFrameCount );
5289 if ( FAILED( hr ) ) {
5290 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer size.";
5294 hr = renderAudioClient->GetCurrentPadding( &numFramesPadding );
5295 if ( FAILED( hr ) ) {
5296 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer padding.";
5300 bufferFrameCount -= numFramesPadding;
5302 if ( bufferFrameCount != 0 ) {
5303 hr = renderClient->GetBuffer( bufferFrameCount, &streamBuffer );
5304 if ( FAILED( hr ) ) {
5305 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer.";
5309 // Pull next buffer from outputBuffer
5310 // Fill render buffer with next buffer
5311 if ( renderBuffer.pullBuffer( ( char* ) streamBuffer,
5312 bufferFrameCount * stream_.nDeviceChannels[OUTPUT],
5313 stream_.deviceFormat[OUTPUT] ) )
5315 // Release render buffer
5316 hr = renderClient->ReleaseBuffer( bufferFrameCount, 0 );
5317 if ( FAILED( hr ) ) {
5318 errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
5324 // Inform WASAPI that render was unsuccessful
5325 hr = renderClient->ReleaseBuffer( 0, 0 );
5326 if ( FAILED( hr ) ) {
5327 errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
5334 // Inform WASAPI that render was unsuccessful
5335 hr = renderClient->ReleaseBuffer( 0, 0 );
5336 if ( FAILED( hr ) ) {
5337 errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
5343 // if the callback buffer was pushed renderBuffer reset callbackPulled flag
5344 if ( callbackPushed ) {
5345 // unsetting the callbackPulled flag lets the stream know that
5346 // the audio device is ready for another callback output buffer.
5347 callbackPulled = false;
5350 RtApi::tickStreamTime();
5357 CoTaskMemFree( captureFormat );
5358 CoTaskMemFree( renderFormat );
5360 free ( convBuffer );
5361 delete renderResampler;
5362 delete captureResampler;
5366 // update stream state
5367 stream_.state = STREAM_STOPPED;
5369 if ( errorText_.empty() )
5375 //******************** End of __WINDOWS_WASAPI__ *********************//
5379 #if defined(__WINDOWS_DS__) // Windows DirectSound API
5381 // Modified by Robin Davies, October 2005
5382 // - Improvements to DirectX pointer chasing.
5383 // - Bug fix for non-power-of-two Asio granularity used by Edirol PCR-A30.
5384 // - Auto-call CoInitialize for DSOUND and ASIO platforms.
5385 // Various revisions for RtAudio 4.0 by Gary Scavone, April 2007
5386 // Changed device query structure for RtAudio 4.0.7, January 2010
5388 #include <windows.h>
5389 #include <process.h>
5390 #include <mmsystem.h>
5394 #include <algorithm>
5396 #if defined(__MINGW32__)
5397 // missing from latest mingw winapi
5398 #define WAVE_FORMAT_96M08 0x00010000 /* 96 kHz, Mono, 8-bit */
5399 #define WAVE_FORMAT_96S08 0x00020000 /* 96 kHz, Stereo, 8-bit */
5400 #define WAVE_FORMAT_96M16 0x00040000 /* 96 kHz, Mono, 16-bit */
5401 #define WAVE_FORMAT_96S16 0x00080000 /* 96 kHz, Stereo, 16-bit */
5404 #define MINIMUM_DEVICE_BUFFER_SIZE 32768
5406 #ifdef _MSC_VER // if Microsoft Visual C++
5407 #pragma comment( lib, "winmm.lib" ) // then, auto-link winmm.lib. Otherwise, it has to be added manually.
5410 static inline DWORD dsPointerBetween( DWORD pointer, DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize )
5412 if ( pointer > bufferSize ) pointer -= bufferSize;
5413 if ( laterPointer < earlierPointer ) laterPointer += bufferSize;
5414 if ( pointer < earlierPointer ) pointer += bufferSize;
5415 return pointer >= earlierPointer && pointer < laterPointer;
5418 // A structure to hold various information related to the DirectSound
5419 // API implementation.
5421 unsigned int drainCounter; // Tracks callback counts when draining
5422 bool internalDrain; // Indicates if stop is initiated from callback or not.
5426 UINT bufferPointer[2];
5427 DWORD dsBufferSize[2];
5428 DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by.
5432 :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; }
5435 // Declarations for utility functions, callbacks, and structures
5436 // specific to the DirectSound implementation.
5437 static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
5438 LPCTSTR description,
5442 static const char* getErrorString( int code );
5444 static unsigned __stdcall callbackHandler( void *ptr );
5453 : found(false) { validId[0] = false; validId[1] = false; }
5456 struct DsProbeData {
5458 std::vector<struct DsDevice>* dsDevices;
5461 RtApiDs :: RtApiDs()
5463 // Dsound will run both-threaded. If CoInitialize fails, then just
5464 // accept whatever the mainline chose for a threading model.
5465 coInitialized_ = false;
5466 HRESULT hr = CoInitialize( NULL );
5467 if ( !FAILED( hr ) ) coInitialized_ = true;
5470 RtApiDs :: ~RtApiDs()
5472 if ( stream_.state != STREAM_CLOSED ) closeStream();
5473 if ( coInitialized_ ) CoUninitialize(); // balanced call.
5476 // The DirectSound default output is always the first device.
5477 unsigned int RtApiDs :: getDefaultOutputDevice( void )
5482 // The DirectSound default input is always the first input device,
5483 // which is the first capture device enumerated.
5484 unsigned int RtApiDs :: getDefaultInputDevice( void )
5489 unsigned int RtApiDs :: getDeviceCount( void )
5491 // Set query flag for previously found devices to false, so that we
5492 // can check for any devices that have disappeared.
5493 for ( unsigned int i=0; i<dsDevices.size(); i++ )
5494 dsDevices[i].found = false;
5496 // Query DirectSound devices.
5497 struct DsProbeData probeInfo;
5498 probeInfo.isInput = false;
5499 probeInfo.dsDevices = &dsDevices;
5500 HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );
5501 if ( FAILED( result ) ) {
5502 errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!";
5503 errorText_ = errorStream_.str();
5504 error( RtAudioError::WARNING );
5507 // Query DirectSoundCapture devices.
5508 probeInfo.isInput = true;
5509 result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );
5510 if ( FAILED( result ) ) {
5511 errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!";
5512 errorText_ = errorStream_.str();
5513 error( RtAudioError::WARNING );
5516 // Clean out any devices that may have disappeared (code update submitted by Eli Zehngut).
5517 for ( unsigned int i=0; i<dsDevices.size(); ) {
5518 if ( dsDevices[i].found == false ) dsDevices.erase( dsDevices.begin() + i );
5522 return static_cast<unsigned int>(dsDevices.size());
5525 RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device )
5527 RtAudio::DeviceInfo info;
5528 info.probed = false;
5530 if ( dsDevices.size() == 0 ) {
5531 // Force a query of all devices
5533 if ( dsDevices.size() == 0 ) {
5534 errorText_ = "RtApiDs::getDeviceInfo: no devices found!";
5535 error( RtAudioError::INVALID_USE );
5540 if ( device >= dsDevices.size() ) {
5541 errorText_ = "RtApiDs::getDeviceInfo: device ID is invalid!";
5542 error( RtAudioError::INVALID_USE );
5547 if ( dsDevices[ device ].validId[0] == false ) goto probeInput;
5549 LPDIRECTSOUND output;
5551 result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );
5552 if ( FAILED( result ) ) {
5553 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!";
5554 errorText_ = errorStream_.str();
5555 error( RtAudioError::WARNING );
5559 outCaps.dwSize = sizeof( outCaps );
5560 result = output->GetCaps( &outCaps );
5561 if ( FAILED( result ) ) {
5563 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting capabilities!";
5564 errorText_ = errorStream_.str();
5565 error( RtAudioError::WARNING );
5569 // Get output channel information.
5570 info.outputChannels = ( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1;
5572 // Get sample rate information.
5573 info.sampleRates.clear();
5574 for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
5575 if ( SAMPLE_RATES[k] >= (unsigned int) outCaps.dwMinSecondarySampleRate &&
5576 SAMPLE_RATES[k] <= (unsigned int) outCaps.dwMaxSecondarySampleRate ) {
5577 info.sampleRates.push_back( SAMPLE_RATES[k] );
5579 if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
5580 info.preferredSampleRate = SAMPLE_RATES[k];
5584 // Get format information.
5585 if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT ) info.nativeFormats |= RTAUDIO_SINT16;
5586 if ( outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) info.nativeFormats |= RTAUDIO_SINT8;
5590 if ( getDefaultOutputDevice() == device )
5591 info.isDefaultOutput = true;
5593 if ( dsDevices[ device ].validId[1] == false ) {
5594 info.name = dsDevices[ device ].name;
5601 LPDIRECTSOUNDCAPTURE input;
5602 result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );
5603 if ( FAILED( result ) ) {
5604 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";
5605 errorText_ = errorStream_.str();
5606 error( RtAudioError::WARNING );
5611 inCaps.dwSize = sizeof( inCaps );
5612 result = input->GetCaps( &inCaps );
5613 if ( FAILED( result ) ) {
5615 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsDevices[ device ].name << ")!";
5616 errorText_ = errorStream_.str();
5617 error( RtAudioError::WARNING );
5621 // Get input channel information.
5622 info.inputChannels = inCaps.dwChannels;
5624 // Get sample rate and format information.
5625 std::vector<unsigned int> rates;
5626 if ( inCaps.dwChannels >= 2 ) {
5627 if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.nativeFormats |= RTAUDIO_SINT16;
5628 if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.nativeFormats |= RTAUDIO_SINT16;
5629 if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.nativeFormats |= RTAUDIO_SINT16;
5630 if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.nativeFormats |= RTAUDIO_SINT16;
5631 if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.nativeFormats |= RTAUDIO_SINT8;
5632 if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.nativeFormats |= RTAUDIO_SINT8;
5633 if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.nativeFormats |= RTAUDIO_SINT8;
5634 if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.nativeFormats |= RTAUDIO_SINT8;
5636 if ( info.nativeFormats & RTAUDIO_SINT16 ) {
5637 if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) rates.push_back( 11025 );
5638 if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) rates.push_back( 22050 );
5639 if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) rates.push_back( 44100 );
5640 if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) rates.push_back( 96000 );
5642 else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
5643 if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) rates.push_back( 11025 );
5644 if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) rates.push_back( 22050 );
5645 if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) rates.push_back( 44100 );
5646 if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) rates.push_back( 96000 );
5649 else if ( inCaps.dwChannels == 1 ) {
5650 if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.nativeFormats |= RTAUDIO_SINT16;
5651 if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.nativeFormats |= RTAUDIO_SINT16;
5652 if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.nativeFormats |= RTAUDIO_SINT16;
5653 if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.nativeFormats |= RTAUDIO_SINT16;
5654 if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.nativeFormats |= RTAUDIO_SINT8;
5655 if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.nativeFormats |= RTAUDIO_SINT8;
5656 if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.nativeFormats |= RTAUDIO_SINT8;
5657 if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.nativeFormats |= RTAUDIO_SINT8;
5659 if ( info.nativeFormats & RTAUDIO_SINT16 ) {
5660 if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) rates.push_back( 11025 );
5661 if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) rates.push_back( 22050 );
5662 if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) rates.push_back( 44100 );
5663 if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) rates.push_back( 96000 );
5665 else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
5666 if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) rates.push_back( 11025 );
5667 if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) rates.push_back( 22050 );
5668 if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) rates.push_back( 44100 );
5669 if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) rates.push_back( 96000 );
5672 else info.inputChannels = 0; // technically, this would be an error
5676 if ( info.inputChannels == 0 ) return info;
5678 // Copy the supported rates to the info structure but avoid duplication.
5680 for ( unsigned int i=0; i<rates.size(); i++ ) {
5682 for ( unsigned int j=0; j<info.sampleRates.size(); j++ ) {
5683 if ( rates[i] == info.sampleRates[j] ) {
5688 if ( found == false ) info.sampleRates.push_back( rates[i] );
5690 std::sort( info.sampleRates.begin(), info.sampleRates.end() );
5692 // If device opens for both playback and capture, we determine the channels.
5693 if ( info.outputChannels > 0 && info.inputChannels > 0 )
5694 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
5696 if ( device == 0 ) info.isDefaultInput = true;
5698 // Copy name and return.
5699 info.name = dsDevices[ device ].name;
5704 bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
5705 unsigned int firstChannel, unsigned int sampleRate,
5706 RtAudioFormat format, unsigned int *bufferSize,
5707 RtAudio::StreamOptions *options )
5709 if ( channels + firstChannel > 2 ) {
5710 errorText_ = "RtApiDs::probeDeviceOpen: DirectSound does not support more than 2 channels per device.";
5714 size_t nDevices = dsDevices.size();
5715 if ( nDevices == 0 ) {
5716 // This should not happen because a check is made before this function is called.
5717 errorText_ = "RtApiDs::probeDeviceOpen: no devices found!";
5721 if ( device >= nDevices ) {
5722 // This should not happen because a check is made before this function is called.
5723 errorText_ = "RtApiDs::probeDeviceOpen: device ID is invalid!";
5727 if ( mode == OUTPUT ) {
5728 if ( dsDevices[ device ].validId[0] == false ) {
5729 errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support output!";
5730 errorText_ = errorStream_.str();
5734 else { // mode == INPUT
5735 if ( dsDevices[ device ].validId[1] == false ) {
5736 errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support input!";
5737 errorText_ = errorStream_.str();
5742 // According to a note in PortAudio, using GetDesktopWindow()
5743 // instead of GetForegroundWindow() is supposed to avoid problems
5744 // that occur when the application's window is not the foreground
5745 // window. Also, if the application window closes before the
5746 // DirectSound buffer, DirectSound can crash. In the past, I had
5747 // problems when using GetDesktopWindow() but it seems fine now
5748 // (January 2010). I'll leave it commented here.
5749 // HWND hWnd = GetForegroundWindow();
5750 HWND hWnd = GetDesktopWindow();
5752 // Check the numberOfBuffers parameter and limit the lowest value to
5753 // two. This is a judgement call and a value of two is probably too
5754 // low for capture, but it should work for playback.
5756 if ( options ) nBuffers = options->numberOfBuffers;
5757 if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) nBuffers = 2;
5758 if ( nBuffers < 2 ) nBuffers = 3;
5760 // Check the lower range of the user-specified buffer size and set
5761 // (arbitrarily) to a lower bound of 32.
5762 if ( *bufferSize < 32 ) *bufferSize = 32;
5764 // Create the wave format structure. The data format setting will
5765 // be determined later.
5766 WAVEFORMATEX waveFormat;
5767 ZeroMemory( &waveFormat, sizeof(WAVEFORMATEX) );
5768 waveFormat.wFormatTag = WAVE_FORMAT_PCM;
5769 waveFormat.nChannels = channels + firstChannel;
5770 waveFormat.nSamplesPerSec = (unsigned long) sampleRate;
5772 // Determine the device buffer size. By default, we'll use the value
5773 // defined above (32K), but we will grow it to make allowances for
5774 // very large software buffer sizes.
5775 DWORD dsBufferSize = MINIMUM_DEVICE_BUFFER_SIZE;
5776 DWORD dsPointerLeadTime = 0;
5778 void *ohandle = 0, *bhandle = 0;
5780 if ( mode == OUTPUT ) {
5782 LPDIRECTSOUND output;
5783 result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );
5784 if ( FAILED( result ) ) {
5785 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!";
5786 errorText_ = errorStream_.str();
5791 outCaps.dwSize = sizeof( outCaps );
5792 result = output->GetCaps( &outCaps );
5793 if ( FAILED( result ) ) {
5795 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting capabilities (" << dsDevices[ device ].name << ")!";
5796 errorText_ = errorStream_.str();
5800 // Check channel information.
5801 if ( channels + firstChannel == 2 && !( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ) {
5802 errorStream_ << "RtApiDs::getDeviceInfo: the output device (" << dsDevices[ device ].name << ") does not support stereo playback.";
5803 errorText_ = errorStream_.str();
5807 // Check format information. Use 16-bit format unless not
5808 // supported or user requests 8-bit.
5809 if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT &&
5810 !( format == RTAUDIO_SINT8 && outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) ) {
5811 waveFormat.wBitsPerSample = 16;
5812 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
5815 waveFormat.wBitsPerSample = 8;
5816 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
5818 stream_.userFormat = format;
5820 // Update wave format structure and buffer information.
5821 waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
5822 waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
5823 dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
5825 // If the user wants an even bigger buffer, increase the device buffer size accordingly.
5826 while ( dsPointerLeadTime * 2U > dsBufferSize )
5829 // Set cooperative level to DSSCL_EXCLUSIVE ... sound stops when window focus changes.
5830 // result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE );
5831 // Set cooperative level to DSSCL_PRIORITY ... sound remains when window focus changes.
5832 result = output->SetCooperativeLevel( hWnd, DSSCL_PRIORITY );
5833 if ( FAILED( result ) ) {
5835 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting cooperative level (" << dsDevices[ device ].name << ")!";
5836 errorText_ = errorStream_.str();
5840 // Even though we will write to the secondary buffer, we need to
5841 // access the primary buffer to set the correct output format
5842 // (since the default is 8-bit, 22 kHz!). Setup the DS primary
5843 // buffer description.
5844 DSBUFFERDESC bufferDescription;
5845 ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
5846 bufferDescription.dwSize = sizeof( DSBUFFERDESC );
5847 bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER;
5849 // Obtain the primary buffer
5850 LPDIRECTSOUNDBUFFER buffer;
5851 result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
5852 if ( FAILED( result ) ) {
5854 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") accessing primary buffer (" << dsDevices[ device ].name << ")!";
5855 errorText_ = errorStream_.str();
5859 // Set the primary DS buffer sound format.
5860 result = buffer->SetFormat( &waveFormat );
5861 if ( FAILED( result ) ) {
5863 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting primary buffer format (" << dsDevices[ device ].name << ")!";
5864 errorText_ = errorStream_.str();
5868 // Setup the secondary DS buffer description.
5869 ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
5870 bufferDescription.dwSize = sizeof( DSBUFFERDESC );
5871 bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
5872 DSBCAPS_GLOBALFOCUS |
5873 DSBCAPS_GETCURRENTPOSITION2 |
5874 DSBCAPS_LOCHARDWARE ); // Force hardware mixing
5875 bufferDescription.dwBufferBytes = dsBufferSize;
5876 bufferDescription.lpwfxFormat = &waveFormat;
5878 // Try to create the secondary DS buffer. If that doesn't work,
5879 // try to use software mixing. Otherwise, there's a problem.
5880 result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
5881 if ( FAILED( result ) ) {
5882 bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
5883 DSBCAPS_GLOBALFOCUS |
5884 DSBCAPS_GETCURRENTPOSITION2 |
5885 DSBCAPS_LOCSOFTWARE ); // Force software mixing
5886 result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
5887 if ( FAILED( result ) ) {
5889 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating secondary buffer (" << dsDevices[ device ].name << ")!";
5890 errorText_ = errorStream_.str();
5895 // Get the buffer size ... might be different from what we specified.
5897 dsbcaps.dwSize = sizeof( DSBCAPS );
5898 result = buffer->GetCaps( &dsbcaps );
5899 if ( FAILED( result ) ) {
5902 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";
5903 errorText_ = errorStream_.str();
5907 dsBufferSize = dsbcaps.dwBufferBytes;
5909 // Lock the DS buffer
5912 result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );
5913 if ( FAILED( result ) ) {
5916 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking buffer (" << dsDevices[ device ].name << ")!";
5917 errorText_ = errorStream_.str();
5921 // Zero the DS buffer
5922 ZeroMemory( audioPtr, dataLen );
5924 // Unlock the DS buffer
5925 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
5926 if ( FAILED( result ) ) {
5929 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking buffer (" << dsDevices[ device ].name << ")!";
5930 errorText_ = errorStream_.str();
5934 ohandle = (void *) output;
5935 bhandle = (void *) buffer;
5938 if ( mode == INPUT ) {
5940 LPDIRECTSOUNDCAPTURE input;
5941 result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );
5942 if ( FAILED( result ) ) {
5943 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";
5944 errorText_ = errorStream_.str();
5949 inCaps.dwSize = sizeof( inCaps );
5950 result = input->GetCaps( &inCaps );
5951 if ( FAILED( result ) ) {
5953 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsDevices[ device ].name << ")!";
5954 errorText_ = errorStream_.str();
5958 // Check channel information.
5959 if ( inCaps.dwChannels < channels + firstChannel ) {
5960 errorText_ = "RtApiDs::getDeviceInfo: the input device does not support requested input channels.";
5964 // Check format information. Use 16-bit format unless user
5966 DWORD deviceFormats;
5967 if ( channels + firstChannel == 2 ) {
5968 deviceFormats = WAVE_FORMAT_1S08 | WAVE_FORMAT_2S08 | WAVE_FORMAT_4S08 | WAVE_FORMAT_96S08;
5969 if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
5970 waveFormat.wBitsPerSample = 8;
5971 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
5973 else { // assume 16-bit is supported
5974 waveFormat.wBitsPerSample = 16;
5975 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
5978 else { // channel == 1
5979 deviceFormats = WAVE_FORMAT_1M08 | WAVE_FORMAT_2M08 | WAVE_FORMAT_4M08 | WAVE_FORMAT_96M08;
5980 if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
5981 waveFormat.wBitsPerSample = 8;
5982 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
5984 else { // assume 16-bit is supported
5985 waveFormat.wBitsPerSample = 16;
5986 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
5989 stream_.userFormat = format;
5991 // Update wave format structure and buffer information.
5992 waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
5993 waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
5994 dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
5996 // If the user wants an even bigger buffer, increase the device buffer size accordingly.
5997 while ( dsPointerLeadTime * 2U > dsBufferSize )
6000 // Setup the secondary DS buffer description.
6001 DSCBUFFERDESC bufferDescription;
6002 ZeroMemory( &bufferDescription, sizeof( DSCBUFFERDESC ) );
6003 bufferDescription.dwSize = sizeof( DSCBUFFERDESC );
6004 bufferDescription.dwFlags = 0;
6005 bufferDescription.dwReserved = 0;
6006 bufferDescription.dwBufferBytes = dsBufferSize;
6007 bufferDescription.lpwfxFormat = &waveFormat;
6009 // Create the capture buffer.
6010 LPDIRECTSOUNDCAPTUREBUFFER buffer;
6011 result = input->CreateCaptureBuffer( &bufferDescription, &buffer, NULL );
6012 if ( FAILED( result ) ) {
6014 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating input buffer (" << dsDevices[ device ].name << ")!";
6015 errorText_ = errorStream_.str();
6019 // Get the buffer size ... might be different from what we specified.
6021 dscbcaps.dwSize = sizeof( DSCBCAPS );
6022 result = buffer->GetCaps( &dscbcaps );
6023 if ( FAILED( result ) ) {
6026 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";
6027 errorText_ = errorStream_.str();
6031 dsBufferSize = dscbcaps.dwBufferBytes;
6033 // NOTE: We could have a problem here if this is a duplex stream
6034 // and the play and capture hardware buffer sizes are different
6035 // (I'm actually not sure if that is a problem or not).
6036 // Currently, we are not verifying that.
6038 // Lock the capture buffer
6041 result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );
6042 if ( FAILED( result ) ) {
6045 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking input buffer (" << dsDevices[ device ].name << ")!";
6046 errorText_ = errorStream_.str();
6051 ZeroMemory( audioPtr, dataLen );
6053 // Unlock the buffer
6054 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
6055 if ( FAILED( result ) ) {
6058 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking input buffer (" << dsDevices[ device ].name << ")!";
6059 errorText_ = errorStream_.str();
6063 ohandle = (void *) input;
6064 bhandle = (void *) buffer;
6067 // Set various stream parameters
6068 DsHandle *handle = 0;
6069 stream_.nDeviceChannels[mode] = channels + firstChannel;
6070 stream_.nUserChannels[mode] = channels;
6071 stream_.bufferSize = *bufferSize;
6072 stream_.channelOffset[mode] = firstChannel;
6073 stream_.deviceInterleaved[mode] = true;
6074 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
6075 else stream_.userInterleaved = true;
6077 // Set flag for buffer conversion
6078 stream_.doConvertBuffer[mode] = false;
6079 if (stream_.nUserChannels[mode] != stream_.nDeviceChannels[mode])
6080 stream_.doConvertBuffer[mode] = true;
6081 if (stream_.userFormat != stream_.deviceFormat[mode])
6082 stream_.doConvertBuffer[mode] = true;
6083 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
6084 stream_.nUserChannels[mode] > 1 )
6085 stream_.doConvertBuffer[mode] = true;
6087 // Allocate necessary internal buffers
6088 long bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
6089 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
6090 if ( stream_.userBuffer[mode] == NULL ) {
6091 errorText_ = "RtApiDs::probeDeviceOpen: error allocating user buffer memory.";
6095 if ( stream_.doConvertBuffer[mode] ) {
6097 bool makeBuffer = true;
6098 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
6099 if ( mode == INPUT ) {
6100 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
6101 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
6102 if ( bufferBytes <= (long) bytesOut ) makeBuffer = false;
6107 bufferBytes *= *bufferSize;
6108 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
6109 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
6110 if ( stream_.deviceBuffer == NULL ) {
6111 errorText_ = "RtApiDs::probeDeviceOpen: error allocating device buffer memory.";
6117 // Allocate our DsHandle structures for the stream.
6118 if ( stream_.apiHandle == 0 ) {
6120 handle = new DsHandle;
6122 catch ( std::bad_alloc& ) {
6123 errorText_ = "RtApiDs::probeDeviceOpen: error allocating AsioHandle memory.";
6127 // Create a manual-reset event.
6128 handle->condition = CreateEvent( NULL, // no security
6129 TRUE, // manual-reset
6130 FALSE, // non-signaled initially
6132 stream_.apiHandle = (void *) handle;
6135 handle = (DsHandle *) stream_.apiHandle;
6136 handle->id[mode] = ohandle;
6137 handle->buffer[mode] = bhandle;
6138 handle->dsBufferSize[mode] = dsBufferSize;
6139 handle->dsPointerLeadTime[mode] = dsPointerLeadTime;
6141 stream_.device[mode] = device;
6142 stream_.state = STREAM_STOPPED;
6143 if ( stream_.mode == OUTPUT && mode == INPUT )
6144 // We had already set up an output stream.
6145 stream_.mode = DUPLEX;
6147 stream_.mode = mode;
6148 stream_.nBuffers = nBuffers;
6149 stream_.sampleRate = sampleRate;
6151 // Setup the buffer conversion information structure.
6152 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
6154 // Setup the callback thread.
6155 if ( stream_.callbackInfo.isRunning == false ) {
6157 stream_.callbackInfo.isRunning = true;
6158 stream_.callbackInfo.object = (void *) this;
6159 stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &callbackHandler,
6160 &stream_.callbackInfo, 0, &threadId );
6161 if ( stream_.callbackInfo.thread == 0 ) {
6162 errorText_ = "RtApiDs::probeDeviceOpen: error creating callback thread!";
6166 // Boost DS thread priority
6167 SetThreadPriority( (HANDLE) stream_.callbackInfo.thread, THREAD_PRIORITY_HIGHEST );
6173 if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
6174 LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
6175 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6176 if ( buffer ) buffer->Release();
6179 if ( handle->buffer[1] ) {
6180 LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
6181 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6182 if ( buffer ) buffer->Release();
6185 CloseHandle( handle->condition );
6187 stream_.apiHandle = 0;
6190 for ( int i=0; i<2; i++ ) {
6191 if ( stream_.userBuffer[i] ) {
6192 free( stream_.userBuffer[i] );
6193 stream_.userBuffer[i] = 0;
6197 if ( stream_.deviceBuffer ) {
6198 free( stream_.deviceBuffer );
6199 stream_.deviceBuffer = 0;
6202 stream_.state = STREAM_CLOSED;
6206 void RtApiDs :: closeStream()
6208 if ( stream_.state == STREAM_CLOSED ) {
6209 errorText_ = "RtApiDs::closeStream(): no open stream to close!";
6210 error( RtAudioError::WARNING );
6214 // Stop the callback thread.
6215 stream_.callbackInfo.isRunning = false;
6216 WaitForSingleObject( (HANDLE) stream_.callbackInfo.thread, INFINITE );
6217 CloseHandle( (HANDLE) stream_.callbackInfo.thread );
6219 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6221 if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
6222 LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
6223 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6230 if ( handle->buffer[1] ) {
6231 LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
6232 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6239 CloseHandle( handle->condition );
6241 stream_.apiHandle = 0;
6244 for ( int i=0; i<2; i++ ) {
6245 if ( stream_.userBuffer[i] ) {
6246 free( stream_.userBuffer[i] );
6247 stream_.userBuffer[i] = 0;
6251 if ( stream_.deviceBuffer ) {
6252 free( stream_.deviceBuffer );
6253 stream_.deviceBuffer = 0;
6256 stream_.mode = UNINITIALIZED;
6257 stream_.state = STREAM_CLOSED;
6260 void RtApiDs :: startStream()
6263 if ( stream_.state == STREAM_RUNNING ) {
6264 errorText_ = "RtApiDs::startStream(): the stream is already running!";
6265 error( RtAudioError::WARNING );
6269 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6271 // Increase scheduler frequency on lesser windows (a side-effect of
6272 // increasing timer accuracy). On greater windows (Win2K or later),
6273 // this is already in effect.
6274 timeBeginPeriod( 1 );
6276 buffersRolling = false;
6277 duplexPrerollBytes = 0;
6279 if ( stream_.mode == DUPLEX ) {
6280 // 0.5 seconds of silence in DUPLEX mode while the devices spin up and synchronize.
6281 duplexPrerollBytes = (int) ( 0.5 * stream_.sampleRate * formatBytes( stream_.deviceFormat[1] ) * stream_.nDeviceChannels[1] );
6285 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
6287 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6288 result = buffer->Play( 0, 0, DSBPLAY_LOOPING );
6289 if ( FAILED( result ) ) {
6290 errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting output buffer!";
6291 errorText_ = errorStream_.str();
6296 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
6298 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6299 result = buffer->Start( DSCBSTART_LOOPING );
6300 if ( FAILED( result ) ) {
6301 errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting input buffer!";
6302 errorText_ = errorStream_.str();
6307 handle->drainCounter = 0;
6308 handle->internalDrain = false;
6309 ResetEvent( handle->condition );
6310 stream_.state = STREAM_RUNNING;
6313 if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );
6316 void RtApiDs :: stopStream()
6319 if ( stream_.state == STREAM_STOPPED ) {
6320 errorText_ = "RtApiDs::stopStream(): the stream is already stopped!";
6321 error( RtAudioError::WARNING );
6328 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6329 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
6330 if ( handle->drainCounter == 0 ) {
6331 handle->drainCounter = 2;
6332 WaitForSingleObject( handle->condition, INFINITE ); // block until signaled
6335 stream_.state = STREAM_STOPPED;
6337 MUTEX_LOCK( &stream_.mutex );
6339 // Stop the buffer and clear memory
6340 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6341 result = buffer->Stop();
6342 if ( FAILED( result ) ) {
6343 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping output buffer!";
6344 errorText_ = errorStream_.str();
6348 // Lock the buffer and clear it so that if we start to play again,
6349 // we won't have old data playing.
6350 result = buffer->Lock( 0, handle->dsBufferSize[0], &audioPtr, &dataLen, NULL, NULL, 0 );
6351 if ( FAILED( result ) ) {
6352 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking output buffer!";
6353 errorText_ = errorStream_.str();
6357 // Zero the DS buffer
6358 ZeroMemory( audioPtr, dataLen );
6360 // Unlock the DS buffer
6361 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
6362 if ( FAILED( result ) ) {
6363 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking output buffer!";
6364 errorText_ = errorStream_.str();
6368 // If we start playing again, we must begin at beginning of buffer.
6369 handle->bufferPointer[0] = 0;
6372 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
6373 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6377 stream_.state = STREAM_STOPPED;
6379 if ( stream_.mode != DUPLEX )
6380 MUTEX_LOCK( &stream_.mutex );
6382 result = buffer->Stop();
6383 if ( FAILED( result ) ) {
6384 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping input buffer!";
6385 errorText_ = errorStream_.str();
6389 // Lock the buffer and clear it so that if we start to play again,
6390 // we won't have old data playing.
6391 result = buffer->Lock( 0, handle->dsBufferSize[1], &audioPtr, &dataLen, NULL, NULL, 0 );
6392 if ( FAILED( result ) ) {
6393 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking input buffer!";
6394 errorText_ = errorStream_.str();
6398 // Zero the DS buffer
6399 ZeroMemory( audioPtr, dataLen );
6401 // Unlock the DS buffer
6402 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
6403 if ( FAILED( result ) ) {
6404 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking input buffer!";
6405 errorText_ = errorStream_.str();
6409 // If we start recording again, we must begin at beginning of buffer.
6410 handle->bufferPointer[1] = 0;
6414 timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows.
6415 MUTEX_UNLOCK( &stream_.mutex );
6417 if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );
6420 void RtApiDs :: abortStream()
6423 if ( stream_.state == STREAM_STOPPED ) {
6424 errorText_ = "RtApiDs::abortStream(): the stream is already stopped!";
6425 error( RtAudioError::WARNING );
6429 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6430 handle->drainCounter = 2;
6435 void RtApiDs :: callbackEvent()
6437 if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) {
6438 Sleep( 50 ); // sleep 50 milliseconds
6442 if ( stream_.state == STREAM_CLOSED ) {
6443 errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!";
6444 error( RtAudioError::WARNING );
6448 CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
6449 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6451 // Check if we were draining the stream and signal is finished.
6452 if ( handle->drainCounter > stream_.nBuffers + 2 ) {
6454 stream_.state = STREAM_STOPPING;
6455 if ( handle->internalDrain == false )
6456 SetEvent( handle->condition );
6462 // Invoke user callback to get fresh output data UNLESS we are
6464 if ( handle->drainCounter == 0 ) {
6465 RtAudioCallback callback = (RtAudioCallback) info->callback;
6466 double streamTime = getStreamTime();
6467 RtAudioStreamStatus status = 0;
6468 if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
6469 status |= RTAUDIO_OUTPUT_UNDERFLOW;
6470 handle->xrun[0] = false;
6472 if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
6473 status |= RTAUDIO_INPUT_OVERFLOW;
6474 handle->xrun[1] = false;
6476 int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
6477 stream_.bufferSize, streamTime, status, info->userData );
6478 if ( cbReturnValue == 2 ) {
6479 stream_.state = STREAM_STOPPING;
6480 handle->drainCounter = 2;
6484 else if ( cbReturnValue == 1 ) {
6485 handle->drainCounter = 1;
6486 handle->internalDrain = true;
6491 DWORD currentWritePointer, safeWritePointer;
6492 DWORD currentReadPointer, safeReadPointer;
6493 UINT nextWritePointer;
6495 LPVOID buffer1 = NULL;
6496 LPVOID buffer2 = NULL;
6497 DWORD bufferSize1 = 0;
6498 DWORD bufferSize2 = 0;
6503 MUTEX_LOCK( &stream_.mutex );
6504 if ( stream_.state == STREAM_STOPPED ) {
6505 MUTEX_UNLOCK( &stream_.mutex );
6509 if ( buffersRolling == false ) {
6510 if ( stream_.mode == DUPLEX ) {
6511 //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
6513 // It takes a while for the devices to get rolling. As a result,
6514 // there's no guarantee that the capture and write device pointers
6515 // will move in lockstep. Wait here for both devices to start
6516 // rolling, and then set our buffer pointers accordingly.
6517 // e.g. Crystal Drivers: the capture buffer starts up 5700 to 9600
6518 // bytes later than the write buffer.
6520 // Stub: a serious risk of having a pre-emptive scheduling round
6521 // take place between the two GetCurrentPosition calls... but I'm
6522 // really not sure how to solve the problem. Temporarily boost to
6523 // Realtime priority, maybe; but I'm not sure what priority the
6524 // DirectSound service threads run at. We *should* be roughly
6525 // within a ms or so of correct.
6527 LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6528 LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6530 DWORD startSafeWritePointer, startSafeReadPointer;
6532 result = dsWriteBuffer->GetCurrentPosition( NULL, &startSafeWritePointer );
6533 if ( FAILED( result ) ) {
6534 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
6535 errorText_ = errorStream_.str();
6536 MUTEX_UNLOCK( &stream_.mutex );
6537 error( RtAudioError::SYSTEM_ERROR );
6540 result = dsCaptureBuffer->GetCurrentPosition( NULL, &startSafeReadPointer );
6541 if ( FAILED( result ) ) {
6542 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
6543 errorText_ = errorStream_.str();
6544 MUTEX_UNLOCK( &stream_.mutex );
6545 error( RtAudioError::SYSTEM_ERROR );
6549 result = dsWriteBuffer->GetCurrentPosition( NULL, &safeWritePointer );
6550 if ( FAILED( result ) ) {
6551 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
6552 errorText_ = errorStream_.str();
6553 MUTEX_UNLOCK( &stream_.mutex );
6554 error( RtAudioError::SYSTEM_ERROR );
6557 result = dsCaptureBuffer->GetCurrentPosition( NULL, &safeReadPointer );
6558 if ( FAILED( result ) ) {
6559 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
6560 errorText_ = errorStream_.str();
6561 MUTEX_UNLOCK( &stream_.mutex );
6562 error( RtAudioError::SYSTEM_ERROR );
6565 if ( safeWritePointer != startSafeWritePointer && safeReadPointer != startSafeReadPointer ) break;
6569 //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
6571 handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];
6572 if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];
6573 handle->bufferPointer[1] = safeReadPointer;
6575 else if ( stream_.mode == OUTPUT ) {
6577 // Set the proper nextWritePosition after initial startup.
6578 LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6579 result = dsWriteBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer );
6580 if ( FAILED( result ) ) {
6581 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
6582 errorText_ = errorStream_.str();
6583 MUTEX_UNLOCK( &stream_.mutex );
6584 error( RtAudioError::SYSTEM_ERROR );
6587 handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];
6588 if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];
6591 buffersRolling = true;
6594 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
6596 LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6598 if ( handle->drainCounter > 1 ) { // write zeros to the output stream
6599 bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
6600 bufferBytes *= formatBytes( stream_.userFormat );
6601 memset( stream_.userBuffer[0], 0, bufferBytes );
6604 // Setup parameters and do buffer conversion if necessary.
6605 if ( stream_.doConvertBuffer[0] ) {
6606 buffer = stream_.deviceBuffer;
6607 convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
6608 bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[0];
6609 bufferBytes *= formatBytes( stream_.deviceFormat[0] );
6612 buffer = stream_.userBuffer[0];
6613 bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
6614 bufferBytes *= formatBytes( stream_.userFormat );
6617 // No byte swapping necessary in DirectSound implementation.
6619 // Ahhh ... windoze. 16-bit data is signed but 8-bit data is
6620 // unsigned. So, we need to convert our signed 8-bit data here to
6622 if ( stream_.deviceFormat[0] == RTAUDIO_SINT8 )
6623 for ( int i=0; i<bufferBytes; i++ ) buffer[i] = (unsigned char) ( buffer[i] + 128 );
6625 DWORD dsBufferSize = handle->dsBufferSize[0];
6626 nextWritePointer = handle->bufferPointer[0];
6628 DWORD endWrite, leadPointer;
6630 // Find out where the read and "safe write" pointers are.
6631 result = dsBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer );
6632 if ( FAILED( result ) ) {
6633 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
6634 errorText_ = errorStream_.str();
6635 MUTEX_UNLOCK( &stream_.mutex );
6636 error( RtAudioError::SYSTEM_ERROR );
6640 // We will copy our output buffer into the region between
6641 // safeWritePointer and leadPointer. If leadPointer is not
6642 // beyond the next endWrite position, wait until it is.
6643 leadPointer = safeWritePointer + handle->dsPointerLeadTime[0];
6644 //std::cout << "safeWritePointer = " << safeWritePointer << ", leadPointer = " << leadPointer << ", nextWritePointer = " << nextWritePointer << std::endl;
6645 if ( leadPointer > dsBufferSize ) leadPointer -= dsBufferSize;
6646 if ( leadPointer < nextWritePointer ) leadPointer += dsBufferSize; // unwrap offset
6647 endWrite = nextWritePointer + bufferBytes;
6649 // Check whether the entire write region is behind the play pointer.
6650 if ( leadPointer >= endWrite ) break;
6652 // If we are here, then we must wait until the leadPointer advances
6653 // beyond the end of our next write region. We use the
6654 // Sleep() function to suspend operation until that happens.
6655 double millis = ( endWrite - leadPointer ) * 1000.0;
6656 millis /= ( formatBytes( stream_.deviceFormat[0]) * stream_.nDeviceChannels[0] * stream_.sampleRate);
6657 if ( millis < 1.0 ) millis = 1.0;
6658 Sleep( (DWORD) millis );
6661 if ( dsPointerBetween( nextWritePointer, safeWritePointer, currentWritePointer, dsBufferSize )
6662 || dsPointerBetween( endWrite, safeWritePointer, currentWritePointer, dsBufferSize ) ) {
6663 // We've strayed into the forbidden zone ... resync the read pointer.
6664 handle->xrun[0] = true;
6665 nextWritePointer = safeWritePointer + handle->dsPointerLeadTime[0] - bufferBytes;
6666 if ( nextWritePointer >= dsBufferSize ) nextWritePointer -= dsBufferSize;
6667 handle->bufferPointer[0] = nextWritePointer;
6668 endWrite = nextWritePointer + bufferBytes;
6671 // Lock free space in the buffer
6672 result = dsBuffer->Lock( nextWritePointer, bufferBytes, &buffer1,
6673 &bufferSize1, &buffer2, &bufferSize2, 0 );
6674 if ( FAILED( result ) ) {
6675 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking buffer during playback!";
6676 errorText_ = errorStream_.str();
6677 MUTEX_UNLOCK( &stream_.mutex );
6678 error( RtAudioError::SYSTEM_ERROR );
6682 // Copy our buffer into the DS buffer
6683 CopyMemory( buffer1, buffer, bufferSize1 );
6684 if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 );
6686 // Update our buffer offset and unlock sound buffer
6687 dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
6688 if ( FAILED( result ) ) {
6689 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!";
6690 errorText_ = errorStream_.str();
6691 MUTEX_UNLOCK( &stream_.mutex );
6692 error( RtAudioError::SYSTEM_ERROR );
6695 nextWritePointer = ( nextWritePointer + bufferSize1 + bufferSize2 ) % dsBufferSize;
6696 handle->bufferPointer[0] = nextWritePointer;
6699 // Don't bother draining input
6700 if ( handle->drainCounter ) {
6701 handle->drainCounter++;
6705 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
6707 // Setup parameters.
6708 if ( stream_.doConvertBuffer[1] ) {
6709 buffer = stream_.deviceBuffer;
6710 bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[1];
6711 bufferBytes *= formatBytes( stream_.deviceFormat[1] );
6714 buffer = stream_.userBuffer[1];
6715 bufferBytes = stream_.bufferSize * stream_.nUserChannels[1];
6716 bufferBytes *= formatBytes( stream_.userFormat );
6719 LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6720 long nextReadPointer = handle->bufferPointer[1];
6721 DWORD dsBufferSize = handle->dsBufferSize[1];
6723 // Find out where the write and "safe read" pointers are.
6724 result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer );
6725 if ( FAILED( result ) ) {
6726 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
6727 errorText_ = errorStream_.str();
6728 MUTEX_UNLOCK( &stream_.mutex );
6729 error( RtAudioError::SYSTEM_ERROR );
6733 if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset
6734 DWORD endRead = nextReadPointer + bufferBytes;
6736 // Handling depends on whether we are INPUT or DUPLEX.
6737 // If we're in INPUT mode then waiting is a good thing. If we're in DUPLEX mode,
6738 // then a wait here will drag the write pointers into the forbidden zone.
6740 // In DUPLEX mode, rather than wait, we will back off the read pointer until
6741 // it's in a safe position. This causes dropouts, but it seems to be the only
6742 // practical way to sync up the read and write pointers reliably, given the
6743 // the very complex relationship between phase and increment of the read and write
6746 // In order to minimize audible dropouts in DUPLEX mode, we will
6747 // provide a pre-roll period of 0.5 seconds in which we return
6748 // zeros from the read buffer while the pointers sync up.
6750 if ( stream_.mode == DUPLEX ) {
6751 if ( safeReadPointer < endRead ) {
6752 if ( duplexPrerollBytes <= 0 ) {
6753 // Pre-roll time over. Be more agressive.
6754 int adjustment = endRead-safeReadPointer;
6756 handle->xrun[1] = true;
6758 // - large adjustments: we've probably run out of CPU cycles, so just resync exactly,
6759 // and perform fine adjustments later.
6760 // - small adjustments: back off by twice as much.
6761 if ( adjustment >= 2*bufferBytes )
6762 nextReadPointer = safeReadPointer-2*bufferBytes;
6764 nextReadPointer = safeReadPointer-bufferBytes-adjustment;
6766 if ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;
6770 // In pre=roll time. Just do it.
6771 nextReadPointer = safeReadPointer - bufferBytes;
6772 while ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;
6774 endRead = nextReadPointer + bufferBytes;
6777 else { // mode == INPUT
6778 while ( safeReadPointer < endRead && stream_.callbackInfo.isRunning ) {
6779 // See comments for playback.
6780 double millis = (endRead - safeReadPointer) * 1000.0;
6781 millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate);
6782 if ( millis < 1.0 ) millis = 1.0;
6783 Sleep( (DWORD) millis );
6785 // Wake up and find out where we are now.
6786 result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer );
6787 if ( FAILED( result ) ) {
6788 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
6789 errorText_ = errorStream_.str();
6790 MUTEX_UNLOCK( &stream_.mutex );
6791 error( RtAudioError::SYSTEM_ERROR );
6795 if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset
6799 // Lock free space in the buffer
6800 result = dsBuffer->Lock( nextReadPointer, bufferBytes, &buffer1,
6801 &bufferSize1, &buffer2, &bufferSize2, 0 );
6802 if ( FAILED( result ) ) {
6803 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking capture buffer!";
6804 errorText_ = errorStream_.str();
6805 MUTEX_UNLOCK( &stream_.mutex );
6806 error( RtAudioError::SYSTEM_ERROR );
6810 if ( duplexPrerollBytes <= 0 ) {
6811 // Copy our buffer into the DS buffer
6812 CopyMemory( buffer, buffer1, bufferSize1 );
6813 if ( buffer2 != NULL ) CopyMemory( buffer+bufferSize1, buffer2, bufferSize2 );
6816 memset( buffer, 0, bufferSize1 );
6817 if ( buffer2 != NULL ) memset( buffer + bufferSize1, 0, bufferSize2 );
6818 duplexPrerollBytes -= bufferSize1 + bufferSize2;
6821 // Update our buffer offset and unlock sound buffer
6822 nextReadPointer = ( nextReadPointer + bufferSize1 + bufferSize2 ) % dsBufferSize;
6823 dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
6824 if ( FAILED( result ) ) {
6825 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking capture buffer!";
6826 errorText_ = errorStream_.str();
6827 MUTEX_UNLOCK( &stream_.mutex );
6828 error( RtAudioError::SYSTEM_ERROR );
6831 handle->bufferPointer[1] = nextReadPointer;
6833 // No byte swapping necessary in DirectSound implementation.
6835 // If necessary, convert 8-bit data from unsigned to signed.
6836 if ( stream_.deviceFormat[1] == RTAUDIO_SINT8 )
6837 for ( int j=0; j<bufferBytes; j++ ) buffer[j] = (signed char) ( buffer[j] - 128 );
6839 // Do buffer conversion if necessary.
6840 if ( stream_.doConvertBuffer[1] )
6841 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
6845 MUTEX_UNLOCK( &stream_.mutex );
6846 RtApi::tickStreamTime();
6849 // Definitions for utility functions and callbacks
6850 // specific to the DirectSound implementation.
6852 static unsigned __stdcall callbackHandler( void *ptr )
6854 CallbackInfo *info = (CallbackInfo *) ptr;
6855 RtApiDs *object = (RtApiDs *) info->object;
6856 bool* isRunning = &info->isRunning;
6858 while ( *isRunning == true ) {
6859 object->callbackEvent();
6866 static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
6867 LPCTSTR description,
6871 struct DsProbeData& probeInfo = *(struct DsProbeData*) lpContext;
6872 std::vector<struct DsDevice>& dsDevices = *probeInfo.dsDevices;
6875 bool validDevice = false;
6876 if ( probeInfo.isInput == true ) {
6878 LPDIRECTSOUNDCAPTURE object;
6880 hr = DirectSoundCaptureCreate( lpguid, &object, NULL );
6881 if ( hr != DS_OK ) return TRUE;
6883 caps.dwSize = sizeof(caps);
6884 hr = object->GetCaps( &caps );
6885 if ( hr == DS_OK ) {
6886 if ( caps.dwChannels > 0 && caps.dwFormats > 0 )
6893 LPDIRECTSOUND object;
6894 hr = DirectSoundCreate( lpguid, &object, NULL );
6895 if ( hr != DS_OK ) return TRUE;
6897 caps.dwSize = sizeof(caps);
6898 hr = object->GetCaps( &caps );
6899 if ( hr == DS_OK ) {
6900 if ( caps.dwFlags & DSCAPS_PRIMARYMONO || caps.dwFlags & DSCAPS_PRIMARYSTEREO )
6906 // If good device, then save its name and guid.
6907 std::string name = convertCharPointerToStdString( description );
6908 //if ( name == "Primary Sound Driver" || name == "Primary Sound Capture Driver" )
6909 if ( lpguid == NULL )
6910 name = "Default Device";
6911 if ( validDevice ) {
6912 for ( unsigned int i=0; i<dsDevices.size(); i++ ) {
6913 if ( dsDevices[i].name == name ) {
6914 dsDevices[i].found = true;
6915 if ( probeInfo.isInput ) {
6916 dsDevices[i].id[1] = lpguid;
6917 dsDevices[i].validId[1] = true;
6920 dsDevices[i].id[0] = lpguid;
6921 dsDevices[i].validId[0] = true;
6929 device.found = true;
6930 if ( probeInfo.isInput ) {
6931 device.id[1] = lpguid;
6932 device.validId[1] = true;
6935 device.id[0] = lpguid;
6936 device.validId[0] = true;
6938 dsDevices.push_back( device );
6944 static const char* getErrorString( int code )
6948 case DSERR_ALLOCATED:
6949 return "Already allocated";
6951 case DSERR_CONTROLUNAVAIL:
6952 return "Control unavailable";
6954 case DSERR_INVALIDPARAM:
6955 return "Invalid parameter";
6957 case DSERR_INVALIDCALL:
6958 return "Invalid call";
6961 return "Generic error";
6963 case DSERR_PRIOLEVELNEEDED:
6964 return "Priority level needed";
6966 case DSERR_OUTOFMEMORY:
6967 return "Out of memory";
6969 case DSERR_BADFORMAT:
6970 return "The sample rate or the channel format is not supported";
6972 case DSERR_UNSUPPORTED:
6973 return "Not supported";
6975 case DSERR_NODRIVER:
6978 case DSERR_ALREADYINITIALIZED:
6979 return "Already initialized";
6981 case DSERR_NOAGGREGATION:
6982 return "No aggregation";
6984 case DSERR_BUFFERLOST:
6985 return "Buffer lost";
6987 case DSERR_OTHERAPPHASPRIO:
6988 return "Another application already has priority";
6990 case DSERR_UNINITIALIZED:
6991 return "Uninitialized";
6994 return "DirectSound unknown error";
6997 //******************** End of __WINDOWS_DS__ *********************//
7001 #if defined(__LINUX_ALSA__)
7003 #include <alsa/asoundlib.h>
7006 // A structure to hold various information related to the ALSA API
7009 snd_pcm_t *handles[2];
7012 pthread_cond_t runnable_cv;
7016 :synchronized(false), runnable(false) { xrun[0] = false; xrun[1] = false; }
7019 static void *alsaCallbackHandler( void * ptr );
7021 RtApiAlsa :: RtApiAlsa()
7023 // Nothing to do here.
7026 RtApiAlsa :: ~RtApiAlsa()
7028 if ( stream_.state != STREAM_CLOSED ) closeStream();
7031 unsigned int RtApiAlsa :: getDeviceCount( void )
7033 unsigned nDevices = 0;
7034 int result, subdevice, card;
7038 // Count cards and devices
7040 snd_card_next( &card );
7041 while ( card >= 0 ) {
7042 sprintf( name, "hw:%d", card );
7043 result = snd_ctl_open( &handle, name, 0 );
7045 errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << ".";
7046 errorText_ = errorStream_.str();
7047 error( RtAudioError::WARNING );
7052 result = snd_ctl_pcm_next_device( handle, &subdevice );
7054 errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
7055 errorText_ = errorStream_.str();
7056 error( RtAudioError::WARNING );
7059 if ( subdevice < 0 )
7064 snd_ctl_close( handle );
7065 snd_card_next( &card );
7068 result = snd_ctl_open( &handle, "default", 0 );
7071 snd_ctl_close( handle );
7077 RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device )
7079 RtAudio::DeviceInfo info;
7080 info.probed = false;
7082 unsigned nDevices = 0;
7083 int result, subdevice, card;
7087 // Count cards and devices
7090 snd_card_next( &card );
7091 while ( card >= 0 ) {
7092 sprintf( name, "hw:%d", card );
7093 result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
7095 errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << ".";
7096 errorText_ = errorStream_.str();
7097 error( RtAudioError::WARNING );
7102 result = snd_ctl_pcm_next_device( chandle, &subdevice );
7104 errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
7105 errorText_ = errorStream_.str();
7106 error( RtAudioError::WARNING );
7109 if ( subdevice < 0 ) break;
7110 if ( nDevices == device ) {
7111 sprintf( name, "hw:%d,%d", card, subdevice );
7117 snd_ctl_close( chandle );
7118 snd_card_next( &card );
7121 result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
7122 if ( result == 0 ) {
7123 if ( nDevices == device ) {
7124 strcpy( name, "default" );
7130 if ( nDevices == 0 ) {
7131 errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!";
7132 error( RtAudioError::INVALID_USE );
7136 if ( device >= nDevices ) {
7137 errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!";
7138 error( RtAudioError::INVALID_USE );
7144 // If a stream is already open, we cannot probe the stream devices.
7145 // Thus, use the saved results.
7146 if ( stream_.state != STREAM_CLOSED &&
7147 ( stream_.device[0] == device || stream_.device[1] == device ) ) {
7148 snd_ctl_close( chandle );
7149 if ( device >= devices_.size() ) {
7150 errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened.";
7151 error( RtAudioError::WARNING );
7154 return devices_[ device ];
7157 int openMode = SND_PCM_ASYNC;
7158 snd_pcm_stream_t stream;
7159 snd_pcm_info_t *pcminfo;
7160 snd_pcm_info_alloca( &pcminfo );
7162 snd_pcm_hw_params_t *params;
7163 snd_pcm_hw_params_alloca( ¶ms );
7165 // First try for playback unless default device (which has subdev -1)
7166 stream = SND_PCM_STREAM_PLAYBACK;
7167 snd_pcm_info_set_stream( pcminfo, stream );
7168 if ( subdevice != -1 ) {
7169 snd_pcm_info_set_device( pcminfo, subdevice );
7170 snd_pcm_info_set_subdevice( pcminfo, 0 );
7172 result = snd_ctl_pcm_info( chandle, pcminfo );
7174 // Device probably doesn't support playback.
7179 result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK );
7181 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
7182 errorText_ = errorStream_.str();
7183 error( RtAudioError::WARNING );
7187 // The device is open ... fill the parameter structure.
7188 result = snd_pcm_hw_params_any( phandle, params );
7190 snd_pcm_close( phandle );
7191 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
7192 errorText_ = errorStream_.str();
7193 error( RtAudioError::WARNING );
7197 // Get output channel information.
7199 result = snd_pcm_hw_params_get_channels_max( params, &value );
7201 snd_pcm_close( phandle );
7202 errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << ".";
7203 errorText_ = errorStream_.str();
7204 error( RtAudioError::WARNING );
7207 info.outputChannels = value;
7208 snd_pcm_close( phandle );
7211 stream = SND_PCM_STREAM_CAPTURE;
7212 snd_pcm_info_set_stream( pcminfo, stream );
7214 // Now try for capture unless default device (with subdev = -1)
7215 if ( subdevice != -1 ) {
7216 result = snd_ctl_pcm_info( chandle, pcminfo );
7217 snd_ctl_close( chandle );
7219 // Device probably doesn't support capture.
7220 if ( info.outputChannels == 0 ) return info;
7221 goto probeParameters;
7225 snd_ctl_close( chandle );
7227 result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
7229 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
7230 errorText_ = errorStream_.str();
7231 error( RtAudioError::WARNING );
7232 if ( info.outputChannels == 0 ) return info;
7233 goto probeParameters;
7236 // The device is open ... fill the parameter structure.
7237 result = snd_pcm_hw_params_any( phandle, params );
7239 snd_pcm_close( phandle );
7240 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
7241 errorText_ = errorStream_.str();
7242 error( RtAudioError::WARNING );
7243 if ( info.outputChannels == 0 ) return info;
7244 goto probeParameters;
7247 result = snd_pcm_hw_params_get_channels_max( params, &value );
7249 snd_pcm_close( phandle );
7250 errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << ".";
7251 errorText_ = errorStream_.str();
7252 error( RtAudioError::WARNING );
7253 if ( info.outputChannels == 0 ) return info;
7254 goto probeParameters;
7256 info.inputChannels = value;
7257 snd_pcm_close( phandle );
7259 // If device opens for both playback and capture, we determine the channels.
7260 if ( info.outputChannels > 0 && info.inputChannels > 0 )
7261 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
7263 // ALSA doesn't provide default devices so we'll use the first available one.
7264 if ( device == 0 && info.outputChannels > 0 )
7265 info.isDefaultOutput = true;
7266 if ( device == 0 && info.inputChannels > 0 )
7267 info.isDefaultInput = true;
7270 // At this point, we just need to figure out the supported data
7271 // formats and sample rates. We'll proceed by opening the device in
7272 // the direction with the maximum number of channels, or playback if
7273 // they are equal. This might limit our sample rate options, but so
7276 if ( info.outputChannels >= info.inputChannels )
7277 stream = SND_PCM_STREAM_PLAYBACK;
7279 stream = SND_PCM_STREAM_CAPTURE;
7280 snd_pcm_info_set_stream( pcminfo, stream );
7282 result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
7284 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
7285 errorText_ = errorStream_.str();
7286 error( RtAudioError::WARNING );
7290 // The device is open ... fill the parameter structure.
7291 result = snd_pcm_hw_params_any( phandle, params );
7293 snd_pcm_close( phandle );
7294 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
7295 errorText_ = errorStream_.str();
7296 error( RtAudioError::WARNING );
7300 // Test our discrete set of sample rate values.
7301 info.sampleRates.clear();
7302 for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {
7303 if ( snd_pcm_hw_params_test_rate( phandle, params, SAMPLE_RATES[i], 0 ) == 0 ) {
7304 info.sampleRates.push_back( SAMPLE_RATES[i] );
7306 if ( !info.preferredSampleRate || ( SAMPLE_RATES[i] <= 48000 && SAMPLE_RATES[i] > info.preferredSampleRate ) )
7307 info.preferredSampleRate = SAMPLE_RATES[i];
7310 if ( info.sampleRates.size() == 0 ) {
7311 snd_pcm_close( phandle );
7312 errorStream_ << "RtApiAlsa::getDeviceInfo: no supported sample rates found for device (" << name << ").";
7313 errorText_ = errorStream_.str();
7314 error( RtAudioError::WARNING );
7318 // Probe the supported data formats ... we don't care about endian-ness just yet
7319 snd_pcm_format_t format;
7320 info.nativeFormats = 0;
7321 format = SND_PCM_FORMAT_S8;
7322 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7323 info.nativeFormats |= RTAUDIO_SINT8;
7324 format = SND_PCM_FORMAT_S16;
7325 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7326 info.nativeFormats |= RTAUDIO_SINT16;
7327 format = SND_PCM_FORMAT_S24;
7328 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7329 info.nativeFormats |= RTAUDIO_SINT24;
7330 format = SND_PCM_FORMAT_S32;
7331 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7332 info.nativeFormats |= RTAUDIO_SINT32;
7333 format = SND_PCM_FORMAT_FLOAT;
7334 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7335 info.nativeFormats |= RTAUDIO_FLOAT32;
7336 format = SND_PCM_FORMAT_FLOAT64;
7337 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7338 info.nativeFormats |= RTAUDIO_FLOAT64;
7340 // Check that we have at least one supported format
7341 if ( info.nativeFormats == 0 ) {
7342 snd_pcm_close( phandle );
7343 errorStream_ << "RtApiAlsa::getDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio.";
7344 errorText_ = errorStream_.str();
7345 error( RtAudioError::WARNING );
7349 // Get the device name
7351 result = snd_card_get_name( card, &cardname );
7352 if ( result >= 0 ) {
7353 sprintf( name, "hw:%s,%d", cardname, subdevice );
7358 // That's all ... close the device and return
7359 snd_pcm_close( phandle );
7364 void RtApiAlsa :: saveDeviceInfo( void )
7368 unsigned int nDevices = getDeviceCount();
7369 devices_.resize( nDevices );
7370 for ( unsigned int i=0; i<nDevices; i++ )
7371 devices_[i] = getDeviceInfo( i );
7374 bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
7375 unsigned int firstChannel, unsigned int sampleRate,
7376 RtAudioFormat format, unsigned int *bufferSize,
7377 RtAudio::StreamOptions *options )
7380 #if defined(__RTAUDIO_DEBUG__)
7382 snd_output_stdio_attach(&out, stderr, 0);
7385 // I'm not using the "plug" interface ... too much inconsistent behavior.
7387 unsigned nDevices = 0;
7388 int result, subdevice, card;
7392 if ( options && options->flags & RTAUDIO_ALSA_USE_DEFAULT )
7393 snprintf(name, sizeof(name), "%s", "default");
7395 // Count cards and devices
7397 snd_card_next( &card );
7398 while ( card >= 0 ) {
7399 sprintf( name, "hw:%d", card );
7400 result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
7402 errorStream_ << "RtApiAlsa::probeDeviceOpen: control open, card = " << card << ", " << snd_strerror( result ) << ".";
7403 errorText_ = errorStream_.str();
7408 result = snd_ctl_pcm_next_device( chandle, &subdevice );
7409 if ( result < 0 ) break;
7410 if ( subdevice < 0 ) break;
7411 if ( nDevices == device ) {
7412 sprintf( name, "hw:%d,%d", card, subdevice );
7413 snd_ctl_close( chandle );
7418 snd_ctl_close( chandle );
7419 snd_card_next( &card );
7422 result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
7423 if ( result == 0 ) {
7424 if ( nDevices == device ) {
7425 strcpy( name, "default" );
7431 if ( nDevices == 0 ) {
7432 // This should not happen because a check is made before this function is called.
7433 errorText_ = "RtApiAlsa::probeDeviceOpen: no devices found!";
7437 if ( device >= nDevices ) {
7438 // This should not happen because a check is made before this function is called.
7439 errorText_ = "RtApiAlsa::probeDeviceOpen: device ID is invalid!";
7446 // The getDeviceInfo() function will not work for a device that is
7447 // already open. Thus, we'll probe the system before opening a
7448 // stream and save the results for use by getDeviceInfo().
7449 if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) // only do once
7450 this->saveDeviceInfo();
7452 snd_pcm_stream_t stream;
7453 if ( mode == OUTPUT )
7454 stream = SND_PCM_STREAM_PLAYBACK;
7456 stream = SND_PCM_STREAM_CAPTURE;
7459 int openMode = SND_PCM_ASYNC;
7460 result = snd_pcm_open( &phandle, name, stream, openMode );
7462 if ( mode == OUTPUT )
7463 errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for output.";
7465 errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for input.";
7466 errorText_ = errorStream_.str();
7470 // Fill the parameter structure.
7471 snd_pcm_hw_params_t *hw_params;
7472 snd_pcm_hw_params_alloca( &hw_params );
7473 result = snd_pcm_hw_params_any( phandle, hw_params );
7475 snd_pcm_close( phandle );
7476 errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") parameters, " << snd_strerror( result ) << ".";
7477 errorText_ = errorStream_.str();
7481 #if defined(__RTAUDIO_DEBUG__)
7482 fprintf( stderr, "\nRtApiAlsa: dump hardware params just after device open:\n\n" );
7483 snd_pcm_hw_params_dump( hw_params, out );
7486 // Set access ... check user preference.
7487 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) {
7488 stream_.userInterleaved = false;
7489 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
7491 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
7492 stream_.deviceInterleaved[mode] = true;
7495 stream_.deviceInterleaved[mode] = false;
7498 stream_.userInterleaved = true;
7499 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
7501 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
7502 stream_.deviceInterleaved[mode] = false;
7505 stream_.deviceInterleaved[mode] = true;
7509 snd_pcm_close( phandle );
7510 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") access, " << snd_strerror( result ) << ".";
7511 errorText_ = errorStream_.str();
7515 // Determine how to set the device format.
7516 stream_.userFormat = format;
7517 snd_pcm_format_t deviceFormat = SND_PCM_FORMAT_UNKNOWN;
7519 if ( format == RTAUDIO_SINT8 )
7520 deviceFormat = SND_PCM_FORMAT_S8;
7521 else if ( format == RTAUDIO_SINT16 )
7522 deviceFormat = SND_PCM_FORMAT_S16;
7523 else if ( format == RTAUDIO_SINT24 )
7524 deviceFormat = SND_PCM_FORMAT_S24;
7525 else if ( format == RTAUDIO_SINT32 )
7526 deviceFormat = SND_PCM_FORMAT_S32;
7527 else if ( format == RTAUDIO_FLOAT32 )
7528 deviceFormat = SND_PCM_FORMAT_FLOAT;
7529 else if ( format == RTAUDIO_FLOAT64 )
7530 deviceFormat = SND_PCM_FORMAT_FLOAT64;
7532 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat) == 0) {
7533 stream_.deviceFormat[mode] = format;
7537 // The user requested format is not natively supported by the device.
7538 deviceFormat = SND_PCM_FORMAT_FLOAT64;
7539 if ( snd_pcm_hw_params_test_format( phandle, hw_params, deviceFormat ) == 0 ) {
7540 stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
7544 deviceFormat = SND_PCM_FORMAT_FLOAT;
7545 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7546 stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
7550 deviceFormat = SND_PCM_FORMAT_S32;
7551 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7552 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
7556 deviceFormat = SND_PCM_FORMAT_S24;
7557 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7558 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
7562 deviceFormat = SND_PCM_FORMAT_S16;
7563 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7564 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
7568 deviceFormat = SND_PCM_FORMAT_S8;
7569 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7570 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
7574 // If we get here, no supported format was found.
7575 snd_pcm_close( phandle );
7576 errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device " << device << " data format not supported by RtAudio.";
7577 errorText_ = errorStream_.str();
7581 result = snd_pcm_hw_params_set_format( phandle, hw_params, deviceFormat );
7583 snd_pcm_close( phandle );
7584 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") data format, " << snd_strerror( result ) << ".";
7585 errorText_ = errorStream_.str();
7589 // Determine whether byte-swaping is necessary.
7590 stream_.doByteSwap[mode] = false;
7591 if ( deviceFormat != SND_PCM_FORMAT_S8 ) {
7592 result = snd_pcm_format_cpu_endian( deviceFormat );
7594 stream_.doByteSwap[mode] = true;
7595 else if (result < 0) {
7596 snd_pcm_close( phandle );
7597 errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") endian-ness, " << snd_strerror( result ) << ".";
7598 errorText_ = errorStream_.str();
7603 // Set the sample rate.
7604 result = snd_pcm_hw_params_set_rate_near( phandle, hw_params, (unsigned int*) &sampleRate, 0 );
7606 snd_pcm_close( phandle );
7607 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting sample rate on device (" << name << "), " << snd_strerror( result ) << ".";
7608 errorText_ = errorStream_.str();
7612 // Determine the number of channels for this device. We support a possible
7613 // minimum device channel number > than the value requested by the user.
7614 stream_.nUserChannels[mode] = channels;
7616 result = snd_pcm_hw_params_get_channels_max( hw_params, &value );
7617 unsigned int deviceChannels = value;
7618 if ( result < 0 || deviceChannels < channels + firstChannel ) {
7619 snd_pcm_close( phandle );
7620 errorStream_ << "RtApiAlsa::probeDeviceOpen: requested channel parameters not supported by device (" << name << "), " << snd_strerror( result ) << ".";
7621 errorText_ = errorStream_.str();
7625 result = snd_pcm_hw_params_get_channels_min( hw_params, &value );
7627 snd_pcm_close( phandle );
7628 errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting minimum channels for device (" << name << "), " << snd_strerror( result ) << ".";
7629 errorText_ = errorStream_.str();
7632 deviceChannels = value;
7633 if ( deviceChannels < channels + firstChannel ) deviceChannels = channels + firstChannel;
7634 stream_.nDeviceChannels[mode] = deviceChannels;
7636 // Set the device channels.
7637 result = snd_pcm_hw_params_set_channels( phandle, hw_params, deviceChannels );
7639 snd_pcm_close( phandle );
7640 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting channels for device (" << name << "), " << snd_strerror( result ) << ".";
7641 errorText_ = errorStream_.str();
7645 // Set the buffer (or period) size.
7647 snd_pcm_uframes_t periodSize = *bufferSize;
7648 result = snd_pcm_hw_params_set_period_size_near( phandle, hw_params, &periodSize, &dir );
7650 snd_pcm_close( phandle );
7651 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting period size for device (" << name << "), " << snd_strerror( result ) << ".";
7652 errorText_ = errorStream_.str();
7655 *bufferSize = periodSize;
7657 // Set the buffer number, which in ALSA is referred to as the "period".
7658 unsigned int periods = 0;
7659 if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2;
7660 if ( options && options->numberOfBuffers > 0 ) periods = options->numberOfBuffers;
7661 if ( periods < 2 ) periods = 4; // a fairly safe default value
7662 result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir );
7664 snd_pcm_close( phandle );
7665 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << ".";
7666 errorText_ = errorStream_.str();
7670 // If attempting to setup a duplex stream, the bufferSize parameter
7671 // MUST be the same in both directions!
7672 if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {
7673 snd_pcm_close( phandle );
7674 errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ").";
7675 errorText_ = errorStream_.str();
7679 stream_.bufferSize = *bufferSize;
7681 // Install the hardware configuration
7682 result = snd_pcm_hw_params( phandle, hw_params );
7684 snd_pcm_close( phandle );
7685 errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing hardware configuration on device (" << name << "), " << snd_strerror( result ) << ".";
7686 errorText_ = errorStream_.str();
7690 #if defined(__RTAUDIO_DEBUG__)
7691 fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n");
7692 snd_pcm_hw_params_dump( hw_params, out );
7695 // Set the software configuration to fill buffers with zeros and prevent device stopping on xruns.
7696 snd_pcm_sw_params_t *sw_params = NULL;
7697 snd_pcm_sw_params_alloca( &sw_params );
7698 snd_pcm_sw_params_current( phandle, sw_params );
7699 snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize );
7700 snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, ULONG_MAX );
7701 snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 );
7703 // The following two settings were suggested by Theo Veenker
7704 //snd_pcm_sw_params_set_avail_min( phandle, sw_params, *bufferSize );
7705 //snd_pcm_sw_params_set_xfer_align( phandle, sw_params, 1 );
7707 // here are two options for a fix
7708 //snd_pcm_sw_params_set_silence_size( phandle, sw_params, ULONG_MAX );
7709 snd_pcm_uframes_t val;
7710 snd_pcm_sw_params_get_boundary( sw_params, &val );
7711 snd_pcm_sw_params_set_silence_size( phandle, sw_params, val );
7713 result = snd_pcm_sw_params( phandle, sw_params );
7715 snd_pcm_close( phandle );
7716 errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing software configuration on device (" << name << "), " << snd_strerror( result ) << ".";
7717 errorText_ = errorStream_.str();
7721 #if defined(__RTAUDIO_DEBUG__)
7722 fprintf(stderr, "\nRtApiAlsa: dump software params after installation:\n\n");
7723 snd_pcm_sw_params_dump( sw_params, out );
7726 // Set flags for buffer conversion
7727 stream_.doConvertBuffer[mode] = false;
7728 if ( stream_.userFormat != stream_.deviceFormat[mode] )
7729 stream_.doConvertBuffer[mode] = true;
7730 if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
7731 stream_.doConvertBuffer[mode] = true;
7732 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
7733 stream_.nUserChannels[mode] > 1 )
7734 stream_.doConvertBuffer[mode] = true;
7736 // Allocate the ApiHandle if necessary and then save.
7737 AlsaHandle *apiInfo = 0;
7738 if ( stream_.apiHandle == 0 ) {
7740 apiInfo = (AlsaHandle *) new AlsaHandle;
7742 catch ( std::bad_alloc& ) {
7743 errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating AlsaHandle memory.";
7747 if ( pthread_cond_init( &apiInfo->runnable_cv, NULL ) ) {
7748 errorText_ = "RtApiAlsa::probeDeviceOpen: error initializing pthread condition variable.";
7752 stream_.apiHandle = (void *) apiInfo;
7753 apiInfo->handles[0] = 0;
7754 apiInfo->handles[1] = 0;
7757 apiInfo = (AlsaHandle *) stream_.apiHandle;
7759 apiInfo->handles[mode] = phandle;
7762 // Allocate necessary internal buffers.
7763 unsigned long bufferBytes;
7764 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
7765 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
7766 if ( stream_.userBuffer[mode] == NULL ) {
7767 errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating user buffer memory.";
7771 if ( stream_.doConvertBuffer[mode] ) {
7773 bool makeBuffer = true;
7774 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
7775 if ( mode == INPUT ) {
7776 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
7777 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
7778 if ( bufferBytes <= bytesOut ) makeBuffer = false;
7783 bufferBytes *= *bufferSize;
7784 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
7785 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
7786 if ( stream_.deviceBuffer == NULL ) {
7787 errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating device buffer memory.";
7793 stream_.sampleRate = sampleRate;
7794 stream_.nBuffers = periods;
7795 stream_.device[mode] = device;
7796 stream_.state = STREAM_STOPPED;
7798 // Setup the buffer conversion information structure.
7799 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
7801 // Setup thread if necessary.
7802 if ( stream_.mode == OUTPUT && mode == INPUT ) {
7803 // We had already set up an output stream.
7804 stream_.mode = DUPLEX;
7805 // Link the streams if possible.
7806 apiInfo->synchronized = false;
7807 if ( snd_pcm_link( apiInfo->handles[0], apiInfo->handles[1] ) == 0 )
7808 apiInfo->synchronized = true;
7810 errorText_ = "RtApiAlsa::probeDeviceOpen: unable to synchronize input and output devices.";
7811 error( RtAudioError::WARNING );
7815 stream_.mode = mode;
7817 // Setup callback thread.
7818 stream_.callbackInfo.object = (void *) this;
7820 // Set the thread attributes for joinable and realtime scheduling
7821 // priority (optional). The higher priority will only take affect
7822 // if the program is run as root or suid. Note, under Linux
7823 // processes with CAP_SYS_NICE privilege, a user can change
7824 // scheduling policy and priority (thus need not be root). See
7825 // POSIX "capabilities".
7826 pthread_attr_t attr;
7827 pthread_attr_init( &attr );
7828 pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
7829 #ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
7830 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
7831 stream_.callbackInfo.doRealtime = true;
7832 struct sched_param param;
7833 int priority = options->priority;
7834 int min = sched_get_priority_min( SCHED_RR );
7835 int max = sched_get_priority_max( SCHED_RR );
7836 if ( priority < min ) priority = min;
7837 else if ( priority > max ) priority = max;
7838 param.sched_priority = priority;
7840 // Set the policy BEFORE the priority. Otherwise it fails.
7841 pthread_attr_setschedpolicy(&attr, SCHED_RR);
7842 pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM);
7843 // This is definitely required. Otherwise it fails.
7844 pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED);
7845 pthread_attr_setschedparam(&attr, ¶m);
7848 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
7850 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
7853 stream_.callbackInfo.isRunning = true;
7854 result = pthread_create( &stream_.callbackInfo.thread, &attr, alsaCallbackHandler, &stream_.callbackInfo );
7855 pthread_attr_destroy( &attr );
7857 // Failed. Try instead with default attributes.
7858 result = pthread_create( &stream_.callbackInfo.thread, NULL, alsaCallbackHandler, &stream_.callbackInfo );
7860 stream_.callbackInfo.isRunning = false;
7861 errorText_ = "RtApiAlsa::error creating callback thread!";
7871 pthread_cond_destroy( &apiInfo->runnable_cv );
7872 if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
7873 if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
7875 stream_.apiHandle = 0;
7878 if ( phandle) snd_pcm_close( phandle );
7880 for ( int i=0; i<2; i++ ) {
7881 if ( stream_.userBuffer[i] ) {
7882 free( stream_.userBuffer[i] );
7883 stream_.userBuffer[i] = 0;
7887 if ( stream_.deviceBuffer ) {
7888 free( stream_.deviceBuffer );
7889 stream_.deviceBuffer = 0;
7892 stream_.state = STREAM_CLOSED;
7896 void RtApiAlsa :: closeStream()
7898 if ( stream_.state == STREAM_CLOSED ) {
7899 errorText_ = "RtApiAlsa::closeStream(): no open stream to close!";
7900 error( RtAudioError::WARNING );
7904 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
7905 stream_.callbackInfo.isRunning = false;
7906 MUTEX_LOCK( &stream_.mutex );
7907 if ( stream_.state == STREAM_STOPPED ) {
7908 apiInfo->runnable = true;
7909 pthread_cond_signal( &apiInfo->runnable_cv );
7911 MUTEX_UNLOCK( &stream_.mutex );
7912 pthread_join( stream_.callbackInfo.thread, NULL );
7914 if ( stream_.state == STREAM_RUNNING ) {
7915 stream_.state = STREAM_STOPPED;
7916 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
7917 snd_pcm_drop( apiInfo->handles[0] );
7918 if ( stream_.mode == INPUT || stream_.mode == DUPLEX )
7919 snd_pcm_drop( apiInfo->handles[1] );
7923 pthread_cond_destroy( &apiInfo->runnable_cv );
7924 if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
7925 if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
7927 stream_.apiHandle = 0;
7930 for ( int i=0; i<2; i++ ) {
7931 if ( stream_.userBuffer[i] ) {
7932 free( stream_.userBuffer[i] );
7933 stream_.userBuffer[i] = 0;
7937 if ( stream_.deviceBuffer ) {
7938 free( stream_.deviceBuffer );
7939 stream_.deviceBuffer = 0;
7942 stream_.mode = UNINITIALIZED;
7943 stream_.state = STREAM_CLOSED;
7946 void RtApiAlsa :: startStream()
7948 // This method calls snd_pcm_prepare if the device isn't already in that state.
7951 if ( stream_.state == STREAM_RUNNING ) {
7952 errorText_ = "RtApiAlsa::startStream(): the stream is already running!";
7953 error( RtAudioError::WARNING );
7957 MUTEX_LOCK( &stream_.mutex );
7960 snd_pcm_state_t state;
7961 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
7962 snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
7963 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
7964 state = snd_pcm_state( handle[0] );
7965 if ( state != SND_PCM_STATE_PREPARED ) {
7966 result = snd_pcm_prepare( handle[0] );
7968 errorStream_ << "RtApiAlsa::startStream: error preparing output pcm device, " << snd_strerror( result ) << ".";
7969 errorText_ = errorStream_.str();
7975 if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
7976 result = snd_pcm_drop(handle[1]); // fix to remove stale data received since device has been open
7977 state = snd_pcm_state( handle[1] );
7978 if ( state != SND_PCM_STATE_PREPARED ) {
7979 result = snd_pcm_prepare( handle[1] );
7981 errorStream_ << "RtApiAlsa::startStream: error preparing input pcm device, " << snd_strerror( result ) << ".";
7982 errorText_ = errorStream_.str();
7988 stream_.state = STREAM_RUNNING;
7991 apiInfo->runnable = true;
7992 pthread_cond_signal( &apiInfo->runnable_cv );
7993 MUTEX_UNLOCK( &stream_.mutex );
7995 if ( result >= 0 ) return;
7996 error( RtAudioError::SYSTEM_ERROR );
7999 void RtApiAlsa :: stopStream()
8002 if ( stream_.state == STREAM_STOPPED ) {
8003 errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!";
8004 error( RtAudioError::WARNING );
8008 stream_.state = STREAM_STOPPED;
8009 MUTEX_LOCK( &stream_.mutex );
8012 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
8013 snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
8014 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
8015 if ( apiInfo->synchronized )
8016 result = snd_pcm_drop( handle[0] );
8018 result = snd_pcm_drain( handle[0] );
8020 errorStream_ << "RtApiAlsa::stopStream: error draining output pcm device, " << snd_strerror( result ) << ".";
8021 errorText_ = errorStream_.str();
8026 if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
8027 result = snd_pcm_drop( handle[1] );
8029 errorStream_ << "RtApiAlsa::stopStream: error stopping input pcm device, " << snd_strerror( result ) << ".";
8030 errorText_ = errorStream_.str();
8036 apiInfo->runnable = false; // fixes high CPU usage when stopped
8037 MUTEX_UNLOCK( &stream_.mutex );
8039 if ( result >= 0 ) return;
8040 error( RtAudioError::SYSTEM_ERROR );
8043 void RtApiAlsa :: abortStream()
8046 if ( stream_.state == STREAM_STOPPED ) {
8047 errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!";
8048 error( RtAudioError::WARNING );
8052 stream_.state = STREAM_STOPPED;
8053 MUTEX_LOCK( &stream_.mutex );
8056 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
8057 snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
8058 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
8059 result = snd_pcm_drop( handle[0] );
8061 errorStream_ << "RtApiAlsa::abortStream: error aborting output pcm device, " << snd_strerror( result ) << ".";
8062 errorText_ = errorStream_.str();
8067 if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
8068 result = snd_pcm_drop( handle[1] );
8070 errorStream_ << "RtApiAlsa::abortStream: error aborting input pcm device, " << snd_strerror( result ) << ".";
8071 errorText_ = errorStream_.str();
8077 apiInfo->runnable = false; // fixes high CPU usage when stopped
8078 MUTEX_UNLOCK( &stream_.mutex );
8080 if ( result >= 0 ) return;
8081 error( RtAudioError::SYSTEM_ERROR );
8084 void RtApiAlsa :: callbackEvent()
8086 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
8087 if ( stream_.state == STREAM_STOPPED ) {
8088 MUTEX_LOCK( &stream_.mutex );
8089 while ( !apiInfo->runnable )
8090 pthread_cond_wait( &apiInfo->runnable_cv, &stream_.mutex );
8092 if ( stream_.state != STREAM_RUNNING ) {
8093 MUTEX_UNLOCK( &stream_.mutex );
8096 MUTEX_UNLOCK( &stream_.mutex );
8099 if ( stream_.state == STREAM_CLOSED ) {
8100 errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!";
8101 error( RtAudioError::WARNING );
8105 int doStopStream = 0;
8106 RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
8107 double streamTime = getStreamTime();
8108 RtAudioStreamStatus status = 0;
8109 if ( stream_.mode != INPUT && apiInfo->xrun[0] == true ) {
8110 status |= RTAUDIO_OUTPUT_UNDERFLOW;
8111 apiInfo->xrun[0] = false;
8113 if ( stream_.mode != OUTPUT && apiInfo->xrun[1] == true ) {
8114 status |= RTAUDIO_INPUT_OVERFLOW;
8115 apiInfo->xrun[1] = false;
8117 doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
8118 stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
8120 if ( doStopStream == 2 ) {
8125 MUTEX_LOCK( &stream_.mutex );
8127 // The state might change while waiting on a mutex.
8128 if ( stream_.state == STREAM_STOPPED ) goto unlock;
8134 snd_pcm_sframes_t frames;
8135 RtAudioFormat format;
8136 handle = (snd_pcm_t **) apiInfo->handles;
8138 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
8140 // Setup parameters.
8141 if ( stream_.doConvertBuffer[1] ) {
8142 buffer = stream_.deviceBuffer;
8143 channels = stream_.nDeviceChannels[1];
8144 format = stream_.deviceFormat[1];
8147 buffer = stream_.userBuffer[1];
8148 channels = stream_.nUserChannels[1];
8149 format = stream_.userFormat;
8152 // Read samples from device in interleaved/non-interleaved format.
8153 if ( stream_.deviceInterleaved[1] )
8154 result = snd_pcm_readi( handle[1], buffer, stream_.bufferSize );
8156 void *bufs[channels];
8157 size_t offset = stream_.bufferSize * formatBytes( format );
8158 for ( int i=0; i<channels; i++ )
8159 bufs[i] = (void *) (buffer + (i * offset));
8160 result = snd_pcm_readn( handle[1], bufs, stream_.bufferSize );
8163 if ( result < (int) stream_.bufferSize ) {
8164 // Either an error or overrun occured.
8165 if ( result == -EPIPE ) {
8166 snd_pcm_state_t state = snd_pcm_state( handle[1] );
8167 if ( state == SND_PCM_STATE_XRUN ) {
8168 apiInfo->xrun[1] = true;
8169 result = snd_pcm_prepare( handle[1] );
8171 errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after overrun, " << snd_strerror( result ) << ".";
8172 errorText_ = errorStream_.str();
8176 errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
8177 errorText_ = errorStream_.str();
8181 errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << ".";
8182 errorText_ = errorStream_.str();
8184 error( RtAudioError::WARNING );
8188 // Do byte swapping if necessary.
8189 if ( stream_.doByteSwap[1] )
8190 byteSwapBuffer( buffer, stream_.bufferSize * channels, format );
8192 // Do buffer conversion if necessary.
8193 if ( stream_.doConvertBuffer[1] )
8194 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
8196 // Check stream latency
8197 result = snd_pcm_delay( handle[1], &frames );
8198 if ( result == 0 && frames > 0 ) stream_.latency[1] = frames;
8203 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
8205 // Setup parameters and do buffer conversion if necessary.
8206 if ( stream_.doConvertBuffer[0] ) {
8207 buffer = stream_.deviceBuffer;
8208 convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
8209 channels = stream_.nDeviceChannels[0];
8210 format = stream_.deviceFormat[0];
8213 buffer = stream_.userBuffer[0];
8214 channels = stream_.nUserChannels[0];
8215 format = stream_.userFormat;
8218 // Do byte swapping if necessary.
8219 if ( stream_.doByteSwap[0] )
8220 byteSwapBuffer(buffer, stream_.bufferSize * channels, format);
8222 // Write samples to device in interleaved/non-interleaved format.
8223 if ( stream_.deviceInterleaved[0] )
8224 result = snd_pcm_writei( handle[0], buffer, stream_.bufferSize );
8226 void *bufs[channels];
8227 size_t offset = stream_.bufferSize * formatBytes( format );
8228 for ( int i=0; i<channels; i++ )
8229 bufs[i] = (void *) (buffer + (i * offset));
8230 result = snd_pcm_writen( handle[0], bufs, stream_.bufferSize );
8233 if ( result < (int) stream_.bufferSize ) {
8234 // Either an error or underrun occured.
8235 if ( result == -EPIPE ) {
8236 snd_pcm_state_t state = snd_pcm_state( handle[0] );
8237 if ( state == SND_PCM_STATE_XRUN ) {
8238 apiInfo->xrun[0] = true;
8239 result = snd_pcm_prepare( handle[0] );
8241 errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after underrun, " << snd_strerror( result ) << ".";
8242 errorText_ = errorStream_.str();
8245 errorText_ = "RtApiAlsa::callbackEvent: audio write error, underrun.";
8248 errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
8249 errorText_ = errorStream_.str();
8253 errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << ".";
8254 errorText_ = errorStream_.str();
8256 error( RtAudioError::WARNING );
8260 // Check stream latency
8261 result = snd_pcm_delay( handle[0], &frames );
8262 if ( result == 0 && frames > 0 ) stream_.latency[0] = frames;
8266 MUTEX_UNLOCK( &stream_.mutex );
8268 RtApi::tickStreamTime();
8269 if ( doStopStream == 1 ) this->stopStream();
8272 static void *alsaCallbackHandler( void *ptr )
8274 CallbackInfo *info = (CallbackInfo *) ptr;
8275 RtApiAlsa *object = (RtApiAlsa *) info->object;
8276 bool *isRunning = &info->isRunning;
8278 #ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
8279 if ( info->doRealtime ) {
8280 std::cerr << "RtAudio alsa: " <<
8281 (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") <<
8282 "running realtime scheduling" << std::endl;
8286 while ( *isRunning == true ) {
8287 pthread_testcancel();
8288 object->callbackEvent();
8291 pthread_exit( NULL );
8294 //******************** End of __LINUX_ALSA__ *********************//
8297 #if defined(__LINUX_PULSE__)
8299 // Code written by Peter Meerwald, pmeerw@pmeerw.net
8300 // and Tristan Matthews.
8302 #include <pulse/error.h>
8303 #include <pulse/simple.h>
8306 static const unsigned int SUPPORTED_SAMPLERATES[] = { 8000, 16000, 22050, 32000,
8307 44100, 48000, 96000, 0};
8309 struct rtaudio_pa_format_mapping_t {
8310 RtAudioFormat rtaudio_format;
8311 pa_sample_format_t pa_format;
8314 static const rtaudio_pa_format_mapping_t supported_sampleformats[] = {
8315 {RTAUDIO_SINT16, PA_SAMPLE_S16LE},
8316 {RTAUDIO_SINT32, PA_SAMPLE_S32LE},
8317 {RTAUDIO_FLOAT32, PA_SAMPLE_FLOAT32LE},
8318 {0, PA_SAMPLE_INVALID}};
8320 struct PulseAudioHandle {
8324 pthread_cond_t runnable_cv;
8326 PulseAudioHandle() : s_play(0), s_rec(0), runnable(false) { }
8329 RtApiPulse::~RtApiPulse()
8331 if ( stream_.state != STREAM_CLOSED )
8335 unsigned int RtApiPulse::getDeviceCount( void )
8340 RtAudio::DeviceInfo RtApiPulse::getDeviceInfo( unsigned int /*device*/ )
8342 RtAudio::DeviceInfo info;
8344 info.name = "PulseAudio";
8345 info.outputChannels = 2;
8346 info.inputChannels = 2;
8347 info.duplexChannels = 2;
8348 info.isDefaultOutput = true;
8349 info.isDefaultInput = true;
8351 for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr )
8352 info.sampleRates.push_back( *sr );
8354 info.preferredSampleRate = 48000;
8355 info.nativeFormats = RTAUDIO_SINT16 | RTAUDIO_SINT32 | RTAUDIO_FLOAT32;
8360 static void *pulseaudio_callback( void * user )
8362 CallbackInfo *cbi = static_cast<CallbackInfo *>( user );
8363 RtApiPulse *context = static_cast<RtApiPulse *>( cbi->object );
8364 volatile bool *isRunning = &cbi->isRunning;
8366 #ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
8367 if (cbi->doRealtime) {
8368 std::cerr << "RtAudio pulse: " <<
8369 (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") <<
8370 "running realtime scheduling" << std::endl;
8374 while ( *isRunning ) {
8375 pthread_testcancel();
8376 context->callbackEvent();
8379 pthread_exit( NULL );
8382 void RtApiPulse::closeStream( void )
8384 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8386 stream_.callbackInfo.isRunning = false;
8388 MUTEX_LOCK( &stream_.mutex );
8389 if ( stream_.state == STREAM_STOPPED ) {
8390 pah->runnable = true;
8391 pthread_cond_signal( &pah->runnable_cv );
8393 MUTEX_UNLOCK( &stream_.mutex );
8395 pthread_join( pah->thread, 0 );
8396 if ( pah->s_play ) {
8397 pa_simple_flush( pah->s_play, NULL );
8398 pa_simple_free( pah->s_play );
8401 pa_simple_free( pah->s_rec );
8403 pthread_cond_destroy( &pah->runnable_cv );
8405 stream_.apiHandle = 0;
8408 if ( stream_.userBuffer[0] ) {
8409 free( stream_.userBuffer[0] );
8410 stream_.userBuffer[0] = 0;
8412 if ( stream_.userBuffer[1] ) {
8413 free( stream_.userBuffer[1] );
8414 stream_.userBuffer[1] = 0;
8417 stream_.state = STREAM_CLOSED;
8418 stream_.mode = UNINITIALIZED;
8421 void RtApiPulse::callbackEvent( void )
8423 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8425 if ( stream_.state == STREAM_STOPPED ) {
8426 MUTEX_LOCK( &stream_.mutex );
8427 while ( !pah->runnable )
8428 pthread_cond_wait( &pah->runnable_cv, &stream_.mutex );
8430 if ( stream_.state != STREAM_RUNNING ) {
8431 MUTEX_UNLOCK( &stream_.mutex );
8434 MUTEX_UNLOCK( &stream_.mutex );
8437 if ( stream_.state == STREAM_CLOSED ) {
8438 errorText_ = "RtApiPulse::callbackEvent(): the stream is closed ... "
8439 "this shouldn't happen!";
8440 error( RtAudioError::WARNING );
8444 RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
8445 double streamTime = getStreamTime();
8446 RtAudioStreamStatus status = 0;
8447 int doStopStream = callback( stream_.userBuffer[OUTPUT], stream_.userBuffer[INPUT],
8448 stream_.bufferSize, streamTime, status,
8449 stream_.callbackInfo.userData );
8451 if ( doStopStream == 2 ) {
8456 MUTEX_LOCK( &stream_.mutex );
8457 void *pulse_in = stream_.doConvertBuffer[INPUT] ? stream_.deviceBuffer : stream_.userBuffer[INPUT];
8458 void *pulse_out = stream_.doConvertBuffer[OUTPUT] ? stream_.deviceBuffer : stream_.userBuffer[OUTPUT];
8460 if ( stream_.state != STREAM_RUNNING )
8465 if (stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
8466 if ( stream_.doConvertBuffer[OUTPUT] ) {
8467 convertBuffer( stream_.deviceBuffer,
8468 stream_.userBuffer[OUTPUT],
8469 stream_.convertInfo[OUTPUT] );
8470 bytes = stream_.nDeviceChannels[OUTPUT] * stream_.bufferSize *
8471 formatBytes( stream_.deviceFormat[OUTPUT] );
8473 bytes = stream_.nUserChannels[OUTPUT] * stream_.bufferSize *
8474 formatBytes( stream_.userFormat );
8476 if ( pa_simple_write( pah->s_play, pulse_out, bytes, &pa_error ) < 0 ) {
8477 errorStream_ << "RtApiPulse::callbackEvent: audio write error, " <<
8478 pa_strerror( pa_error ) << ".";
8479 errorText_ = errorStream_.str();
8480 error( RtAudioError::WARNING );
8484 if ( stream_.mode == INPUT || stream_.mode == DUPLEX) {
8485 if ( stream_.doConvertBuffer[INPUT] )
8486 bytes = stream_.nDeviceChannels[INPUT] * stream_.bufferSize *
8487 formatBytes( stream_.deviceFormat[INPUT] );
8489 bytes = stream_.nUserChannels[INPUT] * stream_.bufferSize *
8490 formatBytes( stream_.userFormat );
8492 if ( pa_simple_read( pah->s_rec, pulse_in, bytes, &pa_error ) < 0 ) {
8493 errorStream_ << "RtApiPulse::callbackEvent: audio read error, " <<
8494 pa_strerror( pa_error ) << ".";
8495 errorText_ = errorStream_.str();
8496 error( RtAudioError::WARNING );
8498 if ( stream_.doConvertBuffer[INPUT] ) {
8499 convertBuffer( stream_.userBuffer[INPUT],
8500 stream_.deviceBuffer,
8501 stream_.convertInfo[INPUT] );
8506 MUTEX_UNLOCK( &stream_.mutex );
8507 RtApi::tickStreamTime();
8509 if ( doStopStream == 1 )
8513 void RtApiPulse::startStream( void )
8515 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8517 if ( stream_.state == STREAM_CLOSED ) {
8518 errorText_ = "RtApiPulse::startStream(): the stream is not open!";
8519 error( RtAudioError::INVALID_USE );
8522 if ( stream_.state == STREAM_RUNNING ) {
8523 errorText_ = "RtApiPulse::startStream(): the stream is already running!";
8524 error( RtAudioError::WARNING );
8528 MUTEX_LOCK( &stream_.mutex );
8530 stream_.state = STREAM_RUNNING;
8532 pah->runnable = true;
8533 pthread_cond_signal( &pah->runnable_cv );
8534 MUTEX_UNLOCK( &stream_.mutex );
8537 void RtApiPulse::stopStream( void )
8539 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8541 if ( stream_.state == STREAM_CLOSED ) {
8542 errorText_ = "RtApiPulse::stopStream(): the stream is not open!";
8543 error( RtAudioError::INVALID_USE );
8546 if ( stream_.state == STREAM_STOPPED ) {
8547 errorText_ = "RtApiPulse::stopStream(): the stream is already stopped!";
8548 error( RtAudioError::WARNING );
8552 stream_.state = STREAM_STOPPED;
8553 MUTEX_LOCK( &stream_.mutex );
8555 if ( pah && pah->s_play ) {
8557 if ( pa_simple_drain( pah->s_play, &pa_error ) < 0 ) {
8558 errorStream_ << "RtApiPulse::stopStream: error draining output device, " <<
8559 pa_strerror( pa_error ) << ".";
8560 errorText_ = errorStream_.str();
8561 MUTEX_UNLOCK( &stream_.mutex );
8562 error( RtAudioError::SYSTEM_ERROR );
8567 stream_.state = STREAM_STOPPED;
8568 MUTEX_UNLOCK( &stream_.mutex );
8571 void RtApiPulse::abortStream( void )
8573 PulseAudioHandle *pah = static_cast<PulseAudioHandle*>( stream_.apiHandle );
8575 if ( stream_.state == STREAM_CLOSED ) {
8576 errorText_ = "RtApiPulse::abortStream(): the stream is not open!";
8577 error( RtAudioError::INVALID_USE );
8580 if ( stream_.state == STREAM_STOPPED ) {
8581 errorText_ = "RtApiPulse::abortStream(): the stream is already stopped!";
8582 error( RtAudioError::WARNING );
8586 stream_.state = STREAM_STOPPED;
8587 MUTEX_LOCK( &stream_.mutex );
8589 if ( pah && pah->s_play ) {
8591 if ( pa_simple_flush( pah->s_play, &pa_error ) < 0 ) {
8592 errorStream_ << "RtApiPulse::abortStream: error flushing output device, " <<
8593 pa_strerror( pa_error ) << ".";
8594 errorText_ = errorStream_.str();
8595 MUTEX_UNLOCK( &stream_.mutex );
8596 error( RtAudioError::SYSTEM_ERROR );
8601 stream_.state = STREAM_STOPPED;
8602 MUTEX_UNLOCK( &stream_.mutex );
8605 bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode,
8606 unsigned int channels, unsigned int firstChannel,
8607 unsigned int sampleRate, RtAudioFormat format,
8608 unsigned int *bufferSize, RtAudio::StreamOptions *options )
8610 PulseAudioHandle *pah = 0;
8611 unsigned long bufferBytes = 0;
8614 if ( device != 0 ) return false;
8615 if ( mode != INPUT && mode != OUTPUT ) return false;
8616 if ( channels != 1 && channels != 2 ) {
8617 errorText_ = "RtApiPulse::probeDeviceOpen: unsupported number of channels.";
8620 ss.channels = channels;
8622 if ( firstChannel != 0 ) return false;
8624 bool sr_found = false;
8625 for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr ) {
8626 if ( sampleRate == *sr ) {
8628 stream_.sampleRate = sampleRate;
8629 ss.rate = sampleRate;
8634 errorText_ = "RtApiPulse::probeDeviceOpen: unsupported sample rate.";
8639 for ( const rtaudio_pa_format_mapping_t *sf = supported_sampleformats;
8640 sf->rtaudio_format && sf->pa_format != PA_SAMPLE_INVALID; ++sf ) {
8641 if ( format == sf->rtaudio_format ) {
8643 stream_.userFormat = sf->rtaudio_format;
8644 stream_.deviceFormat[mode] = stream_.userFormat;
8645 ss.format = sf->pa_format;
8649 if ( !sf_found ) { // Use internal data format conversion.
8650 stream_.userFormat = format;
8651 stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
8652 ss.format = PA_SAMPLE_FLOAT32LE;
8655 // Set other stream parameters.
8656 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
8657 else stream_.userInterleaved = true;
8658 stream_.deviceInterleaved[mode] = true;
8659 stream_.nBuffers = 1;
8660 stream_.doByteSwap[mode] = false;
8661 stream_.nUserChannels[mode] = channels;
8662 stream_.nDeviceChannels[mode] = channels + firstChannel;
8663 stream_.channelOffset[mode] = 0;
8664 std::string streamName = "RtAudio";
8666 // Set flags for buffer conversion.
8667 stream_.doConvertBuffer[mode] = false;
8668 if ( stream_.userFormat != stream_.deviceFormat[mode] )
8669 stream_.doConvertBuffer[mode] = true;
8670 if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
8671 stream_.doConvertBuffer[mode] = true;
8673 // Allocate necessary internal buffers.
8674 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
8675 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
8676 if ( stream_.userBuffer[mode] == NULL ) {
8677 errorText_ = "RtApiPulse::probeDeviceOpen: error allocating user buffer memory.";
8680 stream_.bufferSize = *bufferSize;
8682 if ( stream_.doConvertBuffer[mode] ) {
8684 bool makeBuffer = true;
8685 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
8686 if ( mode == INPUT ) {
8687 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
8688 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
8689 if ( bufferBytes <= bytesOut ) makeBuffer = false;
8694 bufferBytes *= *bufferSize;
8695 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
8696 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
8697 if ( stream_.deviceBuffer == NULL ) {
8698 errorText_ = "RtApiPulse::probeDeviceOpen: error allocating device buffer memory.";
8704 stream_.device[mode] = device;
8706 // Setup the buffer conversion information structure.
8707 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
8709 if ( !stream_.apiHandle ) {
8710 PulseAudioHandle *pah = new PulseAudioHandle;
8712 errorText_ = "RtApiPulse::probeDeviceOpen: error allocating memory for handle.";
8716 stream_.apiHandle = pah;
8717 if ( pthread_cond_init( &pah->runnable_cv, NULL ) != 0 ) {
8718 errorText_ = "RtApiPulse::probeDeviceOpen: error creating condition variable.";
8722 pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8725 if ( options && !options->streamName.empty() ) streamName = options->streamName;
8728 pa_buffer_attr buffer_attr;
8729 buffer_attr.fragsize = bufferBytes;
8730 buffer_attr.maxlength = -1;
8732 pah->s_rec = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_RECORD, NULL, "Record", &ss, NULL, &buffer_attr, &error );
8733 if ( !pah->s_rec ) {
8734 errorText_ = "RtApiPulse::probeDeviceOpen: error connecting input to PulseAudio server.";
8739 pah->s_play = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_PLAYBACK, NULL, "Playback", &ss, NULL, NULL, &error );
8740 if ( !pah->s_play ) {
8741 errorText_ = "RtApiPulse::probeDeviceOpen: error connecting output to PulseAudio server.";
8749 if ( stream_.mode == UNINITIALIZED )
8750 stream_.mode = mode;
8751 else if ( stream_.mode == mode )
8754 stream_.mode = DUPLEX;
8756 if ( !stream_.callbackInfo.isRunning ) {
8757 stream_.callbackInfo.object = this;
8759 stream_.state = STREAM_STOPPED;
8760 // Set the thread attributes for joinable and realtime scheduling
8761 // priority (optional). The higher priority will only take affect
8762 // if the program is run as root or suid. Note, under Linux
8763 // processes with CAP_SYS_NICE privilege, a user can change
8764 // scheduling policy and priority (thus need not be root). See
8765 // POSIX "capabilities".
8766 pthread_attr_t attr;
8767 pthread_attr_init( &attr );
8768 pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
8769 #ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
8770 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
8771 stream_.callbackInfo.doRealtime = true;
8772 struct sched_param param;
8773 int priority = options->priority;
8774 int min = sched_get_priority_min( SCHED_RR );
8775 int max = sched_get_priority_max( SCHED_RR );
8776 if ( priority < min ) priority = min;
8777 else if ( priority > max ) priority = max;
8778 param.sched_priority = priority;
8780 // Set the policy BEFORE the priority. Otherwise it fails.
8781 pthread_attr_setschedpolicy(&attr, SCHED_RR);
8782 pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM);
8783 // This is definitely required. Otherwise it fails.
8784 pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED);
8785 pthread_attr_setschedparam(&attr, ¶m);
8788 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
8790 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
8793 stream_.callbackInfo.isRunning = true;
8794 int result = pthread_create( &pah->thread, &attr, pulseaudio_callback, (void *)&stream_.callbackInfo);
8795 pthread_attr_destroy(&attr);
8797 // Failed. Try instead with default attributes.
8798 result = pthread_create( &pah->thread, NULL, pulseaudio_callback, (void *)&stream_.callbackInfo);
8800 stream_.callbackInfo.isRunning = false;
8801 errorText_ = "RtApiPulse::probeDeviceOpen: error creating thread.";
8810 if ( pah && stream_.callbackInfo.isRunning ) {
8811 pthread_cond_destroy( &pah->runnable_cv );
8813 stream_.apiHandle = 0;
8816 for ( int i=0; i<2; i++ ) {
8817 if ( stream_.userBuffer[i] ) {
8818 free( stream_.userBuffer[i] );
8819 stream_.userBuffer[i] = 0;
8823 if ( stream_.deviceBuffer ) {
8824 free( stream_.deviceBuffer );
8825 stream_.deviceBuffer = 0;
8828 stream_.state = STREAM_CLOSED;
8832 //******************** End of __LINUX_PULSE__ *********************//
8835 #if defined(__LINUX_OSS__)
8838 #include <sys/ioctl.h>
8841 #include <sys/soundcard.h>
8845 static void *ossCallbackHandler(void * ptr);
8847 // A structure to hold various information related to the OSS API
8850 int id[2]; // device ids
8853 pthread_cond_t runnable;
8856 :triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
8859 RtApiOss :: RtApiOss()
8861 // Nothing to do here.
8864 RtApiOss :: ~RtApiOss()
8866 if ( stream_.state != STREAM_CLOSED ) closeStream();
8869 unsigned int RtApiOss :: getDeviceCount( void )
8871 int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
8872 if ( mixerfd == -1 ) {
8873 errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'.";
8874 error( RtAudioError::WARNING );
8878 oss_sysinfo sysinfo;
8879 if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) {
8881 errorText_ = "RtApiOss::getDeviceCount: error getting sysinfo, OSS version >= 4.0 is required.";
8882 error( RtAudioError::WARNING );
8887 return sysinfo.numaudios;
8890 RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device )
8892 RtAudio::DeviceInfo info;
8893 info.probed = false;
8895 int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
8896 if ( mixerfd == -1 ) {
8897 errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'.";
8898 error( RtAudioError::WARNING );
8902 oss_sysinfo sysinfo;
8903 int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
8904 if ( result == -1 ) {
8906 errorText_ = "RtApiOss::getDeviceInfo: error getting sysinfo, OSS version >= 4.0 is required.";
8907 error( RtAudioError::WARNING );
8911 unsigned nDevices = sysinfo.numaudios;
8912 if ( nDevices == 0 ) {
8914 errorText_ = "RtApiOss::getDeviceInfo: no devices found!";
8915 error( RtAudioError::INVALID_USE );
8919 if ( device >= nDevices ) {
8921 errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!";
8922 error( RtAudioError::INVALID_USE );
8926 oss_audioinfo ainfo;
8928 result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
8930 if ( result == -1 ) {
8931 errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
8932 errorText_ = errorStream_.str();
8933 error( RtAudioError::WARNING );
8938 if ( ainfo.caps & PCM_CAP_OUTPUT ) info.outputChannels = ainfo.max_channels;
8939 if ( ainfo.caps & PCM_CAP_INPUT ) info.inputChannels = ainfo.max_channels;
8940 if ( ainfo.caps & PCM_CAP_DUPLEX ) {
8941 if ( info.outputChannels > 0 && info.inputChannels > 0 && ainfo.caps & PCM_CAP_DUPLEX )
8942 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
8945 // Probe data formats ... do for input
8946 unsigned long mask = ainfo.iformats;
8947 if ( mask & AFMT_S16_LE || mask & AFMT_S16_BE )
8948 info.nativeFormats |= RTAUDIO_SINT16;
8949 if ( mask & AFMT_S8 )
8950 info.nativeFormats |= RTAUDIO_SINT8;
8951 if ( mask & AFMT_S32_LE || mask & AFMT_S32_BE )
8952 info.nativeFormats |= RTAUDIO_SINT32;
8954 if ( mask & AFMT_FLOAT )
8955 info.nativeFormats |= RTAUDIO_FLOAT32;
8957 if ( mask & AFMT_S24_LE || mask & AFMT_S24_BE )
8958 info.nativeFormats |= RTAUDIO_SINT24;
8960 // Check that we have at least one supported format
8961 if ( info.nativeFormats == 0 ) {
8962 errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio.";
8963 errorText_ = errorStream_.str();
8964 error( RtAudioError::WARNING );
8968 // Probe the supported sample rates.
8969 info.sampleRates.clear();
8970 if ( ainfo.nrates ) {
8971 for ( unsigned int i=0; i<ainfo.nrates; i++ ) {
8972 for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
8973 if ( ainfo.rates[i] == SAMPLE_RATES[k] ) {
8974 info.sampleRates.push_back( SAMPLE_RATES[k] );
8976 if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
8977 info.preferredSampleRate = SAMPLE_RATES[k];
8985 // Check min and max rate values;
8986 for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
8987 if ( ainfo.min_rate <= (int) SAMPLE_RATES[k] && ainfo.max_rate >= (int) SAMPLE_RATES[k] ) {
8988 info.sampleRates.push_back( SAMPLE_RATES[k] );
8990 if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
8991 info.preferredSampleRate = SAMPLE_RATES[k];
8996 if ( info.sampleRates.size() == 0 ) {
8997 errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ").";
8998 errorText_ = errorStream_.str();
8999 error( RtAudioError::WARNING );
9003 info.name = ainfo.name;
9010 bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
9011 unsigned int firstChannel, unsigned int sampleRate,
9012 RtAudioFormat format, unsigned int *bufferSize,
9013 RtAudio::StreamOptions *options )
9015 int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
9016 if ( mixerfd == -1 ) {
9017 errorText_ = "RtApiOss::probeDeviceOpen: error opening '/dev/mixer'.";
9021 oss_sysinfo sysinfo;
9022 int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
9023 if ( result == -1 ) {
9025 errorText_ = "RtApiOss::probeDeviceOpen: error getting sysinfo, OSS version >= 4.0 is required.";
9029 unsigned nDevices = sysinfo.numaudios;
9030 if ( nDevices == 0 ) {
9031 // This should not happen because a check is made before this function is called.
9033 errorText_ = "RtApiOss::probeDeviceOpen: no devices found!";
9037 if ( device >= nDevices ) {
9038 // This should not happen because a check is made before this function is called.
9040 errorText_ = "RtApiOss::probeDeviceOpen: device ID is invalid!";
9044 oss_audioinfo ainfo;
9046 result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
9048 if ( result == -1 ) {
9049 errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
9050 errorText_ = errorStream_.str();
9054 // Check if device supports input or output
9055 if ( ( mode == OUTPUT && !( ainfo.caps & PCM_CAP_OUTPUT ) ) ||
9056 ( mode == INPUT && !( ainfo.caps & PCM_CAP_INPUT ) ) ) {
9057 if ( mode == OUTPUT )
9058 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support output.";
9060 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support input.";
9061 errorText_ = errorStream_.str();
9066 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9067 if ( mode == OUTPUT )
9069 else { // mode == INPUT
9070 if (stream_.mode == OUTPUT && stream_.device[0] == device) {
9071 // We just set the same device for playback ... close and reopen for duplex (OSS only).
9072 close( handle->id[0] );
9074 if ( !( ainfo.caps & PCM_CAP_DUPLEX ) ) {
9075 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support duplex mode.";
9076 errorText_ = errorStream_.str();
9079 // Check that the number previously set channels is the same.
9080 if ( stream_.nUserChannels[0] != channels ) {
9081 errorStream_ << "RtApiOss::probeDeviceOpen: input/output channels must be equal for OSS duplex device (" << ainfo.name << ").";
9082 errorText_ = errorStream_.str();
9091 // Set exclusive access if specified.
9092 if ( options && options->flags & RTAUDIO_HOG_DEVICE ) flags |= O_EXCL;
9094 // Try to open the device.
9096 fd = open( ainfo.devnode, flags, 0 );
9098 if ( errno == EBUSY )
9099 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") is busy.";
9101 errorStream_ << "RtApiOss::probeDeviceOpen: error opening device (" << ainfo.name << ").";
9102 errorText_ = errorStream_.str();
9106 // For duplex operation, specifically set this mode (this doesn't seem to work).
9108 if ( flags | O_RDWR ) {
9109 result = ioctl( fd, SNDCTL_DSP_SETDUPLEX, NULL );
9110 if ( result == -1) {
9111 errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ").";
9112 errorText_ = errorStream_.str();
9118 // Check the device channel support.
9119 stream_.nUserChannels[mode] = channels;
9120 if ( ainfo.max_channels < (int)(channels + firstChannel) ) {
9122 errorStream_ << "RtApiOss::probeDeviceOpen: the device (" << ainfo.name << ") does not support requested channel parameters.";
9123 errorText_ = errorStream_.str();
9127 // Set the number of channels.
9128 int deviceChannels = channels + firstChannel;
9129 result = ioctl( fd, SNDCTL_DSP_CHANNELS, &deviceChannels );
9130 if ( result == -1 || deviceChannels < (int)(channels + firstChannel) ) {
9132 errorStream_ << "RtApiOss::probeDeviceOpen: error setting channel parameters on device (" << ainfo.name << ").";
9133 errorText_ = errorStream_.str();
9136 stream_.nDeviceChannels[mode] = deviceChannels;
9138 // Get the data format mask
9140 result = ioctl( fd, SNDCTL_DSP_GETFMTS, &mask );
9141 if ( result == -1 ) {
9143 errorStream_ << "RtApiOss::probeDeviceOpen: error getting device (" << ainfo.name << ") data formats.";
9144 errorText_ = errorStream_.str();
9148 // Determine how to set the device format.
9149 stream_.userFormat = format;
9150 int deviceFormat = -1;
9151 stream_.doByteSwap[mode] = false;
9152 if ( format == RTAUDIO_SINT8 ) {
9153 if ( mask & AFMT_S8 ) {
9154 deviceFormat = AFMT_S8;
9155 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
9158 else if ( format == RTAUDIO_SINT16 ) {
9159 if ( mask & AFMT_S16_NE ) {
9160 deviceFormat = AFMT_S16_NE;
9161 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
9163 else if ( mask & AFMT_S16_OE ) {
9164 deviceFormat = AFMT_S16_OE;
9165 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
9166 stream_.doByteSwap[mode] = true;
9169 else if ( format == RTAUDIO_SINT24 ) {
9170 if ( mask & AFMT_S24_NE ) {
9171 deviceFormat = AFMT_S24_NE;
9172 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
9174 else if ( mask & AFMT_S24_OE ) {
9175 deviceFormat = AFMT_S24_OE;
9176 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
9177 stream_.doByteSwap[mode] = true;
9180 else if ( format == RTAUDIO_SINT32 ) {
9181 if ( mask & AFMT_S32_NE ) {
9182 deviceFormat = AFMT_S32_NE;
9183 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
9185 else if ( mask & AFMT_S32_OE ) {
9186 deviceFormat = AFMT_S32_OE;
9187 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
9188 stream_.doByteSwap[mode] = true;
9192 if ( deviceFormat == -1 ) {
9193 // The user requested format is not natively supported by the device.
9194 if ( mask & AFMT_S16_NE ) {
9195 deviceFormat = AFMT_S16_NE;
9196 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
9198 else if ( mask & AFMT_S32_NE ) {
9199 deviceFormat = AFMT_S32_NE;
9200 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
9202 else if ( mask & AFMT_S24_NE ) {
9203 deviceFormat = AFMT_S24_NE;
9204 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
9206 else if ( mask & AFMT_S16_OE ) {
9207 deviceFormat = AFMT_S16_OE;
9208 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
9209 stream_.doByteSwap[mode] = true;
9211 else if ( mask & AFMT_S32_OE ) {
9212 deviceFormat = AFMT_S32_OE;
9213 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
9214 stream_.doByteSwap[mode] = true;
9216 else if ( mask & AFMT_S24_OE ) {
9217 deviceFormat = AFMT_S24_OE;
9218 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
9219 stream_.doByteSwap[mode] = true;
9221 else if ( mask & AFMT_S8) {
9222 deviceFormat = AFMT_S8;
9223 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
9227 if ( stream_.deviceFormat[mode] == 0 ) {
9228 // This really shouldn't happen ...
9230 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") data format not supported by RtAudio.";
9231 errorText_ = errorStream_.str();
9235 // Set the data format.
9236 int temp = deviceFormat;
9237 result = ioctl( fd, SNDCTL_DSP_SETFMT, &deviceFormat );
9238 if ( result == -1 || deviceFormat != temp ) {
9240 errorStream_ << "RtApiOss::probeDeviceOpen: error setting data format on device (" << ainfo.name << ").";
9241 errorText_ = errorStream_.str();
9245 // Attempt to set the buffer size. According to OSS, the minimum
9246 // number of buffers is two. The supposed minimum buffer size is 16
9247 // bytes, so that will be our lower bound. The argument to this
9248 // call is in the form 0xMMMMSSSS (hex), where the buffer size (in
9249 // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM.
9250 // We'll check the actual value used near the end of the setup
9252 int ossBufferBytes = *bufferSize * formatBytes( stream_.deviceFormat[mode] ) * deviceChannels;
9253 if ( ossBufferBytes < 16 ) ossBufferBytes = 16;
9255 if ( options ) buffers = options->numberOfBuffers;
9256 if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) buffers = 2;
9257 if ( buffers < 2 ) buffers = 3;
9258 temp = ((int) buffers << 16) + (int)( log10( (double)ossBufferBytes ) / log10( 2.0 ) );
9259 result = ioctl( fd, SNDCTL_DSP_SETFRAGMENT, &temp );
9260 if ( result == -1 ) {
9262 errorStream_ << "RtApiOss::probeDeviceOpen: error setting buffer size on device (" << ainfo.name << ").";
9263 errorText_ = errorStream_.str();
9266 stream_.nBuffers = buffers;
9268 // Save buffer size (in sample frames).
9269 *bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels );
9270 stream_.bufferSize = *bufferSize;
9272 // Set the sample rate.
9273 int srate = sampleRate;
9274 result = ioctl( fd, SNDCTL_DSP_SPEED, &srate );
9275 if ( result == -1 ) {
9277 errorStream_ << "RtApiOss::probeDeviceOpen: error setting sample rate (" << sampleRate << ") on device (" << ainfo.name << ").";
9278 errorText_ = errorStream_.str();
9282 // Verify the sample rate setup worked.
9283 if ( abs( srate - (int)sampleRate ) > 100 ) {
9285 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ").";
9286 errorText_ = errorStream_.str();
9289 stream_.sampleRate = sampleRate;
9291 if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device) {
9292 // We're doing duplex setup here.
9293 stream_.deviceFormat[0] = stream_.deviceFormat[1];
9294 stream_.nDeviceChannels[0] = deviceChannels;
9297 // Set interleaving parameters.
9298 stream_.userInterleaved = true;
9299 stream_.deviceInterleaved[mode] = true;
9300 if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
9301 stream_.userInterleaved = false;
9303 // Set flags for buffer conversion
9304 stream_.doConvertBuffer[mode] = false;
9305 if ( stream_.userFormat != stream_.deviceFormat[mode] )
9306 stream_.doConvertBuffer[mode] = true;
9307 if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
9308 stream_.doConvertBuffer[mode] = true;
9309 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
9310 stream_.nUserChannels[mode] > 1 )
9311 stream_.doConvertBuffer[mode] = true;
9313 // Allocate the stream handles if necessary and then save.
9314 if ( stream_.apiHandle == 0 ) {
9316 handle = new OssHandle;
9318 catch ( std::bad_alloc& ) {
9319 errorText_ = "RtApiOss::probeDeviceOpen: error allocating OssHandle memory.";
9323 if ( pthread_cond_init( &handle->runnable, NULL ) ) {
9324 errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable.";
9328 stream_.apiHandle = (void *) handle;
9331 handle = (OssHandle *) stream_.apiHandle;
9333 handle->id[mode] = fd;
9335 // Allocate necessary internal buffers.
9336 unsigned long bufferBytes;
9337 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
9338 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
9339 if ( stream_.userBuffer[mode] == NULL ) {
9340 errorText_ = "RtApiOss::probeDeviceOpen: error allocating user buffer memory.";
9344 if ( stream_.doConvertBuffer[mode] ) {
9346 bool makeBuffer = true;
9347 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
9348 if ( mode == INPUT ) {
9349 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
9350 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
9351 if ( bufferBytes <= bytesOut ) makeBuffer = false;
9356 bufferBytes *= *bufferSize;
9357 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
9358 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
9359 if ( stream_.deviceBuffer == NULL ) {
9360 errorText_ = "RtApiOss::probeDeviceOpen: error allocating device buffer memory.";
9366 stream_.device[mode] = device;
9367 stream_.state = STREAM_STOPPED;
9369 // Setup the buffer conversion information structure.
9370 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
9372 // Setup thread if necessary.
9373 if ( stream_.mode == OUTPUT && mode == INPUT ) {
9374 // We had already set up an output stream.
9375 stream_.mode = DUPLEX;
9376 if ( stream_.device[0] == device ) handle->id[0] = fd;
9379 stream_.mode = mode;
9381 // Setup callback thread.
9382 stream_.callbackInfo.object = (void *) this;
9384 // Set the thread attributes for joinable and realtime scheduling
9385 // priority. The higher priority will only take affect if the
9386 // program is run as root or suid.
9387 pthread_attr_t attr;
9388 pthread_attr_init( &attr );
9389 pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
9390 #ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
9391 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
9392 stream_.callbackInfo.doRealtime = true;
9393 struct sched_param param;
9394 int priority = options->priority;
9395 int min = sched_get_priority_min( SCHED_RR );
9396 int max = sched_get_priority_max( SCHED_RR );
9397 if ( priority < min ) priority = min;
9398 else if ( priority > max ) priority = max;
9399 param.sched_priority = priority;
9401 // Set the policy BEFORE the priority. Otherwise it fails.
9402 pthread_attr_setschedpolicy(&attr, SCHED_RR);
9403 pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM);
9404 // This is definitely required. Otherwise it fails.
9405 pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED);
9406 pthread_attr_setschedparam(&attr, ¶m);
9409 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
9411 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
9414 stream_.callbackInfo.isRunning = true;
9415 result = pthread_create( &stream_.callbackInfo.thread, &attr, ossCallbackHandler, &stream_.callbackInfo );
9416 pthread_attr_destroy( &attr );
9418 // Failed. Try instead with default attributes.
9419 result = pthread_create( &stream_.callbackInfo.thread, NULL, ossCallbackHandler, &stream_.callbackInfo );
9421 stream_.callbackInfo.isRunning = false;
9422 errorText_ = "RtApiOss::error creating callback thread!";
9432 pthread_cond_destroy( &handle->runnable );
9433 if ( handle->id[0] ) close( handle->id[0] );
9434 if ( handle->id[1] ) close( handle->id[1] );
9436 stream_.apiHandle = 0;
9439 for ( int i=0; i<2; i++ ) {
9440 if ( stream_.userBuffer[i] ) {
9441 free( stream_.userBuffer[i] );
9442 stream_.userBuffer[i] = 0;
9446 if ( stream_.deviceBuffer ) {
9447 free( stream_.deviceBuffer );
9448 stream_.deviceBuffer = 0;
9451 stream_.state = STREAM_CLOSED;
9455 void RtApiOss :: closeStream()
9457 if ( stream_.state == STREAM_CLOSED ) {
9458 errorText_ = "RtApiOss::closeStream(): no open stream to close!";
9459 error( RtAudioError::WARNING );
9463 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9464 stream_.callbackInfo.isRunning = false;
9465 MUTEX_LOCK( &stream_.mutex );
9466 if ( stream_.state == STREAM_STOPPED )
9467 pthread_cond_signal( &handle->runnable );
9468 MUTEX_UNLOCK( &stream_.mutex );
9469 pthread_join( stream_.callbackInfo.thread, NULL );
9471 if ( stream_.state == STREAM_RUNNING ) {
9472 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
9473 ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
9475 ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
9476 stream_.state = STREAM_STOPPED;
9480 pthread_cond_destroy( &handle->runnable );
9481 if ( handle->id[0] ) close( handle->id[0] );
9482 if ( handle->id[1] ) close( handle->id[1] );
9484 stream_.apiHandle = 0;
9487 for ( int i=0; i<2; i++ ) {
9488 if ( stream_.userBuffer[i] ) {
9489 free( stream_.userBuffer[i] );
9490 stream_.userBuffer[i] = 0;
9494 if ( stream_.deviceBuffer ) {
9495 free( stream_.deviceBuffer );
9496 stream_.deviceBuffer = 0;
9499 stream_.mode = UNINITIALIZED;
9500 stream_.state = STREAM_CLOSED;
9503 void RtApiOss :: startStream()
9506 if ( stream_.state == STREAM_RUNNING ) {
9507 errorText_ = "RtApiOss::startStream(): the stream is already running!";
9508 error( RtAudioError::WARNING );
9512 MUTEX_LOCK( &stream_.mutex );
9514 stream_.state = STREAM_RUNNING;
9516 // No need to do anything else here ... OSS automatically starts
9517 // when fed samples.
9519 MUTEX_UNLOCK( &stream_.mutex );
9521 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9522 pthread_cond_signal( &handle->runnable );
9525 void RtApiOss :: stopStream()
9528 if ( stream_.state == STREAM_STOPPED ) {
9529 errorText_ = "RtApiOss::stopStream(): the stream is already stopped!";
9530 error( RtAudioError::WARNING );
9534 MUTEX_LOCK( &stream_.mutex );
9536 // The state might change while waiting on a mutex.
9537 if ( stream_.state == STREAM_STOPPED ) {
9538 MUTEX_UNLOCK( &stream_.mutex );
9543 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9544 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
9546 // Flush the output with zeros a few times.
9549 RtAudioFormat format;
9551 if ( stream_.doConvertBuffer[0] ) {
9552 buffer = stream_.deviceBuffer;
9553 samples = stream_.bufferSize * stream_.nDeviceChannels[0];
9554 format = stream_.deviceFormat[0];
9557 buffer = stream_.userBuffer[0];
9558 samples = stream_.bufferSize * stream_.nUserChannels[0];
9559 format = stream_.userFormat;
9562 memset( buffer, 0, samples * formatBytes(format) );
9563 for ( unsigned int i=0; i<stream_.nBuffers+1; i++ ) {
9564 result = write( handle->id[0], buffer, samples * formatBytes(format) );
9565 if ( result == -1 ) {
9566 errorText_ = "RtApiOss::stopStream: audio write error.";
9567 error( RtAudioError::WARNING );
9571 result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
9572 if ( result == -1 ) {
9573 errorStream_ << "RtApiOss::stopStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
9574 errorText_ = errorStream_.str();
9577 handle->triggered = false;
9580 if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
9581 result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
9582 if ( result == -1 ) {
9583 errorStream_ << "RtApiOss::stopStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
9584 errorText_ = errorStream_.str();
9590 stream_.state = STREAM_STOPPED;
9591 MUTEX_UNLOCK( &stream_.mutex );
9593 if ( result != -1 ) return;
9594 error( RtAudioError::SYSTEM_ERROR );
9597 void RtApiOss :: abortStream()
9600 if ( stream_.state == STREAM_STOPPED ) {
9601 errorText_ = "RtApiOss::abortStream(): the stream is already stopped!";
9602 error( RtAudioError::WARNING );
9606 MUTEX_LOCK( &stream_.mutex );
9608 // The state might change while waiting on a mutex.
9609 if ( stream_.state == STREAM_STOPPED ) {
9610 MUTEX_UNLOCK( &stream_.mutex );
9615 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9616 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
9617 result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
9618 if ( result == -1 ) {
9619 errorStream_ << "RtApiOss::abortStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
9620 errorText_ = errorStream_.str();
9623 handle->triggered = false;
9626 if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
9627 result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
9628 if ( result == -1 ) {
9629 errorStream_ << "RtApiOss::abortStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
9630 errorText_ = errorStream_.str();
9636 stream_.state = STREAM_STOPPED;
9637 MUTEX_UNLOCK( &stream_.mutex );
9639 if ( result != -1 ) return;
9640 error( RtAudioError::SYSTEM_ERROR );
9643 void RtApiOss :: callbackEvent()
9645 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9646 if ( stream_.state == STREAM_STOPPED ) {
9647 MUTEX_LOCK( &stream_.mutex );
9648 pthread_cond_wait( &handle->runnable, &stream_.mutex );
9649 if ( stream_.state != STREAM_RUNNING ) {
9650 MUTEX_UNLOCK( &stream_.mutex );
9653 MUTEX_UNLOCK( &stream_.mutex );
9656 if ( stream_.state == STREAM_CLOSED ) {
9657 errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!";
9658 error( RtAudioError::WARNING );
9662 // Invoke user callback to get fresh output data.
9663 int doStopStream = 0;
9664 RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
9665 double streamTime = getStreamTime();
9666 RtAudioStreamStatus status = 0;
9667 if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
9668 status |= RTAUDIO_OUTPUT_UNDERFLOW;
9669 handle->xrun[0] = false;
9671 if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
9672 status |= RTAUDIO_INPUT_OVERFLOW;
9673 handle->xrun[1] = false;
9675 doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
9676 stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
9677 if ( doStopStream == 2 ) {
9678 this->abortStream();
9682 MUTEX_LOCK( &stream_.mutex );
9684 // The state might change while waiting on a mutex.
9685 if ( stream_.state == STREAM_STOPPED ) goto unlock;
9690 RtAudioFormat format;
9692 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
9694 // Setup parameters and do buffer conversion if necessary.
9695 if ( stream_.doConvertBuffer[0] ) {
9696 buffer = stream_.deviceBuffer;
9697 convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
9698 samples = stream_.bufferSize * stream_.nDeviceChannels[0];
9699 format = stream_.deviceFormat[0];
9702 buffer = stream_.userBuffer[0];
9703 samples = stream_.bufferSize * stream_.nUserChannels[0];
9704 format = stream_.userFormat;
9707 // Do byte swapping if necessary.
9708 if ( stream_.doByteSwap[0] )
9709 byteSwapBuffer( buffer, samples, format );
9711 if ( stream_.mode == DUPLEX && handle->triggered == false ) {
9713 ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
9714 result = write( handle->id[0], buffer, samples * formatBytes(format) );
9715 trig = PCM_ENABLE_INPUT|PCM_ENABLE_OUTPUT;
9716 ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
9717 handle->triggered = true;
9720 // Write samples to device.
9721 result = write( handle->id[0], buffer, samples * formatBytes(format) );
9723 if ( result == -1 ) {
9724 // We'll assume this is an underrun, though there isn't a
9725 // specific means for determining that.
9726 handle->xrun[0] = true;
9727 errorText_ = "RtApiOss::callbackEvent: audio write error.";
9728 error( RtAudioError::WARNING );
9729 // Continue on to input section.
9733 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
9735 // Setup parameters.
9736 if ( stream_.doConvertBuffer[1] ) {
9737 buffer = stream_.deviceBuffer;
9738 samples = stream_.bufferSize * stream_.nDeviceChannels[1];
9739 format = stream_.deviceFormat[1];
9742 buffer = stream_.userBuffer[1];
9743 samples = stream_.bufferSize * stream_.nUserChannels[1];
9744 format = stream_.userFormat;
9747 // Read samples from device.
9748 result = read( handle->id[1], buffer, samples * formatBytes(format) );
9750 if ( result == -1 ) {
9751 // We'll assume this is an overrun, though there isn't a
9752 // specific means for determining that.
9753 handle->xrun[1] = true;
9754 errorText_ = "RtApiOss::callbackEvent: audio read error.";
9755 error( RtAudioError::WARNING );
9759 // Do byte swapping if necessary.
9760 if ( stream_.doByteSwap[1] )
9761 byteSwapBuffer( buffer, samples, format );
9763 // Do buffer conversion if necessary.
9764 if ( stream_.doConvertBuffer[1] )
9765 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
9769 MUTEX_UNLOCK( &stream_.mutex );
9771 RtApi::tickStreamTime();
9772 if ( doStopStream == 1 ) this->stopStream();
9775 static void *ossCallbackHandler( void *ptr )
9777 CallbackInfo *info = (CallbackInfo *) ptr;
9778 RtApiOss *object = (RtApiOss *) info->object;
9779 bool *isRunning = &info->isRunning;
9781 #ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
9782 if (info->doRealtime) {
9783 std::cerr << "RtAudio oss: " <<
9784 (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") <<
9785 "running realtime scheduling" << std::endl;
9789 while ( *isRunning == true ) {
9790 pthread_testcancel();
9791 object->callbackEvent();
9794 pthread_exit( NULL );
9797 //******************** End of __LINUX_OSS__ *********************//
9801 // *************************************************** //
9803 // Protected common (OS-independent) RtAudio methods.
9805 // *************************************************** //
9807 // This method can be modified to control the behavior of error
9808 // message printing.
9809 void RtApi :: error( RtAudioError::Type type )
9811 errorStream_.str(""); // clear the ostringstream
9813 RtAudioErrorCallback errorCallback = (RtAudioErrorCallback) stream_.callbackInfo.errorCallback;
9814 if ( errorCallback ) {
9815 // abortStream() can generate new error messages. Ignore them. Just keep original one.
9817 if ( firstErrorOccurred_ )
9820 firstErrorOccurred_ = true;
9821 const std::string errorMessage = errorText_;
9823 if ( type != RtAudioError::WARNING && stream_.state != STREAM_STOPPED) {
9824 stream_.callbackInfo.isRunning = false; // exit from the thread
9828 errorCallback( type, errorMessage );
9829 firstErrorOccurred_ = false;
9833 if ( type == RtAudioError::WARNING && showWarnings_ == true )
9834 std::cerr << '\n' << errorText_ << "\n\n";
9835 else if ( type != RtAudioError::WARNING )
9836 throw( RtAudioError( errorText_, type ) );
9839 void RtApi :: verifyStream()
9841 if ( stream_.state == STREAM_CLOSED ) {
9842 errorText_ = "RtApi:: a stream is not open!";
9843 error( RtAudioError::INVALID_USE );
9847 void RtApi :: clearStreamInfo()
9849 stream_.mode = UNINITIALIZED;
9850 stream_.state = STREAM_CLOSED;
9851 stream_.sampleRate = 0;
9852 stream_.bufferSize = 0;
9853 stream_.nBuffers = 0;
9854 stream_.userFormat = 0;
9855 stream_.userInterleaved = true;
9856 stream_.streamTime = 0.0;
9857 stream_.apiHandle = 0;
9858 stream_.deviceBuffer = 0;
9859 stream_.callbackInfo.callback = 0;
9860 stream_.callbackInfo.userData = 0;
9861 stream_.callbackInfo.isRunning = false;
9862 stream_.callbackInfo.errorCallback = 0;
9863 for ( int i=0; i<2; i++ ) {
9864 stream_.device[i] = 11111;
9865 stream_.doConvertBuffer[i] = false;
9866 stream_.deviceInterleaved[i] = true;
9867 stream_.doByteSwap[i] = false;
9868 stream_.nUserChannels[i] = 0;
9869 stream_.nDeviceChannels[i] = 0;
9870 stream_.channelOffset[i] = 0;
9871 stream_.deviceFormat[i] = 0;
9872 stream_.latency[i] = 0;
9873 stream_.userBuffer[i] = 0;
9874 stream_.convertInfo[i].channels = 0;
9875 stream_.convertInfo[i].inJump = 0;
9876 stream_.convertInfo[i].outJump = 0;
9877 stream_.convertInfo[i].inFormat = 0;
9878 stream_.convertInfo[i].outFormat = 0;
9879 stream_.convertInfo[i].inOffset.clear();
9880 stream_.convertInfo[i].outOffset.clear();
9884 unsigned int RtApi :: formatBytes( RtAudioFormat format )
9886 if ( format == RTAUDIO_SINT16 )
9888 else if ( format == RTAUDIO_SINT32 || format == RTAUDIO_FLOAT32 )
9890 else if ( format == RTAUDIO_FLOAT64 )
9892 else if ( format == RTAUDIO_SINT24 )
9894 else if ( format == RTAUDIO_SINT8 )
9897 errorText_ = "RtApi::formatBytes: undefined format.";
9898 error( RtAudioError::WARNING );
9903 void RtApi :: setConvertInfo( StreamMode mode, unsigned int firstChannel )
9905 if ( mode == INPUT ) { // convert device to user buffer
9906 stream_.convertInfo[mode].inJump = stream_.nDeviceChannels[1];
9907 stream_.convertInfo[mode].outJump = stream_.nUserChannels[1];
9908 stream_.convertInfo[mode].inFormat = stream_.deviceFormat[1];
9909 stream_.convertInfo[mode].outFormat = stream_.userFormat;
9911 else { // convert user to device buffer
9912 stream_.convertInfo[mode].inJump = stream_.nUserChannels[0];
9913 stream_.convertInfo[mode].outJump = stream_.nDeviceChannels[0];
9914 stream_.convertInfo[mode].inFormat = stream_.userFormat;
9915 stream_.convertInfo[mode].outFormat = stream_.deviceFormat[0];
9918 if ( stream_.convertInfo[mode].inJump < stream_.convertInfo[mode].outJump )
9919 stream_.convertInfo[mode].channels = stream_.convertInfo[mode].inJump;
9921 stream_.convertInfo[mode].channels = stream_.convertInfo[mode].outJump;
9923 // Set up the interleave/deinterleave offsets.
9924 if ( stream_.deviceInterleaved[mode] != stream_.userInterleaved ) {
9925 if ( ( mode == OUTPUT && stream_.deviceInterleaved[mode] ) ||
9926 ( mode == INPUT && stream_.userInterleaved ) ) {
9927 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
9928 stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
9929 stream_.convertInfo[mode].outOffset.push_back( k );
9930 stream_.convertInfo[mode].inJump = 1;
9934 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
9935 stream_.convertInfo[mode].inOffset.push_back( k );
9936 stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
9937 stream_.convertInfo[mode].outJump = 1;
9941 else { // no (de)interleaving
9942 if ( stream_.userInterleaved ) {
9943 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
9944 stream_.convertInfo[mode].inOffset.push_back( k );
9945 stream_.convertInfo[mode].outOffset.push_back( k );
9949 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
9950 stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
9951 stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
9952 stream_.convertInfo[mode].inJump = 1;
9953 stream_.convertInfo[mode].outJump = 1;
9958 // Add channel offset.
9959 if ( firstChannel > 0 ) {
9960 if ( stream_.deviceInterleaved[mode] ) {
9961 if ( mode == OUTPUT ) {
9962 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
9963 stream_.convertInfo[mode].outOffset[k] += firstChannel;
9966 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
9967 stream_.convertInfo[mode].inOffset[k] += firstChannel;
9971 if ( mode == OUTPUT ) {
9972 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
9973 stream_.convertInfo[mode].outOffset[k] += ( firstChannel * stream_.bufferSize );
9976 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
9977 stream_.convertInfo[mode].inOffset[k] += ( firstChannel * stream_.bufferSize );
9983 void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info )
9985 // This function does format conversion, input/output channel compensation, and
9986 // data interleaving/deinterleaving. 24-bit integers are assumed to occupy
9987 // the lower three bytes of a 32-bit integer.
9989 // Clear our device buffer when in/out duplex device channels are different
9990 if ( outBuffer == stream_.deviceBuffer && stream_.mode == DUPLEX &&
9991 ( stream_.nDeviceChannels[0] < stream_.nDeviceChannels[1] ) )
9992 memset( outBuffer, 0, stream_.bufferSize * info.outJump * formatBytes( info.outFormat ) );
9995 if (info.outFormat == RTAUDIO_FLOAT64) {
9997 Float64 *out = (Float64 *)outBuffer;
9999 if (info.inFormat == RTAUDIO_SINT8) {
10000 signed char *in = (signed char *)inBuffer;
10001 scale = 1.0 / 127.5;
10002 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10003 for (j=0; j<info.channels; j++) {
10004 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
10005 out[info.outOffset[j]] += 0.5;
10006 out[info.outOffset[j]] *= scale;
10009 out += info.outJump;
10012 else if (info.inFormat == RTAUDIO_SINT16) {
10013 Int16 *in = (Int16 *)inBuffer;
10014 scale = 1.0 / 32767.5;
10015 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10016 for (j=0; j<info.channels; j++) {
10017 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
10018 out[info.outOffset[j]] += 0.5;
10019 out[info.outOffset[j]] *= scale;
10022 out += info.outJump;
10025 else if (info.inFormat == RTAUDIO_SINT24) {
10026 Int24 *in = (Int24 *)inBuffer;
10027 scale = 1.0 / 8388607.5;
10028 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10029 for (j=0; j<info.channels; j++) {
10030 out[info.outOffset[j]] = (Float64) (in[info.inOffset[j]].asInt());
10031 out[info.outOffset[j]] += 0.5;
10032 out[info.outOffset[j]] *= scale;
10035 out += info.outJump;
10038 else if (info.inFormat == RTAUDIO_SINT32) {
10039 Int32 *in = (Int32 *)inBuffer;
10040 scale = 1.0 / 2147483647.5;
10041 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10042 for (j=0; j<info.channels; j++) {
10043 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
10044 out[info.outOffset[j]] += 0.5;
10045 out[info.outOffset[j]] *= scale;
10048 out += info.outJump;
10051 else if (info.inFormat == RTAUDIO_FLOAT32) {
10052 Float32 *in = (Float32 *)inBuffer;
10053 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10054 for (j=0; j<info.channels; j++) {
10055 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
10058 out += info.outJump;
10061 else if (info.inFormat == RTAUDIO_FLOAT64) {
10062 // Channel compensation and/or (de)interleaving only.
10063 Float64 *in = (Float64 *)inBuffer;
10064 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10065 for (j=0; j<info.channels; j++) {
10066 out[info.outOffset[j]] = in[info.inOffset[j]];
10069 out += info.outJump;
10073 else if (info.outFormat == RTAUDIO_FLOAT32) {
10075 Float32 *out = (Float32 *)outBuffer;
10077 if (info.inFormat == RTAUDIO_SINT8) {
10078 signed char *in = (signed char *)inBuffer;
10079 scale = (Float32) ( 1.0 / 127.5 );
10080 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10081 for (j=0; j<info.channels; j++) {
10082 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
10083 out[info.outOffset[j]] += 0.5;
10084 out[info.outOffset[j]] *= scale;
10087 out += info.outJump;
10090 else if (info.inFormat == RTAUDIO_SINT16) {
10091 Int16 *in = (Int16 *)inBuffer;
10092 scale = (Float32) ( 1.0 / 32767.5 );
10093 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10094 for (j=0; j<info.channels; j++) {
10095 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
10096 out[info.outOffset[j]] += 0.5;
10097 out[info.outOffset[j]] *= scale;
10100 out += info.outJump;
10103 else if (info.inFormat == RTAUDIO_SINT24) {
10104 Int24 *in = (Int24 *)inBuffer;
10105 scale = (Float32) ( 1.0 / 8388607.5 );
10106 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10107 for (j=0; j<info.channels; j++) {
10108 out[info.outOffset[j]] = (Float32) (in[info.inOffset[j]].asInt());
10109 out[info.outOffset[j]] += 0.5;
10110 out[info.outOffset[j]] *= scale;
10113 out += info.outJump;
10116 else if (info.inFormat == RTAUDIO_SINT32) {
10117 Int32 *in = (Int32 *)inBuffer;
10118 scale = (Float32) ( 1.0 / 2147483647.5 );
10119 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10120 for (j=0; j<info.channels; j++) {
10121 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
10122 out[info.outOffset[j]] += 0.5;
10123 out[info.outOffset[j]] *= scale;
10126 out += info.outJump;
10129 else if (info.inFormat == RTAUDIO_FLOAT32) {
10130 // Channel compensation and/or (de)interleaving only.
10131 Float32 *in = (Float32 *)inBuffer;
10132 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10133 for (j=0; j<info.channels; j++) {
10134 out[info.outOffset[j]] = in[info.inOffset[j]];
10137 out += info.outJump;
10140 else if (info.inFormat == RTAUDIO_FLOAT64) {
10141 Float64 *in = (Float64 *)inBuffer;
10142 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10143 for (j=0; j<info.channels; j++) {
10144 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
10147 out += info.outJump;
10151 else if (info.outFormat == RTAUDIO_SINT32) {
10152 Int32 *out = (Int32 *)outBuffer;
10153 if (info.inFormat == RTAUDIO_SINT8) {
10154 signed char *in = (signed char *)inBuffer;
10155 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10156 for (j=0; j<info.channels; j++) {
10157 out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
10158 out[info.outOffset[j]] <<= 24;
10161 out += info.outJump;
10164 else if (info.inFormat == RTAUDIO_SINT16) {
10165 Int16 *in = (Int16 *)inBuffer;
10166 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10167 for (j=0; j<info.channels; j++) {
10168 out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
10169 out[info.outOffset[j]] <<= 16;
10172 out += info.outJump;
10175 else if (info.inFormat == RTAUDIO_SINT24) {
10176 Int24 *in = (Int24 *)inBuffer;
10177 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10178 for (j=0; j<info.channels; j++) {
10179 out[info.outOffset[j]] = (Int32) in[info.inOffset[j]].asInt();
10180 out[info.outOffset[j]] <<= 8;
10183 out += info.outJump;
10186 else if (info.inFormat == RTAUDIO_SINT32) {
10187 // Channel compensation and/or (de)interleaving only.
10188 Int32 *in = (Int32 *)inBuffer;
10189 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10190 for (j=0; j<info.channels; j++) {
10191 out[info.outOffset[j]] = in[info.inOffset[j]];
10194 out += info.outJump;
10197 else if (info.inFormat == RTAUDIO_FLOAT32) {
10198 Float32 *in = (Float32 *)inBuffer;
10199 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10200 for (j=0; j<info.channels; j++) {
10201 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
10204 out += info.outJump;
10207 else if (info.inFormat == RTAUDIO_FLOAT64) {
10208 Float64 *in = (Float64 *)inBuffer;
10209 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10210 for (j=0; j<info.channels; j++) {
10211 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
10214 out += info.outJump;
10218 else if (info.outFormat == RTAUDIO_SINT24) {
10219 Int24 *out = (Int24 *)outBuffer;
10220 if (info.inFormat == RTAUDIO_SINT8) {
10221 signed char *in = (signed char *)inBuffer;
10222 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10223 for (j=0; j<info.channels; j++) {
10224 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] << 16);
10225 //out[info.outOffset[j]] <<= 16;
10228 out += info.outJump;
10231 else if (info.inFormat == RTAUDIO_SINT16) {
10232 Int16 *in = (Int16 *)inBuffer;
10233 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10234 for (j=0; j<info.channels; j++) {
10235 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] << 8);
10236 //out[info.outOffset[j]] <<= 8;
10239 out += info.outJump;
10242 else if (info.inFormat == RTAUDIO_SINT24) {
10243 // Channel compensation and/or (de)interleaving only.
10244 Int24 *in = (Int24 *)inBuffer;
10245 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10246 for (j=0; j<info.channels; j++) {
10247 out[info.outOffset[j]] = in[info.inOffset[j]];
10250 out += info.outJump;
10253 else if (info.inFormat == RTAUDIO_SINT32) {
10254 Int32 *in = (Int32 *)inBuffer;
10255 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10256 for (j=0; j<info.channels; j++) {
10257 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] >> 8);
10258 //out[info.outOffset[j]] >>= 8;
10261 out += info.outJump;
10264 else if (info.inFormat == RTAUDIO_FLOAT32) {
10265 Float32 *in = (Float32 *)inBuffer;
10266 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10267 for (j=0; j<info.channels; j++) {
10268 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
10271 out += info.outJump;
10274 else if (info.inFormat == RTAUDIO_FLOAT64) {
10275 Float64 *in = (Float64 *)inBuffer;
10276 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10277 for (j=0; j<info.channels; j++) {
10278 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
10281 out += info.outJump;
10285 else if (info.outFormat == RTAUDIO_SINT16) {
10286 Int16 *out = (Int16 *)outBuffer;
10287 if (info.inFormat == RTAUDIO_SINT8) {
10288 signed char *in = (signed char *)inBuffer;
10289 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10290 for (j=0; j<info.channels; j++) {
10291 out[info.outOffset[j]] = (Int16) in[info.inOffset[j]];
10292 out[info.outOffset[j]] <<= 8;
10295 out += info.outJump;
10298 else if (info.inFormat == RTAUDIO_SINT16) {
10299 // Channel compensation and/or (de)interleaving only.
10300 Int16 *in = (Int16 *)inBuffer;
10301 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10302 for (j=0; j<info.channels; j++) {
10303 out[info.outOffset[j]] = in[info.inOffset[j]];
10306 out += info.outJump;
10309 else if (info.inFormat == RTAUDIO_SINT24) {
10310 Int24 *in = (Int24 *)inBuffer;
10311 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10312 for (j=0; j<info.channels; j++) {
10313 out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]].asInt() >> 8);
10316 out += info.outJump;
10319 else if (info.inFormat == RTAUDIO_SINT32) {
10320 Int32 *in = (Int32 *)inBuffer;
10321 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10322 for (j=0; j<info.channels; j++) {
10323 out[info.outOffset[j]] = (Int16) ((in[info.inOffset[j]] >> 16) & 0x0000ffff);
10326 out += info.outJump;
10329 else if (info.inFormat == RTAUDIO_FLOAT32) {
10330 Float32 *in = (Float32 *)inBuffer;
10331 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10332 for (j=0; j<info.channels; j++) {
10333 out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
10336 out += info.outJump;
10339 else if (info.inFormat == RTAUDIO_FLOAT64) {
10340 Float64 *in = (Float64 *)inBuffer;
10341 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10342 for (j=0; j<info.channels; j++) {
10343 out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
10346 out += info.outJump;
10350 else if (info.outFormat == RTAUDIO_SINT8) {
10351 signed char *out = (signed char *)outBuffer;
10352 if (info.inFormat == RTAUDIO_SINT8) {
10353 // Channel compensation and/or (de)interleaving only.
10354 signed char *in = (signed char *)inBuffer;
10355 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10356 for (j=0; j<info.channels; j++) {
10357 out[info.outOffset[j]] = in[info.inOffset[j]];
10360 out += info.outJump;
10363 if (info.inFormat == RTAUDIO_SINT16) {
10364 Int16 *in = (Int16 *)inBuffer;
10365 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10366 for (j=0; j<info.channels; j++) {
10367 out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 8) & 0x00ff);
10370 out += info.outJump;
10373 else if (info.inFormat == RTAUDIO_SINT24) {
10374 Int24 *in = (Int24 *)inBuffer;
10375 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10376 for (j=0; j<info.channels; j++) {
10377 out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]].asInt() >> 16);
10380 out += info.outJump;
10383 else if (info.inFormat == RTAUDIO_SINT32) {
10384 Int32 *in = (Int32 *)inBuffer;
10385 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10386 for (j=0; j<info.channels; j++) {
10387 out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 24) & 0x000000ff);
10390 out += info.outJump;
10393 else if (info.inFormat == RTAUDIO_FLOAT32) {
10394 Float32 *in = (Float32 *)inBuffer;
10395 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10396 for (j=0; j<info.channels; j++) {
10397 out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
10400 out += info.outJump;
10403 else if (info.inFormat == RTAUDIO_FLOAT64) {
10404 Float64 *in = (Float64 *)inBuffer;
10405 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10406 for (j=0; j<info.channels; j++) {
10407 out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
10410 out += info.outJump;
10416 //static inline uint16_t bswap_16(uint16_t x) { return (x>>8) | (x<<8); }
10417 //static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); }
10418 //static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); }
10420 void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format )
10426 if ( format == RTAUDIO_SINT16 ) {
10427 for ( unsigned int i=0; i<samples; i++ ) {
10428 // Swap 1st and 2nd bytes.
10433 // Increment 2 bytes.
10437 else if ( format == RTAUDIO_SINT32 ||
10438 format == RTAUDIO_FLOAT32 ) {
10439 for ( unsigned int i=0; i<samples; i++ ) {
10440 // Swap 1st and 4th bytes.
10445 // Swap 2nd and 3rd bytes.
10451 // Increment 3 more bytes.
10455 else if ( format == RTAUDIO_SINT24 ) {
10456 for ( unsigned int i=0; i<samples; i++ ) {
10457 // Swap 1st and 3rd bytes.
10462 // Increment 2 more bytes.
10466 else if ( format == RTAUDIO_FLOAT64 ) {
10467 for ( unsigned int i=0; i<samples; i++ ) {
10468 // Swap 1st and 8th bytes
10473 // Swap 2nd and 7th bytes
10479 // Swap 3rd and 6th bytes
10485 // Swap 4th and 5th bytes
10491 // Increment 5 more bytes.
10497 // Indentation settings for Vim and Emacs
10499 // Local Variables:
10500 // c-basic-offset: 2
10501 // indent-tabs-mode: nil
10504 // vim: et sts=2 sw=2