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, "wmcodecdspuuid" )
3705 //=============================================================================
3707 #define SAFE_RELEASE( objectPtr )\
3710 objectPtr->Release();\
3714 typedef HANDLE ( __stdcall *TAvSetMmThreadCharacteristicsPtr )( LPCWSTR TaskName, LPDWORD TaskIndex );
3716 //-----------------------------------------------------------------------------
3718 // WASAPI dictates stream sample rate, format, channel count, and in some cases, buffer size.
3719 // Therefore we must perform all necessary conversions to user buffers in order to satisfy these
3720 // requirements. WasapiBuffer ring buffers are used between HwIn->UserIn and UserOut->HwOut to
3721 // provide intermediate storage for read / write synchronization.
3735 // sets the length of the internal ring buffer
3736 void setBufferSize( unsigned int bufferSize, unsigned int formatBytes ) {
3739 buffer_ = ( char* ) calloc( bufferSize, formatBytes );
3741 bufferSize_ = bufferSize;
3746 // attempt to push a buffer into the ring buffer at the current "in" index
3747 bool pushBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format )
3749 if ( !buffer || // incoming buffer is NULL
3750 bufferSize == 0 || // incoming buffer has no data
3751 bufferSize > bufferSize_ ) // incoming buffer too large
3756 unsigned int relOutIndex = outIndex_;
3757 unsigned int inIndexEnd = inIndex_ + bufferSize;
3758 if ( relOutIndex < inIndex_ && inIndexEnd >= bufferSize_ ) {
3759 relOutIndex += bufferSize_;
3762 // "in" index can end on the "out" index but cannot begin at it
3763 if ( inIndex_ <= relOutIndex && inIndexEnd > relOutIndex ) {
3764 return false; // not enough space between "in" index and "out" index
3767 // copy buffer from external to internal
3768 int fromZeroSize = inIndex_ + bufferSize - bufferSize_;
3769 fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize;
3770 int fromInSize = bufferSize - fromZeroSize;
3775 memcpy( &( ( char* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( char ) );
3776 memcpy( buffer_, &( ( char* ) buffer )[fromInSize], fromZeroSize * sizeof( char ) );
3778 case RTAUDIO_SINT16:
3779 memcpy( &( ( short* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( short ) );
3780 memcpy( buffer_, &( ( short* ) buffer )[fromInSize], fromZeroSize * sizeof( short ) );
3782 case RTAUDIO_SINT24:
3783 memcpy( &( ( S24* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( S24 ) );
3784 memcpy( buffer_, &( ( S24* ) buffer )[fromInSize], fromZeroSize * sizeof( S24 ) );
3786 case RTAUDIO_SINT32:
3787 memcpy( &( ( int* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( int ) );
3788 memcpy( buffer_, &( ( int* ) buffer )[fromInSize], fromZeroSize * sizeof( int ) );
3790 case RTAUDIO_FLOAT32:
3791 memcpy( &( ( float* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( float ) );
3792 memcpy( buffer_, &( ( float* ) buffer )[fromInSize], fromZeroSize * sizeof( float ) );
3794 case RTAUDIO_FLOAT64:
3795 memcpy( &( ( double* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( double ) );
3796 memcpy( buffer_, &( ( double* ) buffer )[fromInSize], fromZeroSize * sizeof( double ) );
3800 // update "in" index
3801 inIndex_ += bufferSize;
3802 inIndex_ %= bufferSize_;
3807 // attempt to pull a buffer from the ring buffer from the current "out" index
3808 bool pullBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format )
3810 if ( !buffer || // incoming buffer is NULL
3811 bufferSize == 0 || // incoming buffer has no data
3812 bufferSize > bufferSize_ ) // incoming buffer too large
3817 unsigned int relInIndex = inIndex_;
3818 unsigned int outIndexEnd = outIndex_ + bufferSize;
3819 if ( relInIndex < outIndex_ && outIndexEnd >= bufferSize_ ) {
3820 relInIndex += bufferSize_;
3823 // "out" index can begin at and end on the "in" index
3824 if ( outIndex_ < relInIndex && outIndexEnd > relInIndex ) {
3825 return false; // not enough space between "out" index and "in" index
3828 // copy buffer from internal to external
3829 int fromZeroSize = outIndex_ + bufferSize - bufferSize_;
3830 fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize;
3831 int fromOutSize = bufferSize - fromZeroSize;
3836 memcpy( buffer, &( ( char* ) buffer_ )[outIndex_], fromOutSize * sizeof( char ) );
3837 memcpy( &( ( char* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( char ) );
3839 case RTAUDIO_SINT16:
3840 memcpy( buffer, &( ( short* ) buffer_ )[outIndex_], fromOutSize * sizeof( short ) );
3841 memcpy( &( ( short* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( short ) );
3843 case RTAUDIO_SINT24:
3844 memcpy( buffer, &( ( S24* ) buffer_ )[outIndex_], fromOutSize * sizeof( S24 ) );
3845 memcpy( &( ( S24* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( S24 ) );
3847 case RTAUDIO_SINT32:
3848 memcpy( buffer, &( ( int* ) buffer_ )[outIndex_], fromOutSize * sizeof( int ) );
3849 memcpy( &( ( int* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( int ) );
3851 case RTAUDIO_FLOAT32:
3852 memcpy( buffer, &( ( float* ) buffer_ )[outIndex_], fromOutSize * sizeof( float ) );
3853 memcpy( &( ( float* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( float ) );
3855 case RTAUDIO_FLOAT64:
3856 memcpy( buffer, &( ( double* ) buffer_ )[outIndex_], fromOutSize * sizeof( double ) );
3857 memcpy( &( ( double* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( double ) );
3861 // update "out" index
3862 outIndex_ += bufferSize;
3863 outIndex_ %= bufferSize_;
3870 unsigned int bufferSize_;
3871 unsigned int inIndex_;
3872 unsigned int outIndex_;
3875 //-----------------------------------------------------------------------------
3877 // In order to satisfy WASAPI's buffer requirements, we need a means of converting sample rate
3878 // between HW and the user. The WasapiResampler class is used to perform this conversion between
3879 // HwIn->UserIn and UserOut->HwOut during the stream callback loop.
3880 class WasapiResampler
3883 WasapiResampler( bool isFloat, unsigned int bitsPerSample, unsigned int channelCount,
3884 unsigned int inSampleRate, unsigned int outSampleRate )
3885 : _bytesPerSample( bitsPerSample / 8 )
3886 , _channelCount( channelCount )
3887 , _sampleRatio( ( float ) outSampleRate / inSampleRate )
3888 , _transformUnk( NULL )
3889 , _transform( NULL )
3890 , _resamplerProps( NULL )
3891 , _mediaType( NULL )
3892 , _inputMediaType( NULL )
3893 , _outputMediaType( NULL )
3895 // 1. Initialization
3897 MFStartup( MF_VERSION, MFSTARTUP_NOSOCKET );
3899 // 2. Create Resampler Transform Object
3901 CoCreateInstance( CLSID_CResamplerMediaObject, NULL, CLSCTX_INPROC_SERVER,
3902 IID_IUnknown, ( void** ) &_transformUnk );
3904 _transformUnk->QueryInterface( IID_PPV_ARGS( &_transform ) );
3906 _transformUnk->QueryInterface( IID_PPV_ARGS( &_resamplerProps ) );
3907 _resamplerProps->SetHalfFilterLength( 60 ); // best conversion quality
3909 // 3. Specify input / output format
3911 MFCreateMediaType( &_mediaType );
3912 _mediaType->SetGUID( MF_MT_MAJOR_TYPE, MFMediaType_Audio );
3913 _mediaType->SetGUID( MF_MT_SUBTYPE, isFloat ? MFAudioFormat_Float : MFAudioFormat_PCM );
3914 _mediaType->SetUINT32( MF_MT_AUDIO_NUM_CHANNELS, channelCount );
3915 _mediaType->SetUINT32( MF_MT_AUDIO_SAMPLES_PER_SECOND, inSampleRate );
3916 _mediaType->SetUINT32( MF_MT_AUDIO_BLOCK_ALIGNMENT, _bytesPerSample * channelCount );
3917 _mediaType->SetUINT32( MF_MT_AUDIO_AVG_BYTES_PER_SECOND, _bytesPerSample * channelCount * inSampleRate );
3918 _mediaType->SetUINT32( MF_MT_AUDIO_BITS_PER_SAMPLE, bitsPerSample );
3919 _mediaType->SetUINT32( MF_MT_ALL_SAMPLES_INDEPENDENT, TRUE );
3921 MFCreateMediaType( &_inputMediaType );
3922 _mediaType->CopyAllItems( _inputMediaType );
3924 _transform->SetInputType( 0, _inputMediaType, 0 );
3926 MFCreateMediaType( &_outputMediaType );
3927 _mediaType->CopyAllItems( _outputMediaType );
3929 _outputMediaType->SetUINT32( MF_MT_AUDIO_SAMPLES_PER_SECOND, outSampleRate );
3930 _outputMediaType->SetUINT32( MF_MT_AUDIO_AVG_BYTES_PER_SECOND, _bytesPerSample * channelCount * outSampleRate );
3932 _transform->SetOutputType( 0, _outputMediaType, 0 );
3934 // 4. Send stream start messages to Resampler
3936 _transform->ProcessMessage( MFT_MESSAGE_COMMAND_FLUSH, NULL );
3937 _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_BEGIN_STREAMING, NULL );
3938 _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_START_OF_STREAM, NULL );
3943 // 8. Send stream stop messages to Resampler
3945 _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_END_OF_STREAM, NULL );
3946 _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_END_STREAMING, NULL );
3952 SAFE_RELEASE( _transformUnk );
3953 SAFE_RELEASE( _transform );
3954 SAFE_RELEASE( _resamplerProps );
3955 SAFE_RELEASE( _mediaType );
3956 SAFE_RELEASE( _inputMediaType );
3957 SAFE_RELEASE( _outputMediaType );
3960 void Convert( char* outBuffer, const char* inBuffer, unsigned int inSampleCount, unsigned int& outSampleCount )
3962 unsigned int inputBufferSize = _bytesPerSample * _channelCount * inSampleCount;
3963 if ( _sampleRatio == 1 )
3965 // no sample rate conversion required
3966 memcpy( outBuffer, inBuffer, inputBufferSize );
3967 outSampleCount = inSampleCount;
3971 unsigned int outputBufferSize = ( unsigned int ) ceilf( inputBufferSize * _sampleRatio ) + ( _bytesPerSample * _channelCount );
3973 IMFMediaBuffer* rInBuffer;
3974 IMFSample* rInSample;
3975 BYTE* rInByteBuffer = NULL;
3977 // 5. Create Sample object from input data
3979 MFCreateMemoryBuffer( inputBufferSize, &rInBuffer );
3981 rInBuffer->Lock( &rInByteBuffer, NULL, NULL );
3982 memcpy( rInByteBuffer, inBuffer, inputBufferSize );
3983 rInBuffer->Unlock();
3984 rInByteBuffer = NULL;
3986 rInBuffer->SetCurrentLength( inputBufferSize );
3988 MFCreateSample( &rInSample );
3989 rInSample->AddBuffer( rInBuffer );
3991 // 6. Pass input data to Resampler
3993 _transform->ProcessInput( 0, rInSample, 0 );
3995 SAFE_RELEASE( rInBuffer );
3996 SAFE_RELEASE( rInSample );
3998 // 7. Perform sample rate conversion
4000 IMFMediaBuffer* rOutBuffer = NULL;
4001 BYTE* rOutByteBuffer = NULL;
4003 MFT_OUTPUT_DATA_BUFFER rOutDataBuffer;
4005 DWORD rBytes = outputBufferSize; // maximum bytes accepted per ProcessOutput
4007 // 7.1 Create Sample object for output data
4009 memset( &rOutDataBuffer, 0, sizeof rOutDataBuffer );
4010 MFCreateSample( &( rOutDataBuffer.pSample ) );
4011 MFCreateMemoryBuffer( rBytes, &rOutBuffer );
4012 rOutDataBuffer.pSample->AddBuffer( rOutBuffer );
4013 rOutDataBuffer.dwStreamID = 0;
4014 rOutDataBuffer.dwStatus = 0;
4015 rOutDataBuffer.pEvents = NULL;
4017 // 7.2 Get output data from Resampler
4019 if ( _transform->ProcessOutput( 0, 1, &rOutDataBuffer, &rStatus ) == MF_E_TRANSFORM_NEED_MORE_INPUT )
4022 SAFE_RELEASE( rOutBuffer );
4023 SAFE_RELEASE( rOutDataBuffer.pSample );
4027 // 7.3 Write output data to outBuffer
4029 SAFE_RELEASE( rOutBuffer );
4030 rOutDataBuffer.pSample->ConvertToContiguousBuffer( &rOutBuffer );
4031 rOutBuffer->GetCurrentLength( &rBytes );
4033 rOutBuffer->Lock( &rOutByteBuffer, NULL, NULL );
4034 memcpy( outBuffer, rOutByteBuffer, rBytes );
4035 rOutBuffer->Unlock();
4036 rOutByteBuffer = NULL;
4038 outSampleCount = rBytes / _bytesPerSample / _channelCount;
4039 SAFE_RELEASE( rOutBuffer );
4040 SAFE_RELEASE( rOutDataBuffer.pSample );
4044 unsigned int _bytesPerSample;
4045 unsigned int _channelCount;
4048 IUnknown* _transformUnk;
4049 IMFTransform* _transform;
4050 IWMResamplerProps* _resamplerProps;
4051 IMFMediaType* _mediaType;
4052 IMFMediaType* _inputMediaType;
4053 IMFMediaType* _outputMediaType;
4056 //-----------------------------------------------------------------------------
4058 // A structure to hold various information related to the WASAPI implementation.
4061 IAudioClient* captureAudioClient;
4062 IAudioClient* renderAudioClient;
4063 IAudioCaptureClient* captureClient;
4064 IAudioRenderClient* renderClient;
4065 HANDLE captureEvent;
4069 : captureAudioClient( NULL ),
4070 renderAudioClient( NULL ),
4071 captureClient( NULL ),
4072 renderClient( NULL ),
4073 captureEvent( NULL ),
4074 renderEvent( NULL ) {}
4077 //=============================================================================
4079 RtApiWasapi::RtApiWasapi()
4080 : coInitialized_( false ), deviceEnumerator_( NULL )
4082 // WASAPI can run either apartment or multi-threaded
4083 HRESULT hr = CoInitialize( NULL );
4084 if ( !FAILED( hr ) )
4085 coInitialized_ = true;
4087 // Instantiate device enumerator
4088 hr = CoCreateInstance( __uuidof( MMDeviceEnumerator ), NULL,
4089 CLSCTX_ALL, __uuidof( IMMDeviceEnumerator ),
4090 ( void** ) &deviceEnumerator_ );
4092 if ( FAILED( hr ) ) {
4093 errorText_ = "RtApiWasapi::RtApiWasapi: Unable to instantiate device enumerator";
4094 error( RtAudioError::DRIVER_ERROR );
4098 //-----------------------------------------------------------------------------
4100 RtApiWasapi::~RtApiWasapi()
4102 if ( stream_.state != STREAM_CLOSED )
4105 SAFE_RELEASE( deviceEnumerator_ );
4107 // If this object previously called CoInitialize()
4108 if ( coInitialized_ )
4112 //=============================================================================
4114 unsigned int RtApiWasapi::getDeviceCount( void )
4116 unsigned int captureDeviceCount = 0;
4117 unsigned int renderDeviceCount = 0;
4119 IMMDeviceCollection* captureDevices = NULL;
4120 IMMDeviceCollection* renderDevices = NULL;
4122 // Count capture devices
4124 HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
4125 if ( FAILED( hr ) ) {
4126 errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device collection.";
4130 hr = captureDevices->GetCount( &captureDeviceCount );
4131 if ( FAILED( hr ) ) {
4132 errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device count.";
4136 // Count render devices
4137 hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
4138 if ( FAILED( hr ) ) {
4139 errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device collection.";
4143 hr = renderDevices->GetCount( &renderDeviceCount );
4144 if ( FAILED( hr ) ) {
4145 errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device count.";
4150 // release all references
4151 SAFE_RELEASE( captureDevices );
4152 SAFE_RELEASE( renderDevices );
4154 if ( errorText_.empty() )
4155 return captureDeviceCount + renderDeviceCount;
4157 error( RtAudioError::DRIVER_ERROR );
4161 //-----------------------------------------------------------------------------
4163 RtAudio::DeviceInfo RtApiWasapi::getDeviceInfo( unsigned int device )
4165 RtAudio::DeviceInfo info;
4166 unsigned int captureDeviceCount = 0;
4167 unsigned int renderDeviceCount = 0;
4168 std::string defaultDeviceName;
4169 bool isCaptureDevice = false;
4171 PROPVARIANT deviceNameProp;
4172 PROPVARIANT defaultDeviceNameProp;
4174 IMMDeviceCollection* captureDevices = NULL;
4175 IMMDeviceCollection* renderDevices = NULL;
4176 IMMDevice* devicePtr = NULL;
4177 IMMDevice* defaultDevicePtr = NULL;
4178 IAudioClient* audioClient = NULL;
4179 IPropertyStore* devicePropStore = NULL;
4180 IPropertyStore* defaultDevicePropStore = NULL;
4182 WAVEFORMATEX* deviceFormat = NULL;
4183 WAVEFORMATEX* closestMatchFormat = NULL;
4186 info.probed = false;
4188 // Count capture devices
4190 RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
4191 HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
4192 if ( FAILED( hr ) ) {
4193 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device collection.";
4197 hr = captureDevices->GetCount( &captureDeviceCount );
4198 if ( FAILED( hr ) ) {
4199 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device count.";
4203 // Count render devices
4204 hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
4205 if ( FAILED( hr ) ) {
4206 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device collection.";
4210 hr = renderDevices->GetCount( &renderDeviceCount );
4211 if ( FAILED( hr ) ) {
4212 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device count.";
4216 // validate device index
4217 if ( device >= captureDeviceCount + renderDeviceCount ) {
4218 errorText_ = "RtApiWasapi::getDeviceInfo: Invalid device index.";
4219 errorType = RtAudioError::INVALID_USE;
4223 // determine whether index falls within capture or render devices
4224 if ( device >= renderDeviceCount ) {
4225 hr = captureDevices->Item( device - renderDeviceCount, &devicePtr );
4226 if ( FAILED( hr ) ) {
4227 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device handle.";
4230 isCaptureDevice = true;
4233 hr = renderDevices->Item( device, &devicePtr );
4234 if ( FAILED( hr ) ) {
4235 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device handle.";
4238 isCaptureDevice = false;
4241 // get default device name
4242 if ( isCaptureDevice ) {
4243 hr = deviceEnumerator_->GetDefaultAudioEndpoint( eCapture, eConsole, &defaultDevicePtr );
4244 if ( FAILED( hr ) ) {
4245 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default capture device handle.";
4250 hr = deviceEnumerator_->GetDefaultAudioEndpoint( eRender, eConsole, &defaultDevicePtr );
4251 if ( FAILED( hr ) ) {
4252 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default render device handle.";
4257 hr = defaultDevicePtr->OpenPropertyStore( STGM_READ, &defaultDevicePropStore );
4258 if ( FAILED( hr ) ) {
4259 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open default device property store.";
4262 PropVariantInit( &defaultDeviceNameProp );
4264 hr = defaultDevicePropStore->GetValue( PKEY_Device_FriendlyName, &defaultDeviceNameProp );
4265 if ( FAILED( hr ) ) {
4266 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default device property: PKEY_Device_FriendlyName.";
4270 defaultDeviceName = convertCharPointerToStdString(defaultDeviceNameProp.pwszVal);
4273 hr = devicePtr->OpenPropertyStore( STGM_READ, &devicePropStore );
4274 if ( FAILED( hr ) ) {
4275 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open device property store.";
4279 PropVariantInit( &deviceNameProp );
4281 hr = devicePropStore->GetValue( PKEY_Device_FriendlyName, &deviceNameProp );
4282 if ( FAILED( hr ) ) {
4283 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device property: PKEY_Device_FriendlyName.";
4287 info.name =convertCharPointerToStdString(deviceNameProp.pwszVal);
4290 if ( isCaptureDevice ) {
4291 info.isDefaultInput = info.name == defaultDeviceName;
4292 info.isDefaultOutput = false;
4295 info.isDefaultInput = false;
4296 info.isDefaultOutput = info.name == defaultDeviceName;
4300 hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, NULL, ( void** ) &audioClient );
4301 if ( FAILED( hr ) ) {
4302 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device audio client.";
4306 hr = audioClient->GetMixFormat( &deviceFormat );
4307 if ( FAILED( hr ) ) {
4308 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device mix format.";
4312 if ( isCaptureDevice ) {
4313 info.inputChannels = deviceFormat->nChannels;
4314 info.outputChannels = 0;
4315 info.duplexChannels = 0;
4318 info.inputChannels = 0;
4319 info.outputChannels = deviceFormat->nChannels;
4320 info.duplexChannels = 0;
4324 info.sampleRates.clear();
4326 // allow support for all sample rates as we have a built-in sample rate converter
4327 for ( unsigned int i = 0; i < MAX_SAMPLE_RATES; i++ ) {
4328 info.sampleRates.push_back( SAMPLE_RATES[i] );
4330 info.preferredSampleRate = deviceFormat->nSamplesPerSec;
4333 info.nativeFormats = 0;
4335 if ( deviceFormat->wFormatTag == WAVE_FORMAT_IEEE_FLOAT ||
4336 ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
4337 ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_IEEE_FLOAT ) )
4339 if ( deviceFormat->wBitsPerSample == 32 ) {
4340 info.nativeFormats |= RTAUDIO_FLOAT32;
4342 else if ( deviceFormat->wBitsPerSample == 64 ) {
4343 info.nativeFormats |= RTAUDIO_FLOAT64;
4346 else if ( deviceFormat->wFormatTag == WAVE_FORMAT_PCM ||
4347 ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
4348 ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_PCM ) )
4350 if ( deviceFormat->wBitsPerSample == 8 ) {
4351 info.nativeFormats |= RTAUDIO_SINT8;
4353 else if ( deviceFormat->wBitsPerSample == 16 ) {
4354 info.nativeFormats |= RTAUDIO_SINT16;
4356 else if ( deviceFormat->wBitsPerSample == 24 ) {
4357 info.nativeFormats |= RTAUDIO_SINT24;
4359 else if ( deviceFormat->wBitsPerSample == 32 ) {
4360 info.nativeFormats |= RTAUDIO_SINT32;
4368 // release all references
4369 PropVariantClear( &deviceNameProp );
4370 PropVariantClear( &defaultDeviceNameProp );
4372 SAFE_RELEASE( captureDevices );
4373 SAFE_RELEASE( renderDevices );
4374 SAFE_RELEASE( devicePtr );
4375 SAFE_RELEASE( defaultDevicePtr );
4376 SAFE_RELEASE( audioClient );
4377 SAFE_RELEASE( devicePropStore );
4378 SAFE_RELEASE( defaultDevicePropStore );
4380 CoTaskMemFree( deviceFormat );
4381 CoTaskMemFree( closestMatchFormat );
4383 if ( !errorText_.empty() )
4388 //-----------------------------------------------------------------------------
4390 unsigned int RtApiWasapi::getDefaultOutputDevice( void )
4392 for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {
4393 if ( getDeviceInfo( i ).isDefaultOutput ) {
4401 //-----------------------------------------------------------------------------
4403 unsigned int RtApiWasapi::getDefaultInputDevice( void )
4405 for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {
4406 if ( getDeviceInfo( i ).isDefaultInput ) {
4414 //-----------------------------------------------------------------------------
4416 void RtApiWasapi::closeStream( void )
4418 if ( stream_.state == STREAM_CLOSED ) {
4419 errorText_ = "RtApiWasapi::closeStream: No open stream to close.";
4420 error( RtAudioError::WARNING );
4424 if ( stream_.state != STREAM_STOPPED )
4427 // clean up stream memory
4428 SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient )
4429 SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient )
4431 SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureClient )
4432 SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderClient )
4434 if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent )
4435 CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent );
4437 if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent )
4438 CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent );
4440 delete ( WasapiHandle* ) stream_.apiHandle;
4441 stream_.apiHandle = NULL;
4443 for ( int i = 0; i < 2; i++ ) {
4444 if ( stream_.userBuffer[i] ) {
4445 free( stream_.userBuffer[i] );
4446 stream_.userBuffer[i] = 0;
4450 if ( stream_.deviceBuffer ) {
4451 free( stream_.deviceBuffer );
4452 stream_.deviceBuffer = 0;
4455 // update stream state
4456 stream_.state = STREAM_CLOSED;
4459 //-----------------------------------------------------------------------------
4461 void RtApiWasapi::startStream( void )
4465 if ( stream_.state == STREAM_RUNNING ) {
4466 errorText_ = "RtApiWasapi::startStream: The stream is already running.";
4467 error( RtAudioError::WARNING );
4471 // update stream state
4472 stream_.state = STREAM_RUNNING;
4474 // create WASAPI stream thread
4475 stream_.callbackInfo.thread = ( ThreadHandle ) CreateThread( NULL, 0, runWasapiThread, this, CREATE_SUSPENDED, NULL );
4477 if ( !stream_.callbackInfo.thread ) {
4478 errorText_ = "RtApiWasapi::startStream: Unable to instantiate callback thread.";
4479 error( RtAudioError::THREAD_ERROR );
4482 SetThreadPriority( ( void* ) stream_.callbackInfo.thread, stream_.callbackInfo.priority );
4483 ResumeThread( ( void* ) stream_.callbackInfo.thread );
4487 //-----------------------------------------------------------------------------
4489 void RtApiWasapi::stopStream( void )
4493 if ( stream_.state == STREAM_STOPPED ) {
4494 errorText_ = "RtApiWasapi::stopStream: The stream is already stopped.";
4495 error( RtAudioError::WARNING );
4499 // inform stream thread by setting stream state to STREAM_STOPPING
4500 stream_.state = STREAM_STOPPING;
4502 // wait until stream thread is stopped
4503 while( stream_.state != STREAM_STOPPED ) {
4507 // Wait for the last buffer to play before stopping.
4508 Sleep( 1000 * stream_.bufferSize / stream_.sampleRate );
4510 // stop capture client if applicable
4511 if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) {
4512 HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop();
4513 if ( FAILED( hr ) ) {
4514 errorText_ = "RtApiWasapi::stopStream: Unable to stop capture stream.";
4515 error( RtAudioError::DRIVER_ERROR );
4520 // stop render client if applicable
4521 if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) {
4522 HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop();
4523 if ( FAILED( hr ) ) {
4524 errorText_ = "RtApiWasapi::stopStream: Unable to stop render stream.";
4525 error( RtAudioError::DRIVER_ERROR );
4530 // close thread handle
4531 if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
4532 errorText_ = "RtApiWasapi::stopStream: Unable to close callback thread.";
4533 error( RtAudioError::THREAD_ERROR );
4537 stream_.callbackInfo.thread = (ThreadHandle) NULL;
4540 //-----------------------------------------------------------------------------
4542 void RtApiWasapi::abortStream( void )
4546 if ( stream_.state == STREAM_STOPPED ) {
4547 errorText_ = "RtApiWasapi::abortStream: The stream is already stopped.";
4548 error( RtAudioError::WARNING );
4552 // inform stream thread by setting stream state to STREAM_STOPPING
4553 stream_.state = STREAM_STOPPING;
4555 // wait until stream thread is stopped
4556 while ( stream_.state != STREAM_STOPPED ) {
4560 // stop capture client if applicable
4561 if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) {
4562 HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop();
4563 if ( FAILED( hr ) ) {
4564 errorText_ = "RtApiWasapi::abortStream: Unable to stop capture stream.";
4565 error( RtAudioError::DRIVER_ERROR );
4570 // stop render client if applicable
4571 if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) {
4572 HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop();
4573 if ( FAILED( hr ) ) {
4574 errorText_ = "RtApiWasapi::abortStream: Unable to stop render stream.";
4575 error( RtAudioError::DRIVER_ERROR );
4580 // close thread handle
4581 if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
4582 errorText_ = "RtApiWasapi::abortStream: Unable to close callback thread.";
4583 error( RtAudioError::THREAD_ERROR );
4587 stream_.callbackInfo.thread = (ThreadHandle) NULL;
4590 //-----------------------------------------------------------------------------
4592 bool RtApiWasapi::probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
4593 unsigned int firstChannel, unsigned int sampleRate,
4594 RtAudioFormat format, unsigned int* bufferSize,
4595 RtAudio::StreamOptions* options )
4597 bool methodResult = FAILURE;
4598 unsigned int captureDeviceCount = 0;
4599 unsigned int renderDeviceCount = 0;
4601 IMMDeviceCollection* captureDevices = NULL;
4602 IMMDeviceCollection* renderDevices = NULL;
4603 IMMDevice* devicePtr = NULL;
4604 WAVEFORMATEX* deviceFormat = NULL;
4605 unsigned int bufferBytes;
4606 stream_.state = STREAM_STOPPED;
4608 // create API Handle if not already created
4609 if ( !stream_.apiHandle )
4610 stream_.apiHandle = ( void* ) new WasapiHandle();
4612 // Count capture devices
4614 RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
4615 HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
4616 if ( FAILED( hr ) ) {
4617 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device collection.";
4621 hr = captureDevices->GetCount( &captureDeviceCount );
4622 if ( FAILED( hr ) ) {
4623 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device count.";
4627 // Count render devices
4628 hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
4629 if ( FAILED( hr ) ) {
4630 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device collection.";
4634 hr = renderDevices->GetCount( &renderDeviceCount );
4635 if ( FAILED( hr ) ) {
4636 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device count.";
4640 // validate device index
4641 if ( device >= captureDeviceCount + renderDeviceCount ) {
4642 errorType = RtAudioError::INVALID_USE;
4643 errorText_ = "RtApiWasapi::probeDeviceOpen: Invalid device index.";
4647 // determine whether index falls within capture or render devices
4648 if ( device >= renderDeviceCount ) {
4649 if ( mode != INPUT ) {
4650 errorType = RtAudioError::INVALID_USE;
4651 errorText_ = "RtApiWasapi::probeDeviceOpen: Capture device selected as output device.";
4655 // retrieve captureAudioClient from devicePtr
4656 IAudioClient*& captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
4658 hr = captureDevices->Item( device - renderDeviceCount, &devicePtr );
4659 if ( FAILED( hr ) ) {
4660 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device handle.";
4664 hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
4665 NULL, ( void** ) &captureAudioClient );
4666 if ( FAILED( hr ) ) {
4667 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client.";
4671 hr = captureAudioClient->GetMixFormat( &deviceFormat );
4672 if ( FAILED( hr ) ) {
4673 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format.";
4677 stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
4678 captureAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
4681 if ( mode != OUTPUT ) {
4682 errorType = RtAudioError::INVALID_USE;
4683 errorText_ = "RtApiWasapi::probeDeviceOpen: Render device selected as input device.";
4687 // retrieve renderAudioClient from devicePtr
4688 IAudioClient*& renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
4690 hr = renderDevices->Item( device, &devicePtr );
4691 if ( FAILED( hr ) ) {
4692 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device handle.";
4696 hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
4697 NULL, ( void** ) &renderAudioClient );
4698 if ( FAILED( hr ) ) {
4699 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client.";
4703 hr = renderAudioClient->GetMixFormat( &deviceFormat );
4704 if ( FAILED( hr ) ) {
4705 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format.";
4709 stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
4710 renderAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
4714 if ( ( stream_.mode == OUTPUT && mode == INPUT ) ||
4715 ( stream_.mode == INPUT && mode == OUTPUT ) ) {
4716 stream_.mode = DUPLEX;
4719 stream_.mode = mode;
4722 stream_.device[mode] = device;
4723 stream_.doByteSwap[mode] = false;
4724 stream_.sampleRate = sampleRate;
4725 stream_.bufferSize = *bufferSize;
4726 stream_.nBuffers = 1;
4727 stream_.nUserChannels[mode] = channels;
4728 stream_.channelOffset[mode] = firstChannel;
4729 stream_.userFormat = format;
4730 stream_.deviceFormat[mode] = getDeviceInfo( device ).nativeFormats;
4732 if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
4733 stream_.userInterleaved = false;
4735 stream_.userInterleaved = true;
4736 stream_.deviceInterleaved[mode] = true;
4738 // Set flags for buffer conversion.
4739 stream_.doConvertBuffer[mode] = false;
4740 if ( stream_.userFormat != stream_.deviceFormat[mode] ||
4741 stream_.nUserChannels != stream_.nDeviceChannels )
4742 stream_.doConvertBuffer[mode] = true;
4743 else if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
4744 stream_.nUserChannels[mode] > 1 )
4745 stream_.doConvertBuffer[mode] = true;
4747 if ( stream_.doConvertBuffer[mode] )
4748 setConvertInfo( mode, 0 );
4750 // Allocate necessary internal buffers
4751 bufferBytes = stream_.nUserChannels[mode] * stream_.bufferSize * formatBytes( stream_.userFormat );
4753 stream_.userBuffer[mode] = ( char* ) calloc( bufferBytes, 1 );
4754 if ( !stream_.userBuffer[mode] ) {
4755 errorType = RtAudioError::MEMORY_ERROR;
4756 errorText_ = "RtApiWasapi::probeDeviceOpen: Error allocating user buffer memory.";
4760 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME )
4761 stream_.callbackInfo.priority = 15;
4763 stream_.callbackInfo.priority = 0;
4765 ///! TODO: RTAUDIO_MINIMIZE_LATENCY // Provide stream buffers directly to callback
4766 ///! TODO: RTAUDIO_HOG_DEVICE // Exclusive mode
4768 methodResult = SUCCESS;
4772 SAFE_RELEASE( captureDevices );
4773 SAFE_RELEASE( renderDevices );
4774 SAFE_RELEASE( devicePtr );
4775 CoTaskMemFree( deviceFormat );
4777 // if method failed, close the stream
4778 if ( methodResult == FAILURE )
4781 if ( !errorText_.empty() )
4783 return methodResult;
4786 //=============================================================================
4788 DWORD WINAPI RtApiWasapi::runWasapiThread( void* wasapiPtr )
4791 ( ( RtApiWasapi* ) wasapiPtr )->wasapiThread();
4796 DWORD WINAPI RtApiWasapi::stopWasapiThread( void* wasapiPtr )
4799 ( ( RtApiWasapi* ) wasapiPtr )->stopStream();
4804 DWORD WINAPI RtApiWasapi::abortWasapiThread( void* wasapiPtr )
4807 ( ( RtApiWasapi* ) wasapiPtr )->abortStream();
4812 //-----------------------------------------------------------------------------
4814 void RtApiWasapi::wasapiThread()
4816 // as this is a new thread, we must CoInitialize it
4817 CoInitialize( NULL );
4821 IAudioClient* captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
4822 IAudioClient* renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
4823 IAudioCaptureClient* captureClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureClient;
4824 IAudioRenderClient* renderClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderClient;
4825 HANDLE captureEvent = ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent;
4826 HANDLE renderEvent = ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent;
4828 WAVEFORMATEX* captureFormat = NULL;
4829 WAVEFORMATEX* renderFormat = NULL;
4830 float captureSrRatio = 0.0f;
4831 float renderSrRatio = 0.0f;
4832 WasapiBuffer captureBuffer;
4833 WasapiBuffer renderBuffer;
4835 // declare local stream variables
4836 RtAudioCallback callback = ( RtAudioCallback ) stream_.callbackInfo.callback;
4837 BYTE* streamBuffer = NULL;
4838 unsigned long captureFlags = 0;
4839 unsigned int bufferFrameCount = 0;
4840 unsigned int numFramesPadding = 0;
4841 unsigned int convBufferSize = 0;
4842 bool callbackPushed = false;
4843 bool callbackPulled = false;
4844 bool callbackStopped = false;
4845 int callbackResult = 0;
4847 // convBuffer is used to store converted buffers between WASAPI and the user
4848 char* convBuffer = NULL;
4849 unsigned int convBuffSize = 0;
4850 unsigned int deviceBuffSize = 0;
4853 RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
4855 // Attempt to assign "Pro Audio" characteristic to thread
4856 HMODULE AvrtDll = LoadLibrary( (LPCTSTR) "AVRT.dll" );
4858 DWORD taskIndex = 0;
4859 TAvSetMmThreadCharacteristicsPtr AvSetMmThreadCharacteristicsPtr = ( TAvSetMmThreadCharacteristicsPtr ) GetProcAddress( AvrtDll, "AvSetMmThreadCharacteristicsW" );
4860 AvSetMmThreadCharacteristicsPtr( L"Pro Audio", &taskIndex );
4861 FreeLibrary( AvrtDll );
4864 // start capture stream if applicable
4865 if ( captureAudioClient ) {
4866 hr = captureAudioClient->GetMixFormat( &captureFormat );
4867 if ( FAILED( hr ) ) {
4868 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
4872 captureSrRatio = ( ( float ) captureFormat->nSamplesPerSec / stream_.sampleRate );
4874 // initialize capture stream according to desire buffer size
4875 float desiredBufferSize = stream_.bufferSize * captureSrRatio;
4876 REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / captureFormat->nSamplesPerSec );
4878 if ( !captureClient ) {
4879 hr = captureAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
4880 AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
4881 desiredBufferPeriod,
4882 desiredBufferPeriod,
4885 if ( FAILED( hr ) ) {
4886 errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize capture audio client.";
4890 hr = captureAudioClient->GetService( __uuidof( IAudioCaptureClient ),
4891 ( void** ) &captureClient );
4892 if ( FAILED( hr ) ) {
4893 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture client handle.";
4897 // configure captureEvent to trigger on every available capture buffer
4898 captureEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
4899 if ( !captureEvent ) {
4900 errorType = RtAudioError::SYSTEM_ERROR;
4901 errorText_ = "RtApiWasapi::wasapiThread: Unable to create capture event.";
4905 hr = captureAudioClient->SetEventHandle( captureEvent );
4906 if ( FAILED( hr ) ) {
4907 errorText_ = "RtApiWasapi::wasapiThread: Unable to set capture event handle.";
4911 ( ( WasapiHandle* ) stream_.apiHandle )->captureClient = captureClient;
4912 ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent = captureEvent;
4915 unsigned int inBufferSize = 0;
4916 hr = captureAudioClient->GetBufferSize( &inBufferSize );
4917 if ( FAILED( hr ) ) {
4918 errorText_ = "RtApiWasapi::wasapiThread: Unable to get capture buffer size.";
4922 // scale outBufferSize according to stream->user sample rate ratio
4923 unsigned int outBufferSize = ( unsigned int ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT];
4924 inBufferSize *= stream_.nDeviceChannels[INPUT];
4926 // set captureBuffer size
4927 captureBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[INPUT] ) );
4929 // reset the capture stream
4930 hr = captureAudioClient->Reset();
4931 if ( FAILED( hr ) ) {
4932 errorText_ = "RtApiWasapi::wasapiThread: Unable to reset capture stream.";
4936 // start the capture stream
4937 hr = captureAudioClient->Start();
4938 if ( FAILED( hr ) ) {
4939 errorText_ = "RtApiWasapi::wasapiThread: Unable to start capture stream.";
4944 // start render stream if applicable
4945 if ( renderAudioClient ) {
4946 hr = renderAudioClient->GetMixFormat( &renderFormat );
4947 if ( FAILED( hr ) ) {
4948 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
4952 renderSrRatio = ( ( float ) renderFormat->nSamplesPerSec / stream_.sampleRate );
4954 // initialize render stream according to desire buffer size
4955 float desiredBufferSize = stream_.bufferSize * renderSrRatio;
4956 REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / renderFormat->nSamplesPerSec );
4958 if ( !renderClient ) {
4959 hr = renderAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
4960 AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
4961 desiredBufferPeriod,
4962 desiredBufferPeriod,
4965 if ( FAILED( hr ) ) {
4966 errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize render audio client.";
4970 hr = renderAudioClient->GetService( __uuidof( IAudioRenderClient ),
4971 ( void** ) &renderClient );
4972 if ( FAILED( hr ) ) {
4973 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render client handle.";
4977 // configure renderEvent to trigger on every available render buffer
4978 renderEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
4979 if ( !renderEvent ) {
4980 errorType = RtAudioError::SYSTEM_ERROR;
4981 errorText_ = "RtApiWasapi::wasapiThread: Unable to create render event.";
4985 hr = renderAudioClient->SetEventHandle( renderEvent );
4986 if ( FAILED( hr ) ) {
4987 errorText_ = "RtApiWasapi::wasapiThread: Unable to set render event handle.";
4991 ( ( WasapiHandle* ) stream_.apiHandle )->renderClient = renderClient;
4992 ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent = renderEvent;
4995 unsigned int outBufferSize = 0;
4996 hr = renderAudioClient->GetBufferSize( &outBufferSize );
4997 if ( FAILED( hr ) ) {
4998 errorText_ = "RtApiWasapi::wasapiThread: Unable to get render buffer size.";
5002 // scale inBufferSize according to user->stream sample rate ratio
5003 unsigned int inBufferSize = ( unsigned int ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT];
5004 outBufferSize *= stream_.nDeviceChannels[OUTPUT];
5006 // set renderBuffer size
5007 renderBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[OUTPUT] ) );
5009 // reset the render stream
5010 hr = renderAudioClient->Reset();
5011 if ( FAILED( hr ) ) {
5012 errorText_ = "RtApiWasapi::wasapiThread: Unable to reset render stream.";
5016 // start the render stream
5017 hr = renderAudioClient->Start();
5018 if ( FAILED( hr ) ) {
5019 errorText_ = "RtApiWasapi::wasapiThread: Unable to start render stream.";
5024 if ( stream_.mode == INPUT ) {
5025 using namespace std; // for roundf
5026 convBuffSize = ( size_t ) roundf( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
5027 deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
5029 else if ( stream_.mode == OUTPUT ) {
5030 convBuffSize = ( size_t ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
5031 deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
5033 else if ( stream_.mode == DUPLEX ) {
5034 convBuffSize = std::max( ( size_t ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
5035 ( size_t ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
5036 deviceBuffSize = std::max( stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
5037 stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
5040 convBuffer = ( char* ) malloc( convBuffSize );
5041 stream_.deviceBuffer = ( char* ) malloc( deviceBuffSize );
5042 if ( !convBuffer || !stream_.deviceBuffer ) {
5043 errorType = RtAudioError::MEMORY_ERROR;
5044 errorText_ = "RtApiWasapi::wasapiThread: Error allocating device buffer memory.";
5048 // stream process loop
5049 while ( stream_.state != STREAM_STOPPING ) {
5050 if ( !callbackPulled ) {
5053 // 1. Pull callback buffer from inputBuffer
5054 // 2. If 1. was successful: Convert callback buffer to user sample rate and channel count
5055 // Convert callback buffer to user format
5057 if ( captureAudioClient ) {
5058 // Pull callback buffer from inputBuffer
5059 callbackPulled = captureBuffer.pullBuffer( convBuffer,
5060 ( unsigned int ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT],
5061 stream_.deviceFormat[INPUT] );
5063 if ( callbackPulled ) {
5064 // Convert callback buffer to user sample rate
5065 convertBufferWasapi( stream_.deviceBuffer,
5067 stream_.nDeviceChannels[INPUT],
5068 captureFormat->nSamplesPerSec,
5070 ( unsigned int ) ( stream_.bufferSize * captureSrRatio ),
5072 stream_.deviceFormat[INPUT] );
5074 if ( stream_.doConvertBuffer[INPUT] ) {
5075 // Convert callback buffer to user format
5076 convertBuffer( stream_.userBuffer[INPUT],
5077 stream_.deviceBuffer,
5078 stream_.convertInfo[INPUT] );
5081 // no further conversion, simple copy deviceBuffer to userBuffer
5082 memcpy( stream_.userBuffer[INPUT],
5083 stream_.deviceBuffer,
5084 stream_.bufferSize * stream_.nUserChannels[INPUT] * formatBytes( stream_.userFormat ) );
5089 // if there is no capture stream, set callbackPulled flag
5090 callbackPulled = true;
5095 // 1. Execute user callback method
5096 // 2. Handle return value from callback
5098 // if callback has not requested the stream to stop
5099 if ( callbackPulled && !callbackStopped ) {
5100 // Execute user callback method
5101 callbackResult = callback( stream_.userBuffer[OUTPUT],
5102 stream_.userBuffer[INPUT],
5105 captureFlags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY ? RTAUDIO_INPUT_OVERFLOW : 0,
5106 stream_.callbackInfo.userData );
5108 // Handle return value from callback
5109 if ( callbackResult == 1 ) {
5110 // instantiate a thread to stop this thread
5111 HANDLE threadHandle = CreateThread( NULL, 0, stopWasapiThread, this, 0, NULL );
5112 if ( !threadHandle ) {
5113 errorType = RtAudioError::THREAD_ERROR;
5114 errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream stop thread.";
5117 else if ( !CloseHandle( threadHandle ) ) {
5118 errorType = RtAudioError::THREAD_ERROR;
5119 errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream stop thread handle.";
5123 callbackStopped = true;
5125 else if ( callbackResult == 2 ) {
5126 // instantiate a thread to stop this thread
5127 HANDLE threadHandle = CreateThread( NULL, 0, abortWasapiThread, this, 0, NULL );
5128 if ( !threadHandle ) {
5129 errorType = RtAudioError::THREAD_ERROR;
5130 errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream abort thread.";
5133 else if ( !CloseHandle( threadHandle ) ) {
5134 errorType = RtAudioError::THREAD_ERROR;
5135 errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream abort thread handle.";
5139 callbackStopped = true;
5146 // 1. Convert callback buffer to stream format
5147 // 2. Convert callback buffer to stream sample rate and channel count
5148 // 3. Push callback buffer into outputBuffer
5150 if ( renderAudioClient && callbackPulled ) {
5151 if ( stream_.doConvertBuffer[OUTPUT] ) {
5152 // Convert callback buffer to stream format
5153 convertBuffer( stream_.deviceBuffer,
5154 stream_.userBuffer[OUTPUT],
5155 stream_.convertInfo[OUTPUT] );
5159 // Convert callback buffer to stream sample rate
5160 convertBufferWasapi( convBuffer,
5161 stream_.deviceBuffer,
5162 stream_.nDeviceChannels[OUTPUT],
5164 renderFormat->nSamplesPerSec,
5167 stream_.deviceFormat[OUTPUT] );
5169 // Push callback buffer into outputBuffer
5170 callbackPushed = renderBuffer.pushBuffer( convBuffer,
5171 convBufferSize * stream_.nDeviceChannels[OUTPUT],
5172 stream_.deviceFormat[OUTPUT] );
5175 // if there is no render stream, set callbackPushed flag
5176 callbackPushed = true;
5181 // 1. Get capture buffer from stream
5182 // 2. Push capture buffer into inputBuffer
5183 // 3. If 2. was successful: Release capture buffer
5185 if ( captureAudioClient ) {
5186 // if the callback input buffer was not pulled from captureBuffer, wait for next capture event
5187 if ( !callbackPulled ) {
5188 WaitForSingleObject( captureEvent, INFINITE );
5191 // Get capture buffer from stream
5192 hr = captureClient->GetBuffer( &streamBuffer,
5194 &captureFlags, NULL, NULL );
5195 if ( FAILED( hr ) ) {
5196 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture buffer.";
5200 if ( bufferFrameCount != 0 ) {
5201 // Push capture buffer into inputBuffer
5202 if ( captureBuffer.pushBuffer( ( char* ) streamBuffer,
5203 bufferFrameCount * stream_.nDeviceChannels[INPUT],
5204 stream_.deviceFormat[INPUT] ) )
5206 // Release capture buffer
5207 hr = captureClient->ReleaseBuffer( bufferFrameCount );
5208 if ( FAILED( hr ) ) {
5209 errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
5215 // Inform WASAPI that capture was unsuccessful
5216 hr = captureClient->ReleaseBuffer( 0 );
5217 if ( FAILED( hr ) ) {
5218 errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
5225 // Inform WASAPI that capture was unsuccessful
5226 hr = captureClient->ReleaseBuffer( 0 );
5227 if ( FAILED( hr ) ) {
5228 errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
5236 // 1. Get render buffer from stream
5237 // 2. Pull next buffer from outputBuffer
5238 // 3. If 2. was successful: Fill render buffer with next buffer
5239 // Release render buffer
5241 if ( renderAudioClient ) {
5242 // if the callback output buffer was not pushed to renderBuffer, wait for next render event
5243 if ( callbackPulled && !callbackPushed ) {
5244 WaitForSingleObject( renderEvent, INFINITE );
5247 // Get render buffer from stream
5248 hr = renderAudioClient->GetBufferSize( &bufferFrameCount );
5249 if ( FAILED( hr ) ) {
5250 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer size.";
5254 hr = renderAudioClient->GetCurrentPadding( &numFramesPadding );
5255 if ( FAILED( hr ) ) {
5256 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer padding.";
5260 bufferFrameCount -= numFramesPadding;
5262 if ( bufferFrameCount != 0 ) {
5263 hr = renderClient->GetBuffer( bufferFrameCount, &streamBuffer );
5264 if ( FAILED( hr ) ) {
5265 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer.";
5269 // Pull next buffer from outputBuffer
5270 // Fill render buffer with next buffer
5271 if ( renderBuffer.pullBuffer( ( char* ) streamBuffer,
5272 bufferFrameCount * stream_.nDeviceChannels[OUTPUT],
5273 stream_.deviceFormat[OUTPUT] ) )
5275 // Release render buffer
5276 hr = renderClient->ReleaseBuffer( bufferFrameCount, 0 );
5277 if ( FAILED( hr ) ) {
5278 errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
5284 // Inform WASAPI that render was unsuccessful
5285 hr = renderClient->ReleaseBuffer( 0, 0 );
5286 if ( FAILED( hr ) ) {
5287 errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
5294 // Inform WASAPI that render was unsuccessful
5295 hr = renderClient->ReleaseBuffer( 0, 0 );
5296 if ( FAILED( hr ) ) {
5297 errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
5303 // if the callback buffer was pushed renderBuffer reset callbackPulled flag
5304 if ( callbackPushed ) {
5305 callbackPulled = false;
5307 RtApi::tickStreamTime();
5314 CoTaskMemFree( captureFormat );
5315 CoTaskMemFree( renderFormat );
5317 free ( convBuffer );
5321 // update stream state
5322 stream_.state = STREAM_STOPPED;
5324 if ( errorText_.empty() )
5330 //******************** End of __WINDOWS_WASAPI__ *********************//
5334 #if defined(__WINDOWS_DS__) // Windows DirectSound API
5336 // Modified by Robin Davies, October 2005
5337 // - Improvements to DirectX pointer chasing.
5338 // - Bug fix for non-power-of-two Asio granularity used by Edirol PCR-A30.
5339 // - Auto-call CoInitialize for DSOUND and ASIO platforms.
5340 // Various revisions for RtAudio 4.0 by Gary Scavone, April 2007
5341 // Changed device query structure for RtAudio 4.0.7, January 2010
5343 #include <windows.h>
5344 #include <process.h>
5345 #include <mmsystem.h>
5349 #include <algorithm>
5351 #if defined(__MINGW32__)
5352 // missing from latest mingw winapi
5353 #define WAVE_FORMAT_96M08 0x00010000 /* 96 kHz, Mono, 8-bit */
5354 #define WAVE_FORMAT_96S08 0x00020000 /* 96 kHz, Stereo, 8-bit */
5355 #define WAVE_FORMAT_96M16 0x00040000 /* 96 kHz, Mono, 16-bit */
5356 #define WAVE_FORMAT_96S16 0x00080000 /* 96 kHz, Stereo, 16-bit */
5359 #define MINIMUM_DEVICE_BUFFER_SIZE 32768
5361 #ifdef _MSC_VER // if Microsoft Visual C++
5362 #pragma comment( lib, "winmm.lib" ) // then, auto-link winmm.lib. Otherwise, it has to be added manually.
5365 static inline DWORD dsPointerBetween( DWORD pointer, DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize )
5367 if ( pointer > bufferSize ) pointer -= bufferSize;
5368 if ( laterPointer < earlierPointer ) laterPointer += bufferSize;
5369 if ( pointer < earlierPointer ) pointer += bufferSize;
5370 return pointer >= earlierPointer && pointer < laterPointer;
5373 // A structure to hold various information related to the DirectSound
5374 // API implementation.
5376 unsigned int drainCounter; // Tracks callback counts when draining
5377 bool internalDrain; // Indicates if stop is initiated from callback or not.
5381 UINT bufferPointer[2];
5382 DWORD dsBufferSize[2];
5383 DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by.
5387 :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; }
5390 // Declarations for utility functions, callbacks, and structures
5391 // specific to the DirectSound implementation.
5392 static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
5393 LPCTSTR description,
5397 static const char* getErrorString( int code );
5399 static unsigned __stdcall callbackHandler( void *ptr );
5408 : found(false) { validId[0] = false; validId[1] = false; }
5411 struct DsProbeData {
5413 std::vector<struct DsDevice>* dsDevices;
5416 RtApiDs :: RtApiDs()
5418 // Dsound will run both-threaded. If CoInitialize fails, then just
5419 // accept whatever the mainline chose for a threading model.
5420 coInitialized_ = false;
5421 HRESULT hr = CoInitialize( NULL );
5422 if ( !FAILED( hr ) ) coInitialized_ = true;
5425 RtApiDs :: ~RtApiDs()
5427 if ( stream_.state != STREAM_CLOSED ) closeStream();
5428 if ( coInitialized_ ) CoUninitialize(); // balanced call.
5431 // The DirectSound default output is always the first device.
5432 unsigned int RtApiDs :: getDefaultOutputDevice( void )
5437 // The DirectSound default input is always the first input device,
5438 // which is the first capture device enumerated.
5439 unsigned int RtApiDs :: getDefaultInputDevice( void )
5444 unsigned int RtApiDs :: getDeviceCount( void )
5446 // Set query flag for previously found devices to false, so that we
5447 // can check for any devices that have disappeared.
5448 for ( unsigned int i=0; i<dsDevices.size(); i++ )
5449 dsDevices[i].found = false;
5451 // Query DirectSound devices.
5452 struct DsProbeData probeInfo;
5453 probeInfo.isInput = false;
5454 probeInfo.dsDevices = &dsDevices;
5455 HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );
5456 if ( FAILED( result ) ) {
5457 errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!";
5458 errorText_ = errorStream_.str();
5459 error( RtAudioError::WARNING );
5462 // Query DirectSoundCapture devices.
5463 probeInfo.isInput = true;
5464 result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );
5465 if ( FAILED( result ) ) {
5466 errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!";
5467 errorText_ = errorStream_.str();
5468 error( RtAudioError::WARNING );
5471 // Clean out any devices that may have disappeared (code update submitted by Eli Zehngut).
5472 for ( unsigned int i=0; i<dsDevices.size(); ) {
5473 if ( dsDevices[i].found == false ) dsDevices.erase( dsDevices.begin() + i );
5477 return static_cast<unsigned int>(dsDevices.size());
5480 RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device )
5482 RtAudio::DeviceInfo info;
5483 info.probed = false;
5485 if ( dsDevices.size() == 0 ) {
5486 // Force a query of all devices
5488 if ( dsDevices.size() == 0 ) {
5489 errorText_ = "RtApiDs::getDeviceInfo: no devices found!";
5490 error( RtAudioError::INVALID_USE );
5495 if ( device >= dsDevices.size() ) {
5496 errorText_ = "RtApiDs::getDeviceInfo: device ID is invalid!";
5497 error( RtAudioError::INVALID_USE );
5502 if ( dsDevices[ device ].validId[0] == false ) goto probeInput;
5504 LPDIRECTSOUND output;
5506 result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );
5507 if ( FAILED( result ) ) {
5508 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!";
5509 errorText_ = errorStream_.str();
5510 error( RtAudioError::WARNING );
5514 outCaps.dwSize = sizeof( outCaps );
5515 result = output->GetCaps( &outCaps );
5516 if ( FAILED( result ) ) {
5518 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting capabilities!";
5519 errorText_ = errorStream_.str();
5520 error( RtAudioError::WARNING );
5524 // Get output channel information.
5525 info.outputChannels = ( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1;
5527 // Get sample rate information.
5528 info.sampleRates.clear();
5529 for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
5530 if ( SAMPLE_RATES[k] >= (unsigned int) outCaps.dwMinSecondarySampleRate &&
5531 SAMPLE_RATES[k] <= (unsigned int) outCaps.dwMaxSecondarySampleRate ) {
5532 info.sampleRates.push_back( SAMPLE_RATES[k] );
5534 if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
5535 info.preferredSampleRate = SAMPLE_RATES[k];
5539 // Get format information.
5540 if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT ) info.nativeFormats |= RTAUDIO_SINT16;
5541 if ( outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) info.nativeFormats |= RTAUDIO_SINT8;
5545 if ( getDefaultOutputDevice() == device )
5546 info.isDefaultOutput = true;
5548 if ( dsDevices[ device ].validId[1] == false ) {
5549 info.name = dsDevices[ device ].name;
5556 LPDIRECTSOUNDCAPTURE input;
5557 result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );
5558 if ( FAILED( result ) ) {
5559 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";
5560 errorText_ = errorStream_.str();
5561 error( RtAudioError::WARNING );
5566 inCaps.dwSize = sizeof( inCaps );
5567 result = input->GetCaps( &inCaps );
5568 if ( FAILED( result ) ) {
5570 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsDevices[ device ].name << ")!";
5571 errorText_ = errorStream_.str();
5572 error( RtAudioError::WARNING );
5576 // Get input channel information.
5577 info.inputChannels = inCaps.dwChannels;
5579 // Get sample rate and format information.
5580 std::vector<unsigned int> rates;
5581 if ( inCaps.dwChannels >= 2 ) {
5582 if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.nativeFormats |= RTAUDIO_SINT16;
5583 if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.nativeFormats |= RTAUDIO_SINT16;
5584 if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.nativeFormats |= RTAUDIO_SINT16;
5585 if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.nativeFormats |= RTAUDIO_SINT16;
5586 if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.nativeFormats |= RTAUDIO_SINT8;
5587 if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.nativeFormats |= RTAUDIO_SINT8;
5588 if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.nativeFormats |= RTAUDIO_SINT8;
5589 if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.nativeFormats |= RTAUDIO_SINT8;
5591 if ( info.nativeFormats & RTAUDIO_SINT16 ) {
5592 if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) rates.push_back( 11025 );
5593 if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) rates.push_back( 22050 );
5594 if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) rates.push_back( 44100 );
5595 if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) rates.push_back( 96000 );
5597 else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
5598 if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) rates.push_back( 11025 );
5599 if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) rates.push_back( 22050 );
5600 if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) rates.push_back( 44100 );
5601 if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) rates.push_back( 96000 );
5604 else if ( inCaps.dwChannels == 1 ) {
5605 if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.nativeFormats |= RTAUDIO_SINT16;
5606 if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.nativeFormats |= RTAUDIO_SINT16;
5607 if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.nativeFormats |= RTAUDIO_SINT16;
5608 if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.nativeFormats |= RTAUDIO_SINT16;
5609 if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.nativeFormats |= RTAUDIO_SINT8;
5610 if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.nativeFormats |= RTAUDIO_SINT8;
5611 if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.nativeFormats |= RTAUDIO_SINT8;
5612 if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.nativeFormats |= RTAUDIO_SINT8;
5614 if ( info.nativeFormats & RTAUDIO_SINT16 ) {
5615 if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) rates.push_back( 11025 );
5616 if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) rates.push_back( 22050 );
5617 if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) rates.push_back( 44100 );
5618 if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) rates.push_back( 96000 );
5620 else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
5621 if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) rates.push_back( 11025 );
5622 if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) rates.push_back( 22050 );
5623 if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) rates.push_back( 44100 );
5624 if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) rates.push_back( 96000 );
5627 else info.inputChannels = 0; // technically, this would be an error
5631 if ( info.inputChannels == 0 ) return info;
5633 // Copy the supported rates to the info structure but avoid duplication.
5635 for ( unsigned int i=0; i<rates.size(); i++ ) {
5637 for ( unsigned int j=0; j<info.sampleRates.size(); j++ ) {
5638 if ( rates[i] == info.sampleRates[j] ) {
5643 if ( found == false ) info.sampleRates.push_back( rates[i] );
5645 std::sort( info.sampleRates.begin(), info.sampleRates.end() );
5647 // If device opens for both playback and capture, we determine the channels.
5648 if ( info.outputChannels > 0 && info.inputChannels > 0 )
5649 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
5651 if ( device == 0 ) info.isDefaultInput = true;
5653 // Copy name and return.
5654 info.name = dsDevices[ device ].name;
5659 bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
5660 unsigned int firstChannel, unsigned int sampleRate,
5661 RtAudioFormat format, unsigned int *bufferSize,
5662 RtAudio::StreamOptions *options )
5664 if ( channels + firstChannel > 2 ) {
5665 errorText_ = "RtApiDs::probeDeviceOpen: DirectSound does not support more than 2 channels per device.";
5669 size_t nDevices = dsDevices.size();
5670 if ( nDevices == 0 ) {
5671 // This should not happen because a check is made before this function is called.
5672 errorText_ = "RtApiDs::probeDeviceOpen: no devices found!";
5676 if ( device >= nDevices ) {
5677 // This should not happen because a check is made before this function is called.
5678 errorText_ = "RtApiDs::probeDeviceOpen: device ID is invalid!";
5682 if ( mode == OUTPUT ) {
5683 if ( dsDevices[ device ].validId[0] == false ) {
5684 errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support output!";
5685 errorText_ = errorStream_.str();
5689 else { // mode == INPUT
5690 if ( dsDevices[ device ].validId[1] == false ) {
5691 errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support input!";
5692 errorText_ = errorStream_.str();
5697 // According to a note in PortAudio, using GetDesktopWindow()
5698 // instead of GetForegroundWindow() is supposed to avoid problems
5699 // that occur when the application's window is not the foreground
5700 // window. Also, if the application window closes before the
5701 // DirectSound buffer, DirectSound can crash. In the past, I had
5702 // problems when using GetDesktopWindow() but it seems fine now
5703 // (January 2010). I'll leave it commented here.
5704 // HWND hWnd = GetForegroundWindow();
5705 HWND hWnd = GetDesktopWindow();
5707 // Check the numberOfBuffers parameter and limit the lowest value to
5708 // two. This is a judgement call and a value of two is probably too
5709 // low for capture, but it should work for playback.
5711 if ( options ) nBuffers = options->numberOfBuffers;
5712 if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) nBuffers = 2;
5713 if ( nBuffers < 2 ) nBuffers = 3;
5715 // Check the lower range of the user-specified buffer size and set
5716 // (arbitrarily) to a lower bound of 32.
5717 if ( *bufferSize < 32 ) *bufferSize = 32;
5719 // Create the wave format structure. The data format setting will
5720 // be determined later.
5721 WAVEFORMATEX waveFormat;
5722 ZeroMemory( &waveFormat, sizeof(WAVEFORMATEX) );
5723 waveFormat.wFormatTag = WAVE_FORMAT_PCM;
5724 waveFormat.nChannels = channels + firstChannel;
5725 waveFormat.nSamplesPerSec = (unsigned long) sampleRate;
5727 // Determine the device buffer size. By default, we'll use the value
5728 // defined above (32K), but we will grow it to make allowances for
5729 // very large software buffer sizes.
5730 DWORD dsBufferSize = MINIMUM_DEVICE_BUFFER_SIZE;
5731 DWORD dsPointerLeadTime = 0;
5733 void *ohandle = 0, *bhandle = 0;
5735 if ( mode == OUTPUT ) {
5737 LPDIRECTSOUND output;
5738 result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );
5739 if ( FAILED( result ) ) {
5740 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!";
5741 errorText_ = errorStream_.str();
5746 outCaps.dwSize = sizeof( outCaps );
5747 result = output->GetCaps( &outCaps );
5748 if ( FAILED( result ) ) {
5750 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting capabilities (" << dsDevices[ device ].name << ")!";
5751 errorText_ = errorStream_.str();
5755 // Check channel information.
5756 if ( channels + firstChannel == 2 && !( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ) {
5757 errorStream_ << "RtApiDs::getDeviceInfo: the output device (" << dsDevices[ device ].name << ") does not support stereo playback.";
5758 errorText_ = errorStream_.str();
5762 // Check format information. Use 16-bit format unless not
5763 // supported or user requests 8-bit.
5764 if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT &&
5765 !( format == RTAUDIO_SINT8 && outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) ) {
5766 waveFormat.wBitsPerSample = 16;
5767 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
5770 waveFormat.wBitsPerSample = 8;
5771 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
5773 stream_.userFormat = format;
5775 // Update wave format structure and buffer information.
5776 waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
5777 waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
5778 dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
5780 // If the user wants an even bigger buffer, increase the device buffer size accordingly.
5781 while ( dsPointerLeadTime * 2U > dsBufferSize )
5784 // Set cooperative level to DSSCL_EXCLUSIVE ... sound stops when window focus changes.
5785 // result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE );
5786 // Set cooperative level to DSSCL_PRIORITY ... sound remains when window focus changes.
5787 result = output->SetCooperativeLevel( hWnd, DSSCL_PRIORITY );
5788 if ( FAILED( result ) ) {
5790 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting cooperative level (" << dsDevices[ device ].name << ")!";
5791 errorText_ = errorStream_.str();
5795 // Even though we will write to the secondary buffer, we need to
5796 // access the primary buffer to set the correct output format
5797 // (since the default is 8-bit, 22 kHz!). Setup the DS primary
5798 // buffer description.
5799 DSBUFFERDESC bufferDescription;
5800 ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
5801 bufferDescription.dwSize = sizeof( DSBUFFERDESC );
5802 bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER;
5804 // Obtain the primary buffer
5805 LPDIRECTSOUNDBUFFER buffer;
5806 result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
5807 if ( FAILED( result ) ) {
5809 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") accessing primary buffer (" << dsDevices[ device ].name << ")!";
5810 errorText_ = errorStream_.str();
5814 // Set the primary DS buffer sound format.
5815 result = buffer->SetFormat( &waveFormat );
5816 if ( FAILED( result ) ) {
5818 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting primary buffer format (" << dsDevices[ device ].name << ")!";
5819 errorText_ = errorStream_.str();
5823 // Setup the secondary DS buffer description.
5824 ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
5825 bufferDescription.dwSize = sizeof( DSBUFFERDESC );
5826 bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
5827 DSBCAPS_GLOBALFOCUS |
5828 DSBCAPS_GETCURRENTPOSITION2 |
5829 DSBCAPS_LOCHARDWARE ); // Force hardware mixing
5830 bufferDescription.dwBufferBytes = dsBufferSize;
5831 bufferDescription.lpwfxFormat = &waveFormat;
5833 // Try to create the secondary DS buffer. If that doesn't work,
5834 // try to use software mixing. Otherwise, there's a problem.
5835 result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
5836 if ( FAILED( result ) ) {
5837 bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
5838 DSBCAPS_GLOBALFOCUS |
5839 DSBCAPS_GETCURRENTPOSITION2 |
5840 DSBCAPS_LOCSOFTWARE ); // Force software mixing
5841 result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
5842 if ( FAILED( result ) ) {
5844 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating secondary buffer (" << dsDevices[ device ].name << ")!";
5845 errorText_ = errorStream_.str();
5850 // Get the buffer size ... might be different from what we specified.
5852 dsbcaps.dwSize = sizeof( DSBCAPS );
5853 result = buffer->GetCaps( &dsbcaps );
5854 if ( FAILED( result ) ) {
5857 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";
5858 errorText_ = errorStream_.str();
5862 dsBufferSize = dsbcaps.dwBufferBytes;
5864 // Lock the DS buffer
5867 result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );
5868 if ( FAILED( result ) ) {
5871 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking buffer (" << dsDevices[ device ].name << ")!";
5872 errorText_ = errorStream_.str();
5876 // Zero the DS buffer
5877 ZeroMemory( audioPtr, dataLen );
5879 // Unlock the DS buffer
5880 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
5881 if ( FAILED( result ) ) {
5884 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking buffer (" << dsDevices[ device ].name << ")!";
5885 errorText_ = errorStream_.str();
5889 ohandle = (void *) output;
5890 bhandle = (void *) buffer;
5893 if ( mode == INPUT ) {
5895 LPDIRECTSOUNDCAPTURE input;
5896 result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );
5897 if ( FAILED( result ) ) {
5898 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";
5899 errorText_ = errorStream_.str();
5904 inCaps.dwSize = sizeof( inCaps );
5905 result = input->GetCaps( &inCaps );
5906 if ( FAILED( result ) ) {
5908 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsDevices[ device ].name << ")!";
5909 errorText_ = errorStream_.str();
5913 // Check channel information.
5914 if ( inCaps.dwChannels < channels + firstChannel ) {
5915 errorText_ = "RtApiDs::getDeviceInfo: the input device does not support requested input channels.";
5919 // Check format information. Use 16-bit format unless user
5921 DWORD deviceFormats;
5922 if ( channels + firstChannel == 2 ) {
5923 deviceFormats = WAVE_FORMAT_1S08 | WAVE_FORMAT_2S08 | WAVE_FORMAT_4S08 | WAVE_FORMAT_96S08;
5924 if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
5925 waveFormat.wBitsPerSample = 8;
5926 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
5928 else { // assume 16-bit is supported
5929 waveFormat.wBitsPerSample = 16;
5930 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
5933 else { // channel == 1
5934 deviceFormats = WAVE_FORMAT_1M08 | WAVE_FORMAT_2M08 | WAVE_FORMAT_4M08 | WAVE_FORMAT_96M08;
5935 if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
5936 waveFormat.wBitsPerSample = 8;
5937 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
5939 else { // assume 16-bit is supported
5940 waveFormat.wBitsPerSample = 16;
5941 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
5944 stream_.userFormat = format;
5946 // Update wave format structure and buffer information.
5947 waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
5948 waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
5949 dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
5951 // If the user wants an even bigger buffer, increase the device buffer size accordingly.
5952 while ( dsPointerLeadTime * 2U > dsBufferSize )
5955 // Setup the secondary DS buffer description.
5956 DSCBUFFERDESC bufferDescription;
5957 ZeroMemory( &bufferDescription, sizeof( DSCBUFFERDESC ) );
5958 bufferDescription.dwSize = sizeof( DSCBUFFERDESC );
5959 bufferDescription.dwFlags = 0;
5960 bufferDescription.dwReserved = 0;
5961 bufferDescription.dwBufferBytes = dsBufferSize;
5962 bufferDescription.lpwfxFormat = &waveFormat;
5964 // Create the capture buffer.
5965 LPDIRECTSOUNDCAPTUREBUFFER buffer;
5966 result = input->CreateCaptureBuffer( &bufferDescription, &buffer, NULL );
5967 if ( FAILED( result ) ) {
5969 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating input buffer (" << dsDevices[ device ].name << ")!";
5970 errorText_ = errorStream_.str();
5974 // Get the buffer size ... might be different from what we specified.
5976 dscbcaps.dwSize = sizeof( DSCBCAPS );
5977 result = buffer->GetCaps( &dscbcaps );
5978 if ( FAILED( result ) ) {
5981 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";
5982 errorText_ = errorStream_.str();
5986 dsBufferSize = dscbcaps.dwBufferBytes;
5988 // NOTE: We could have a problem here if this is a duplex stream
5989 // and the play and capture hardware buffer sizes are different
5990 // (I'm actually not sure if that is a problem or not).
5991 // Currently, we are not verifying that.
5993 // Lock the capture buffer
5996 result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );
5997 if ( FAILED( result ) ) {
6000 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking input buffer (" << dsDevices[ device ].name << ")!";
6001 errorText_ = errorStream_.str();
6006 ZeroMemory( audioPtr, dataLen );
6008 // Unlock the buffer
6009 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
6010 if ( FAILED( result ) ) {
6013 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking input buffer (" << dsDevices[ device ].name << ")!";
6014 errorText_ = errorStream_.str();
6018 ohandle = (void *) input;
6019 bhandle = (void *) buffer;
6022 // Set various stream parameters
6023 DsHandle *handle = 0;
6024 stream_.nDeviceChannels[mode] = channels + firstChannel;
6025 stream_.nUserChannels[mode] = channels;
6026 stream_.bufferSize = *bufferSize;
6027 stream_.channelOffset[mode] = firstChannel;
6028 stream_.deviceInterleaved[mode] = true;
6029 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
6030 else stream_.userInterleaved = true;
6032 // Set flag for buffer conversion
6033 stream_.doConvertBuffer[mode] = false;
6034 if (stream_.nUserChannels[mode] != stream_.nDeviceChannels[mode])
6035 stream_.doConvertBuffer[mode] = true;
6036 if (stream_.userFormat != stream_.deviceFormat[mode])
6037 stream_.doConvertBuffer[mode] = true;
6038 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
6039 stream_.nUserChannels[mode] > 1 )
6040 stream_.doConvertBuffer[mode] = true;
6042 // Allocate necessary internal buffers
6043 long bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
6044 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
6045 if ( stream_.userBuffer[mode] == NULL ) {
6046 errorText_ = "RtApiDs::probeDeviceOpen: error allocating user buffer memory.";
6050 if ( stream_.doConvertBuffer[mode] ) {
6052 bool makeBuffer = true;
6053 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
6054 if ( mode == INPUT ) {
6055 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
6056 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
6057 if ( bufferBytes <= (long) bytesOut ) makeBuffer = false;
6062 bufferBytes *= *bufferSize;
6063 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
6064 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
6065 if ( stream_.deviceBuffer == NULL ) {
6066 errorText_ = "RtApiDs::probeDeviceOpen: error allocating device buffer memory.";
6072 // Allocate our DsHandle structures for the stream.
6073 if ( stream_.apiHandle == 0 ) {
6075 handle = new DsHandle;
6077 catch ( std::bad_alloc& ) {
6078 errorText_ = "RtApiDs::probeDeviceOpen: error allocating AsioHandle memory.";
6082 // Create a manual-reset event.
6083 handle->condition = CreateEvent( NULL, // no security
6084 TRUE, // manual-reset
6085 FALSE, // non-signaled initially
6087 stream_.apiHandle = (void *) handle;
6090 handle = (DsHandle *) stream_.apiHandle;
6091 handle->id[mode] = ohandle;
6092 handle->buffer[mode] = bhandle;
6093 handle->dsBufferSize[mode] = dsBufferSize;
6094 handle->dsPointerLeadTime[mode] = dsPointerLeadTime;
6096 stream_.device[mode] = device;
6097 stream_.state = STREAM_STOPPED;
6098 if ( stream_.mode == OUTPUT && mode == INPUT )
6099 // We had already set up an output stream.
6100 stream_.mode = DUPLEX;
6102 stream_.mode = mode;
6103 stream_.nBuffers = nBuffers;
6104 stream_.sampleRate = sampleRate;
6106 // Setup the buffer conversion information structure.
6107 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
6109 // Setup the callback thread.
6110 if ( stream_.callbackInfo.isRunning == false ) {
6112 stream_.callbackInfo.isRunning = true;
6113 stream_.callbackInfo.object = (void *) this;
6114 stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &callbackHandler,
6115 &stream_.callbackInfo, 0, &threadId );
6116 if ( stream_.callbackInfo.thread == 0 ) {
6117 errorText_ = "RtApiDs::probeDeviceOpen: error creating callback thread!";
6121 // Boost DS thread priority
6122 SetThreadPriority( (HANDLE) stream_.callbackInfo.thread, THREAD_PRIORITY_HIGHEST );
6128 if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
6129 LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
6130 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6131 if ( buffer ) buffer->Release();
6134 if ( handle->buffer[1] ) {
6135 LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
6136 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6137 if ( buffer ) buffer->Release();
6140 CloseHandle( handle->condition );
6142 stream_.apiHandle = 0;
6145 for ( int i=0; i<2; i++ ) {
6146 if ( stream_.userBuffer[i] ) {
6147 free( stream_.userBuffer[i] );
6148 stream_.userBuffer[i] = 0;
6152 if ( stream_.deviceBuffer ) {
6153 free( stream_.deviceBuffer );
6154 stream_.deviceBuffer = 0;
6157 stream_.state = STREAM_CLOSED;
6161 void RtApiDs :: closeStream()
6163 if ( stream_.state == STREAM_CLOSED ) {
6164 errorText_ = "RtApiDs::closeStream(): no open stream to close!";
6165 error( RtAudioError::WARNING );
6169 // Stop the callback thread.
6170 stream_.callbackInfo.isRunning = false;
6171 WaitForSingleObject( (HANDLE) stream_.callbackInfo.thread, INFINITE );
6172 CloseHandle( (HANDLE) stream_.callbackInfo.thread );
6174 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6176 if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
6177 LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
6178 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6185 if ( handle->buffer[1] ) {
6186 LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
6187 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6194 CloseHandle( handle->condition );
6196 stream_.apiHandle = 0;
6199 for ( int i=0; i<2; i++ ) {
6200 if ( stream_.userBuffer[i] ) {
6201 free( stream_.userBuffer[i] );
6202 stream_.userBuffer[i] = 0;
6206 if ( stream_.deviceBuffer ) {
6207 free( stream_.deviceBuffer );
6208 stream_.deviceBuffer = 0;
6211 stream_.mode = UNINITIALIZED;
6212 stream_.state = STREAM_CLOSED;
6215 void RtApiDs :: startStream()
6218 if ( stream_.state == STREAM_RUNNING ) {
6219 errorText_ = "RtApiDs::startStream(): the stream is already running!";
6220 error( RtAudioError::WARNING );
6224 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6226 // Increase scheduler frequency on lesser windows (a side-effect of
6227 // increasing timer accuracy). On greater windows (Win2K or later),
6228 // this is already in effect.
6229 timeBeginPeriod( 1 );
6231 buffersRolling = false;
6232 duplexPrerollBytes = 0;
6234 if ( stream_.mode == DUPLEX ) {
6235 // 0.5 seconds of silence in DUPLEX mode while the devices spin up and synchronize.
6236 duplexPrerollBytes = (int) ( 0.5 * stream_.sampleRate * formatBytes( stream_.deviceFormat[1] ) * stream_.nDeviceChannels[1] );
6240 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
6242 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6243 result = buffer->Play( 0, 0, DSBPLAY_LOOPING );
6244 if ( FAILED( result ) ) {
6245 errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting output buffer!";
6246 errorText_ = errorStream_.str();
6251 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
6253 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6254 result = buffer->Start( DSCBSTART_LOOPING );
6255 if ( FAILED( result ) ) {
6256 errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting input buffer!";
6257 errorText_ = errorStream_.str();
6262 handle->drainCounter = 0;
6263 handle->internalDrain = false;
6264 ResetEvent( handle->condition );
6265 stream_.state = STREAM_RUNNING;
6268 if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );
6271 void RtApiDs :: stopStream()
6274 if ( stream_.state == STREAM_STOPPED ) {
6275 errorText_ = "RtApiDs::stopStream(): the stream is already stopped!";
6276 error( RtAudioError::WARNING );
6283 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6284 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
6285 if ( handle->drainCounter == 0 ) {
6286 handle->drainCounter = 2;
6287 WaitForSingleObject( handle->condition, INFINITE ); // block until signaled
6290 stream_.state = STREAM_STOPPED;
6292 MUTEX_LOCK( &stream_.mutex );
6294 // Stop the buffer and clear memory
6295 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6296 result = buffer->Stop();
6297 if ( FAILED( result ) ) {
6298 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping output buffer!";
6299 errorText_ = errorStream_.str();
6303 // Lock the buffer and clear it so that if we start to play again,
6304 // we won't have old data playing.
6305 result = buffer->Lock( 0, handle->dsBufferSize[0], &audioPtr, &dataLen, NULL, NULL, 0 );
6306 if ( FAILED( result ) ) {
6307 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking output buffer!";
6308 errorText_ = errorStream_.str();
6312 // Zero the DS buffer
6313 ZeroMemory( audioPtr, dataLen );
6315 // Unlock the DS buffer
6316 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
6317 if ( FAILED( result ) ) {
6318 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking output buffer!";
6319 errorText_ = errorStream_.str();
6323 // If we start playing again, we must begin at beginning of buffer.
6324 handle->bufferPointer[0] = 0;
6327 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
6328 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6332 stream_.state = STREAM_STOPPED;
6334 if ( stream_.mode != DUPLEX )
6335 MUTEX_LOCK( &stream_.mutex );
6337 result = buffer->Stop();
6338 if ( FAILED( result ) ) {
6339 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping input buffer!";
6340 errorText_ = errorStream_.str();
6344 // Lock the buffer and clear it so that if we start to play again,
6345 // we won't have old data playing.
6346 result = buffer->Lock( 0, handle->dsBufferSize[1], &audioPtr, &dataLen, NULL, NULL, 0 );
6347 if ( FAILED( result ) ) {
6348 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking input buffer!";
6349 errorText_ = errorStream_.str();
6353 // Zero the DS buffer
6354 ZeroMemory( audioPtr, dataLen );
6356 // Unlock the DS buffer
6357 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
6358 if ( FAILED( result ) ) {
6359 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking input buffer!";
6360 errorText_ = errorStream_.str();
6364 // If we start recording again, we must begin at beginning of buffer.
6365 handle->bufferPointer[1] = 0;
6369 timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows.
6370 MUTEX_UNLOCK( &stream_.mutex );
6372 if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );
6375 void RtApiDs :: abortStream()
6378 if ( stream_.state == STREAM_STOPPED ) {
6379 errorText_ = "RtApiDs::abortStream(): the stream is already stopped!";
6380 error( RtAudioError::WARNING );
6384 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6385 handle->drainCounter = 2;
6390 void RtApiDs :: callbackEvent()
6392 if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) {
6393 Sleep( 50 ); // sleep 50 milliseconds
6397 if ( stream_.state == STREAM_CLOSED ) {
6398 errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!";
6399 error( RtAudioError::WARNING );
6403 CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
6404 DsHandle *handle = (DsHandle *) stream_.apiHandle;
6406 // Check if we were draining the stream and signal is finished.
6407 if ( handle->drainCounter > stream_.nBuffers + 2 ) {
6409 stream_.state = STREAM_STOPPING;
6410 if ( handle->internalDrain == false )
6411 SetEvent( handle->condition );
6417 // Invoke user callback to get fresh output data UNLESS we are
6419 if ( handle->drainCounter == 0 ) {
6420 RtAudioCallback callback = (RtAudioCallback) info->callback;
6421 double streamTime = getStreamTime();
6422 RtAudioStreamStatus status = 0;
6423 if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
6424 status |= RTAUDIO_OUTPUT_UNDERFLOW;
6425 handle->xrun[0] = false;
6427 if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
6428 status |= RTAUDIO_INPUT_OVERFLOW;
6429 handle->xrun[1] = false;
6431 int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
6432 stream_.bufferSize, streamTime, status, info->userData );
6433 if ( cbReturnValue == 2 ) {
6434 stream_.state = STREAM_STOPPING;
6435 handle->drainCounter = 2;
6439 else if ( cbReturnValue == 1 ) {
6440 handle->drainCounter = 1;
6441 handle->internalDrain = true;
6446 DWORD currentWritePointer, safeWritePointer;
6447 DWORD currentReadPointer, safeReadPointer;
6448 UINT nextWritePointer;
6450 LPVOID buffer1 = NULL;
6451 LPVOID buffer2 = NULL;
6452 DWORD bufferSize1 = 0;
6453 DWORD bufferSize2 = 0;
6458 MUTEX_LOCK( &stream_.mutex );
6459 if ( stream_.state == STREAM_STOPPED ) {
6460 MUTEX_UNLOCK( &stream_.mutex );
6464 if ( buffersRolling == false ) {
6465 if ( stream_.mode == DUPLEX ) {
6466 //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
6468 // It takes a while for the devices to get rolling. As a result,
6469 // there's no guarantee that the capture and write device pointers
6470 // will move in lockstep. Wait here for both devices to start
6471 // rolling, and then set our buffer pointers accordingly.
6472 // e.g. Crystal Drivers: the capture buffer starts up 5700 to 9600
6473 // bytes later than the write buffer.
6475 // Stub: a serious risk of having a pre-emptive scheduling round
6476 // take place between the two GetCurrentPosition calls... but I'm
6477 // really not sure how to solve the problem. Temporarily boost to
6478 // Realtime priority, maybe; but I'm not sure what priority the
6479 // DirectSound service threads run at. We *should* be roughly
6480 // within a ms or so of correct.
6482 LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6483 LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6485 DWORD startSafeWritePointer, startSafeReadPointer;
6487 result = dsWriteBuffer->GetCurrentPosition( NULL, &startSafeWritePointer );
6488 if ( FAILED( result ) ) {
6489 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
6490 errorText_ = errorStream_.str();
6491 MUTEX_UNLOCK( &stream_.mutex );
6492 error( RtAudioError::SYSTEM_ERROR );
6495 result = dsCaptureBuffer->GetCurrentPosition( NULL, &startSafeReadPointer );
6496 if ( FAILED( result ) ) {
6497 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
6498 errorText_ = errorStream_.str();
6499 MUTEX_UNLOCK( &stream_.mutex );
6500 error( RtAudioError::SYSTEM_ERROR );
6504 result = dsWriteBuffer->GetCurrentPosition( NULL, &safeWritePointer );
6505 if ( FAILED( result ) ) {
6506 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
6507 errorText_ = errorStream_.str();
6508 MUTEX_UNLOCK( &stream_.mutex );
6509 error( RtAudioError::SYSTEM_ERROR );
6512 result = dsCaptureBuffer->GetCurrentPosition( NULL, &safeReadPointer );
6513 if ( FAILED( result ) ) {
6514 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
6515 errorText_ = errorStream_.str();
6516 MUTEX_UNLOCK( &stream_.mutex );
6517 error( RtAudioError::SYSTEM_ERROR );
6520 if ( safeWritePointer != startSafeWritePointer && safeReadPointer != startSafeReadPointer ) break;
6524 //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
6526 handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];
6527 if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];
6528 handle->bufferPointer[1] = safeReadPointer;
6530 else if ( stream_.mode == OUTPUT ) {
6532 // Set the proper nextWritePosition after initial startup.
6533 LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6534 result = dsWriteBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer );
6535 if ( FAILED( result ) ) {
6536 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
6537 errorText_ = errorStream_.str();
6538 MUTEX_UNLOCK( &stream_.mutex );
6539 error( RtAudioError::SYSTEM_ERROR );
6542 handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];
6543 if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];
6546 buffersRolling = true;
6549 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
6551 LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
6553 if ( handle->drainCounter > 1 ) { // write zeros to the output stream
6554 bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
6555 bufferBytes *= formatBytes( stream_.userFormat );
6556 memset( stream_.userBuffer[0], 0, bufferBytes );
6559 // Setup parameters and do buffer conversion if necessary.
6560 if ( stream_.doConvertBuffer[0] ) {
6561 buffer = stream_.deviceBuffer;
6562 convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
6563 bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[0];
6564 bufferBytes *= formatBytes( stream_.deviceFormat[0] );
6567 buffer = stream_.userBuffer[0];
6568 bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
6569 bufferBytes *= formatBytes( stream_.userFormat );
6572 // No byte swapping necessary in DirectSound implementation.
6574 // Ahhh ... windoze. 16-bit data is signed but 8-bit data is
6575 // unsigned. So, we need to convert our signed 8-bit data here to
6577 if ( stream_.deviceFormat[0] == RTAUDIO_SINT8 )
6578 for ( int i=0; i<bufferBytes; i++ ) buffer[i] = (unsigned char) ( buffer[i] + 128 );
6580 DWORD dsBufferSize = handle->dsBufferSize[0];
6581 nextWritePointer = handle->bufferPointer[0];
6583 DWORD endWrite, leadPointer;
6585 // Find out where the read and "safe write" pointers are.
6586 result = dsBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer );
6587 if ( FAILED( result ) ) {
6588 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
6589 errorText_ = errorStream_.str();
6590 MUTEX_UNLOCK( &stream_.mutex );
6591 error( RtAudioError::SYSTEM_ERROR );
6595 // We will copy our output buffer into the region between
6596 // safeWritePointer and leadPointer. If leadPointer is not
6597 // beyond the next endWrite position, wait until it is.
6598 leadPointer = safeWritePointer + handle->dsPointerLeadTime[0];
6599 //std::cout << "safeWritePointer = " << safeWritePointer << ", leadPointer = " << leadPointer << ", nextWritePointer = " << nextWritePointer << std::endl;
6600 if ( leadPointer > dsBufferSize ) leadPointer -= dsBufferSize;
6601 if ( leadPointer < nextWritePointer ) leadPointer += dsBufferSize; // unwrap offset
6602 endWrite = nextWritePointer + bufferBytes;
6604 // Check whether the entire write region is behind the play pointer.
6605 if ( leadPointer >= endWrite ) break;
6607 // If we are here, then we must wait until the leadPointer advances
6608 // beyond the end of our next write region. We use the
6609 // Sleep() function to suspend operation until that happens.
6610 double millis = ( endWrite - leadPointer ) * 1000.0;
6611 millis /= ( formatBytes( stream_.deviceFormat[0]) * stream_.nDeviceChannels[0] * stream_.sampleRate);
6612 if ( millis < 1.0 ) millis = 1.0;
6613 Sleep( (DWORD) millis );
6616 if ( dsPointerBetween( nextWritePointer, safeWritePointer, currentWritePointer, dsBufferSize )
6617 || dsPointerBetween( endWrite, safeWritePointer, currentWritePointer, dsBufferSize ) ) {
6618 // We've strayed into the forbidden zone ... resync the read pointer.
6619 handle->xrun[0] = true;
6620 nextWritePointer = safeWritePointer + handle->dsPointerLeadTime[0] - bufferBytes;
6621 if ( nextWritePointer >= dsBufferSize ) nextWritePointer -= dsBufferSize;
6622 handle->bufferPointer[0] = nextWritePointer;
6623 endWrite = nextWritePointer + bufferBytes;
6626 // Lock free space in the buffer
6627 result = dsBuffer->Lock( nextWritePointer, bufferBytes, &buffer1,
6628 &bufferSize1, &buffer2, &bufferSize2, 0 );
6629 if ( FAILED( result ) ) {
6630 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking buffer during playback!";
6631 errorText_ = errorStream_.str();
6632 MUTEX_UNLOCK( &stream_.mutex );
6633 error( RtAudioError::SYSTEM_ERROR );
6637 // Copy our buffer into the DS buffer
6638 CopyMemory( buffer1, buffer, bufferSize1 );
6639 if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 );
6641 // Update our buffer offset and unlock sound buffer
6642 dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
6643 if ( FAILED( result ) ) {
6644 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!";
6645 errorText_ = errorStream_.str();
6646 MUTEX_UNLOCK( &stream_.mutex );
6647 error( RtAudioError::SYSTEM_ERROR );
6650 nextWritePointer = ( nextWritePointer + bufferSize1 + bufferSize2 ) % dsBufferSize;
6651 handle->bufferPointer[0] = nextWritePointer;
6654 // Don't bother draining input
6655 if ( handle->drainCounter ) {
6656 handle->drainCounter++;
6660 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
6662 // Setup parameters.
6663 if ( stream_.doConvertBuffer[1] ) {
6664 buffer = stream_.deviceBuffer;
6665 bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[1];
6666 bufferBytes *= formatBytes( stream_.deviceFormat[1] );
6669 buffer = stream_.userBuffer[1];
6670 bufferBytes = stream_.bufferSize * stream_.nUserChannels[1];
6671 bufferBytes *= formatBytes( stream_.userFormat );
6674 LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
6675 long nextReadPointer = handle->bufferPointer[1];
6676 DWORD dsBufferSize = handle->dsBufferSize[1];
6678 // Find out where the write and "safe read" pointers are.
6679 result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer );
6680 if ( FAILED( result ) ) {
6681 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
6682 errorText_ = errorStream_.str();
6683 MUTEX_UNLOCK( &stream_.mutex );
6684 error( RtAudioError::SYSTEM_ERROR );
6688 if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset
6689 DWORD endRead = nextReadPointer + bufferBytes;
6691 // Handling depends on whether we are INPUT or DUPLEX.
6692 // If we're in INPUT mode then waiting is a good thing. If we're in DUPLEX mode,
6693 // then a wait here will drag the write pointers into the forbidden zone.
6695 // In DUPLEX mode, rather than wait, we will back off the read pointer until
6696 // it's in a safe position. This causes dropouts, but it seems to be the only
6697 // practical way to sync up the read and write pointers reliably, given the
6698 // the very complex relationship between phase and increment of the read and write
6701 // In order to minimize audible dropouts in DUPLEX mode, we will
6702 // provide a pre-roll period of 0.5 seconds in which we return
6703 // zeros from the read buffer while the pointers sync up.
6705 if ( stream_.mode == DUPLEX ) {
6706 if ( safeReadPointer < endRead ) {
6707 if ( duplexPrerollBytes <= 0 ) {
6708 // Pre-roll time over. Be more agressive.
6709 int adjustment = endRead-safeReadPointer;
6711 handle->xrun[1] = true;
6713 // - large adjustments: we've probably run out of CPU cycles, so just resync exactly,
6714 // and perform fine adjustments later.
6715 // - small adjustments: back off by twice as much.
6716 if ( adjustment >= 2*bufferBytes )
6717 nextReadPointer = safeReadPointer-2*bufferBytes;
6719 nextReadPointer = safeReadPointer-bufferBytes-adjustment;
6721 if ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;
6725 // In pre=roll time. Just do it.
6726 nextReadPointer = safeReadPointer - bufferBytes;
6727 while ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;
6729 endRead = nextReadPointer + bufferBytes;
6732 else { // mode == INPUT
6733 while ( safeReadPointer < endRead && stream_.callbackInfo.isRunning ) {
6734 // See comments for playback.
6735 double millis = (endRead - safeReadPointer) * 1000.0;
6736 millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate);
6737 if ( millis < 1.0 ) millis = 1.0;
6738 Sleep( (DWORD) millis );
6740 // Wake up and find out where we are now.
6741 result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer );
6742 if ( FAILED( result ) ) {
6743 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
6744 errorText_ = errorStream_.str();
6745 MUTEX_UNLOCK( &stream_.mutex );
6746 error( RtAudioError::SYSTEM_ERROR );
6750 if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset
6754 // Lock free space in the buffer
6755 result = dsBuffer->Lock( nextReadPointer, bufferBytes, &buffer1,
6756 &bufferSize1, &buffer2, &bufferSize2, 0 );
6757 if ( FAILED( result ) ) {
6758 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking capture buffer!";
6759 errorText_ = errorStream_.str();
6760 MUTEX_UNLOCK( &stream_.mutex );
6761 error( RtAudioError::SYSTEM_ERROR );
6765 if ( duplexPrerollBytes <= 0 ) {
6766 // Copy our buffer into the DS buffer
6767 CopyMemory( buffer, buffer1, bufferSize1 );
6768 if ( buffer2 != NULL ) CopyMemory( buffer+bufferSize1, buffer2, bufferSize2 );
6771 memset( buffer, 0, bufferSize1 );
6772 if ( buffer2 != NULL ) memset( buffer + bufferSize1, 0, bufferSize2 );
6773 duplexPrerollBytes -= bufferSize1 + bufferSize2;
6776 // Update our buffer offset and unlock sound buffer
6777 nextReadPointer = ( nextReadPointer + bufferSize1 + bufferSize2 ) % dsBufferSize;
6778 dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
6779 if ( FAILED( result ) ) {
6780 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking capture buffer!";
6781 errorText_ = errorStream_.str();
6782 MUTEX_UNLOCK( &stream_.mutex );
6783 error( RtAudioError::SYSTEM_ERROR );
6786 handle->bufferPointer[1] = nextReadPointer;
6788 // No byte swapping necessary in DirectSound implementation.
6790 // If necessary, convert 8-bit data from unsigned to signed.
6791 if ( stream_.deviceFormat[1] == RTAUDIO_SINT8 )
6792 for ( int j=0; j<bufferBytes; j++ ) buffer[j] = (signed char) ( buffer[j] - 128 );
6794 // Do buffer conversion if necessary.
6795 if ( stream_.doConvertBuffer[1] )
6796 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
6800 MUTEX_UNLOCK( &stream_.mutex );
6801 RtApi::tickStreamTime();
6804 // Definitions for utility functions and callbacks
6805 // specific to the DirectSound implementation.
6807 static unsigned __stdcall callbackHandler( void *ptr )
6809 CallbackInfo *info = (CallbackInfo *) ptr;
6810 RtApiDs *object = (RtApiDs *) info->object;
6811 bool* isRunning = &info->isRunning;
6813 while ( *isRunning == true ) {
6814 object->callbackEvent();
6821 static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
6822 LPCTSTR description,
6826 struct DsProbeData& probeInfo = *(struct DsProbeData*) lpContext;
6827 std::vector<struct DsDevice>& dsDevices = *probeInfo.dsDevices;
6830 bool validDevice = false;
6831 if ( probeInfo.isInput == true ) {
6833 LPDIRECTSOUNDCAPTURE object;
6835 hr = DirectSoundCaptureCreate( lpguid, &object, NULL );
6836 if ( hr != DS_OK ) return TRUE;
6838 caps.dwSize = sizeof(caps);
6839 hr = object->GetCaps( &caps );
6840 if ( hr == DS_OK ) {
6841 if ( caps.dwChannels > 0 && caps.dwFormats > 0 )
6848 LPDIRECTSOUND object;
6849 hr = DirectSoundCreate( lpguid, &object, NULL );
6850 if ( hr != DS_OK ) return TRUE;
6852 caps.dwSize = sizeof(caps);
6853 hr = object->GetCaps( &caps );
6854 if ( hr == DS_OK ) {
6855 if ( caps.dwFlags & DSCAPS_PRIMARYMONO || caps.dwFlags & DSCAPS_PRIMARYSTEREO )
6861 // If good device, then save its name and guid.
6862 std::string name = convertCharPointerToStdString( description );
6863 //if ( name == "Primary Sound Driver" || name == "Primary Sound Capture Driver" )
6864 if ( lpguid == NULL )
6865 name = "Default Device";
6866 if ( validDevice ) {
6867 for ( unsigned int i=0; i<dsDevices.size(); i++ ) {
6868 if ( dsDevices[i].name == name ) {
6869 dsDevices[i].found = true;
6870 if ( probeInfo.isInput ) {
6871 dsDevices[i].id[1] = lpguid;
6872 dsDevices[i].validId[1] = true;
6875 dsDevices[i].id[0] = lpguid;
6876 dsDevices[i].validId[0] = true;
6884 device.found = true;
6885 if ( probeInfo.isInput ) {
6886 device.id[1] = lpguid;
6887 device.validId[1] = true;
6890 device.id[0] = lpguid;
6891 device.validId[0] = true;
6893 dsDevices.push_back( device );
6899 static const char* getErrorString( int code )
6903 case DSERR_ALLOCATED:
6904 return "Already allocated";
6906 case DSERR_CONTROLUNAVAIL:
6907 return "Control unavailable";
6909 case DSERR_INVALIDPARAM:
6910 return "Invalid parameter";
6912 case DSERR_INVALIDCALL:
6913 return "Invalid call";
6916 return "Generic error";
6918 case DSERR_PRIOLEVELNEEDED:
6919 return "Priority level needed";
6921 case DSERR_OUTOFMEMORY:
6922 return "Out of memory";
6924 case DSERR_BADFORMAT:
6925 return "The sample rate or the channel format is not supported";
6927 case DSERR_UNSUPPORTED:
6928 return "Not supported";
6930 case DSERR_NODRIVER:
6933 case DSERR_ALREADYINITIALIZED:
6934 return "Already initialized";
6936 case DSERR_NOAGGREGATION:
6937 return "No aggregation";
6939 case DSERR_BUFFERLOST:
6940 return "Buffer lost";
6942 case DSERR_OTHERAPPHASPRIO:
6943 return "Another application already has priority";
6945 case DSERR_UNINITIALIZED:
6946 return "Uninitialized";
6949 return "DirectSound unknown error";
6952 //******************** End of __WINDOWS_DS__ *********************//
6956 #if defined(__LINUX_ALSA__)
6958 #include <alsa/asoundlib.h>
6961 // A structure to hold various information related to the ALSA API
6964 snd_pcm_t *handles[2];
6967 pthread_cond_t runnable_cv;
6971 :synchronized(false), runnable(false) { xrun[0] = false; xrun[1] = false; }
6974 static void *alsaCallbackHandler( void * ptr );
6976 RtApiAlsa :: RtApiAlsa()
6978 // Nothing to do here.
6981 RtApiAlsa :: ~RtApiAlsa()
6983 if ( stream_.state != STREAM_CLOSED ) closeStream();
6986 unsigned int RtApiAlsa :: getDeviceCount( void )
6988 unsigned nDevices = 0;
6989 int result, subdevice, card;
6993 // Count cards and devices
6995 snd_card_next( &card );
6996 while ( card >= 0 ) {
6997 sprintf( name, "hw:%d", card );
6998 result = snd_ctl_open( &handle, name, 0 );
7000 errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << ".";
7001 errorText_ = errorStream_.str();
7002 error( RtAudioError::WARNING );
7007 result = snd_ctl_pcm_next_device( handle, &subdevice );
7009 errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
7010 errorText_ = errorStream_.str();
7011 error( RtAudioError::WARNING );
7014 if ( subdevice < 0 )
7019 snd_ctl_close( handle );
7020 snd_card_next( &card );
7023 result = snd_ctl_open( &handle, "default", 0 );
7026 snd_ctl_close( handle );
7032 RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device )
7034 RtAudio::DeviceInfo info;
7035 info.probed = false;
7037 unsigned nDevices = 0;
7038 int result, subdevice, card;
7042 // Count cards and devices
7045 snd_card_next( &card );
7046 while ( card >= 0 ) {
7047 sprintf( name, "hw:%d", card );
7048 result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
7050 errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << ".";
7051 errorText_ = errorStream_.str();
7052 error( RtAudioError::WARNING );
7057 result = snd_ctl_pcm_next_device( chandle, &subdevice );
7059 errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
7060 errorText_ = errorStream_.str();
7061 error( RtAudioError::WARNING );
7064 if ( subdevice < 0 ) break;
7065 if ( nDevices == device ) {
7066 sprintf( name, "hw:%d,%d", card, subdevice );
7072 snd_ctl_close( chandle );
7073 snd_card_next( &card );
7076 result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
7077 if ( result == 0 ) {
7078 if ( nDevices == device ) {
7079 strcpy( name, "default" );
7085 if ( nDevices == 0 ) {
7086 errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!";
7087 error( RtAudioError::INVALID_USE );
7091 if ( device >= nDevices ) {
7092 errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!";
7093 error( RtAudioError::INVALID_USE );
7099 // If a stream is already open, we cannot probe the stream devices.
7100 // Thus, use the saved results.
7101 if ( stream_.state != STREAM_CLOSED &&
7102 ( stream_.device[0] == device || stream_.device[1] == device ) ) {
7103 snd_ctl_close( chandle );
7104 if ( device >= devices_.size() ) {
7105 errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened.";
7106 error( RtAudioError::WARNING );
7109 return devices_[ device ];
7112 int openMode = SND_PCM_ASYNC;
7113 snd_pcm_stream_t stream;
7114 snd_pcm_info_t *pcminfo;
7115 snd_pcm_info_alloca( &pcminfo );
7117 snd_pcm_hw_params_t *params;
7118 snd_pcm_hw_params_alloca( ¶ms );
7120 // First try for playback unless default device (which has subdev -1)
7121 stream = SND_PCM_STREAM_PLAYBACK;
7122 snd_pcm_info_set_stream( pcminfo, stream );
7123 if ( subdevice != -1 ) {
7124 snd_pcm_info_set_device( pcminfo, subdevice );
7125 snd_pcm_info_set_subdevice( pcminfo, 0 );
7127 result = snd_ctl_pcm_info( chandle, pcminfo );
7129 // Device probably doesn't support playback.
7134 result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK );
7136 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
7137 errorText_ = errorStream_.str();
7138 error( RtAudioError::WARNING );
7142 // The device is open ... fill the parameter structure.
7143 result = snd_pcm_hw_params_any( phandle, params );
7145 snd_pcm_close( phandle );
7146 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
7147 errorText_ = errorStream_.str();
7148 error( RtAudioError::WARNING );
7152 // Get output channel information.
7154 result = snd_pcm_hw_params_get_channels_max( params, &value );
7156 snd_pcm_close( phandle );
7157 errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << ".";
7158 errorText_ = errorStream_.str();
7159 error( RtAudioError::WARNING );
7162 info.outputChannels = value;
7163 snd_pcm_close( phandle );
7166 stream = SND_PCM_STREAM_CAPTURE;
7167 snd_pcm_info_set_stream( pcminfo, stream );
7169 // Now try for capture unless default device (with subdev = -1)
7170 if ( subdevice != -1 ) {
7171 result = snd_ctl_pcm_info( chandle, pcminfo );
7172 snd_ctl_close( chandle );
7174 // Device probably doesn't support capture.
7175 if ( info.outputChannels == 0 ) return info;
7176 goto probeParameters;
7180 snd_ctl_close( chandle );
7182 result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
7184 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
7185 errorText_ = errorStream_.str();
7186 error( RtAudioError::WARNING );
7187 if ( info.outputChannels == 0 ) return info;
7188 goto probeParameters;
7191 // The device is open ... fill the parameter structure.
7192 result = snd_pcm_hw_params_any( phandle, params );
7194 snd_pcm_close( phandle );
7195 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
7196 errorText_ = errorStream_.str();
7197 error( RtAudioError::WARNING );
7198 if ( info.outputChannels == 0 ) return info;
7199 goto probeParameters;
7202 result = snd_pcm_hw_params_get_channels_max( params, &value );
7204 snd_pcm_close( phandle );
7205 errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << ".";
7206 errorText_ = errorStream_.str();
7207 error( RtAudioError::WARNING );
7208 if ( info.outputChannels == 0 ) return info;
7209 goto probeParameters;
7211 info.inputChannels = value;
7212 snd_pcm_close( phandle );
7214 // If device opens for both playback and capture, we determine the channels.
7215 if ( info.outputChannels > 0 && info.inputChannels > 0 )
7216 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
7218 // ALSA doesn't provide default devices so we'll use the first available one.
7219 if ( device == 0 && info.outputChannels > 0 )
7220 info.isDefaultOutput = true;
7221 if ( device == 0 && info.inputChannels > 0 )
7222 info.isDefaultInput = true;
7225 // At this point, we just need to figure out the supported data
7226 // formats and sample rates. We'll proceed by opening the device in
7227 // the direction with the maximum number of channels, or playback if
7228 // they are equal. This might limit our sample rate options, but so
7231 if ( info.outputChannels >= info.inputChannels )
7232 stream = SND_PCM_STREAM_PLAYBACK;
7234 stream = SND_PCM_STREAM_CAPTURE;
7235 snd_pcm_info_set_stream( pcminfo, stream );
7237 result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
7239 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
7240 errorText_ = errorStream_.str();
7241 error( RtAudioError::WARNING );
7245 // The device is open ... fill the parameter structure.
7246 result = snd_pcm_hw_params_any( phandle, params );
7248 snd_pcm_close( phandle );
7249 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
7250 errorText_ = errorStream_.str();
7251 error( RtAudioError::WARNING );
7255 // Test our discrete set of sample rate values.
7256 info.sampleRates.clear();
7257 for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {
7258 if ( snd_pcm_hw_params_test_rate( phandle, params, SAMPLE_RATES[i], 0 ) == 0 ) {
7259 info.sampleRates.push_back( SAMPLE_RATES[i] );
7261 if ( !info.preferredSampleRate || ( SAMPLE_RATES[i] <= 48000 && SAMPLE_RATES[i] > info.preferredSampleRate ) )
7262 info.preferredSampleRate = SAMPLE_RATES[i];
7265 if ( info.sampleRates.size() == 0 ) {
7266 snd_pcm_close( phandle );
7267 errorStream_ << "RtApiAlsa::getDeviceInfo: no supported sample rates found for device (" << name << ").";
7268 errorText_ = errorStream_.str();
7269 error( RtAudioError::WARNING );
7273 // Probe the supported data formats ... we don't care about endian-ness just yet
7274 snd_pcm_format_t format;
7275 info.nativeFormats = 0;
7276 format = SND_PCM_FORMAT_S8;
7277 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7278 info.nativeFormats |= RTAUDIO_SINT8;
7279 format = SND_PCM_FORMAT_S16;
7280 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7281 info.nativeFormats |= RTAUDIO_SINT16;
7282 format = SND_PCM_FORMAT_S24;
7283 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7284 info.nativeFormats |= RTAUDIO_SINT24;
7285 format = SND_PCM_FORMAT_S32;
7286 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7287 info.nativeFormats |= RTAUDIO_SINT32;
7288 format = SND_PCM_FORMAT_FLOAT;
7289 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7290 info.nativeFormats |= RTAUDIO_FLOAT32;
7291 format = SND_PCM_FORMAT_FLOAT64;
7292 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
7293 info.nativeFormats |= RTAUDIO_FLOAT64;
7295 // Check that we have at least one supported format
7296 if ( info.nativeFormats == 0 ) {
7297 snd_pcm_close( phandle );
7298 errorStream_ << "RtApiAlsa::getDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio.";
7299 errorText_ = errorStream_.str();
7300 error( RtAudioError::WARNING );
7304 // Get the device name
7306 result = snd_card_get_name( card, &cardname );
7307 if ( result >= 0 ) {
7308 sprintf( name, "hw:%s,%d", cardname, subdevice );
7313 // That's all ... close the device and return
7314 snd_pcm_close( phandle );
7319 void RtApiAlsa :: saveDeviceInfo( void )
7323 unsigned int nDevices = getDeviceCount();
7324 devices_.resize( nDevices );
7325 for ( unsigned int i=0; i<nDevices; i++ )
7326 devices_[i] = getDeviceInfo( i );
7329 bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
7330 unsigned int firstChannel, unsigned int sampleRate,
7331 RtAudioFormat format, unsigned int *bufferSize,
7332 RtAudio::StreamOptions *options )
7335 #if defined(__RTAUDIO_DEBUG__)
7337 snd_output_stdio_attach(&out, stderr, 0);
7340 // I'm not using the "plug" interface ... too much inconsistent behavior.
7342 unsigned nDevices = 0;
7343 int result, subdevice, card;
7347 if ( options && options->flags & RTAUDIO_ALSA_USE_DEFAULT )
7348 snprintf(name, sizeof(name), "%s", "default");
7350 // Count cards and devices
7352 snd_card_next( &card );
7353 while ( card >= 0 ) {
7354 sprintf( name, "hw:%d", card );
7355 result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
7357 errorStream_ << "RtApiAlsa::probeDeviceOpen: control open, card = " << card << ", " << snd_strerror( result ) << ".";
7358 errorText_ = errorStream_.str();
7363 result = snd_ctl_pcm_next_device( chandle, &subdevice );
7364 if ( result < 0 ) break;
7365 if ( subdevice < 0 ) break;
7366 if ( nDevices == device ) {
7367 sprintf( name, "hw:%d,%d", card, subdevice );
7368 snd_ctl_close( chandle );
7373 snd_ctl_close( chandle );
7374 snd_card_next( &card );
7377 result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
7378 if ( result == 0 ) {
7379 if ( nDevices == device ) {
7380 strcpy( name, "default" );
7386 if ( nDevices == 0 ) {
7387 // This should not happen because a check is made before this function is called.
7388 errorText_ = "RtApiAlsa::probeDeviceOpen: no devices found!";
7392 if ( device >= nDevices ) {
7393 // This should not happen because a check is made before this function is called.
7394 errorText_ = "RtApiAlsa::probeDeviceOpen: device ID is invalid!";
7401 // The getDeviceInfo() function will not work for a device that is
7402 // already open. Thus, we'll probe the system before opening a
7403 // stream and save the results for use by getDeviceInfo().
7404 if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) // only do once
7405 this->saveDeviceInfo();
7407 snd_pcm_stream_t stream;
7408 if ( mode == OUTPUT )
7409 stream = SND_PCM_STREAM_PLAYBACK;
7411 stream = SND_PCM_STREAM_CAPTURE;
7414 int openMode = SND_PCM_ASYNC;
7415 result = snd_pcm_open( &phandle, name, stream, openMode );
7417 if ( mode == OUTPUT )
7418 errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for output.";
7420 errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for input.";
7421 errorText_ = errorStream_.str();
7425 // Fill the parameter structure.
7426 snd_pcm_hw_params_t *hw_params;
7427 snd_pcm_hw_params_alloca( &hw_params );
7428 result = snd_pcm_hw_params_any( phandle, hw_params );
7430 snd_pcm_close( phandle );
7431 errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") parameters, " << snd_strerror( result ) << ".";
7432 errorText_ = errorStream_.str();
7436 #if defined(__RTAUDIO_DEBUG__)
7437 fprintf( stderr, "\nRtApiAlsa: dump hardware params just after device open:\n\n" );
7438 snd_pcm_hw_params_dump( hw_params, out );
7441 // Set access ... check user preference.
7442 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) {
7443 stream_.userInterleaved = false;
7444 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
7446 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
7447 stream_.deviceInterleaved[mode] = true;
7450 stream_.deviceInterleaved[mode] = false;
7453 stream_.userInterleaved = true;
7454 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
7456 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
7457 stream_.deviceInterleaved[mode] = false;
7460 stream_.deviceInterleaved[mode] = true;
7464 snd_pcm_close( phandle );
7465 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") access, " << snd_strerror( result ) << ".";
7466 errorText_ = errorStream_.str();
7470 // Determine how to set the device format.
7471 stream_.userFormat = format;
7472 snd_pcm_format_t deviceFormat = SND_PCM_FORMAT_UNKNOWN;
7474 if ( format == RTAUDIO_SINT8 )
7475 deviceFormat = SND_PCM_FORMAT_S8;
7476 else if ( format == RTAUDIO_SINT16 )
7477 deviceFormat = SND_PCM_FORMAT_S16;
7478 else if ( format == RTAUDIO_SINT24 )
7479 deviceFormat = SND_PCM_FORMAT_S24;
7480 else if ( format == RTAUDIO_SINT32 )
7481 deviceFormat = SND_PCM_FORMAT_S32;
7482 else if ( format == RTAUDIO_FLOAT32 )
7483 deviceFormat = SND_PCM_FORMAT_FLOAT;
7484 else if ( format == RTAUDIO_FLOAT64 )
7485 deviceFormat = SND_PCM_FORMAT_FLOAT64;
7487 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat) == 0) {
7488 stream_.deviceFormat[mode] = format;
7492 // The user requested format is not natively supported by the device.
7493 deviceFormat = SND_PCM_FORMAT_FLOAT64;
7494 if ( snd_pcm_hw_params_test_format( phandle, hw_params, deviceFormat ) == 0 ) {
7495 stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
7499 deviceFormat = SND_PCM_FORMAT_FLOAT;
7500 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7501 stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
7505 deviceFormat = SND_PCM_FORMAT_S32;
7506 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7507 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
7511 deviceFormat = SND_PCM_FORMAT_S24;
7512 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7513 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
7517 deviceFormat = SND_PCM_FORMAT_S16;
7518 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7519 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
7523 deviceFormat = SND_PCM_FORMAT_S8;
7524 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
7525 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
7529 // If we get here, no supported format was found.
7530 snd_pcm_close( phandle );
7531 errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device " << device << " data format not supported by RtAudio.";
7532 errorText_ = errorStream_.str();
7536 result = snd_pcm_hw_params_set_format( phandle, hw_params, deviceFormat );
7538 snd_pcm_close( phandle );
7539 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") data format, " << snd_strerror( result ) << ".";
7540 errorText_ = errorStream_.str();
7544 // Determine whether byte-swaping is necessary.
7545 stream_.doByteSwap[mode] = false;
7546 if ( deviceFormat != SND_PCM_FORMAT_S8 ) {
7547 result = snd_pcm_format_cpu_endian( deviceFormat );
7549 stream_.doByteSwap[mode] = true;
7550 else if (result < 0) {
7551 snd_pcm_close( phandle );
7552 errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") endian-ness, " << snd_strerror( result ) << ".";
7553 errorText_ = errorStream_.str();
7558 // Set the sample rate.
7559 result = snd_pcm_hw_params_set_rate_near( phandle, hw_params, (unsigned int*) &sampleRate, 0 );
7561 snd_pcm_close( phandle );
7562 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting sample rate on device (" << name << "), " << snd_strerror( result ) << ".";
7563 errorText_ = errorStream_.str();
7567 // Determine the number of channels for this device. We support a possible
7568 // minimum device channel number > than the value requested by the user.
7569 stream_.nUserChannels[mode] = channels;
7571 result = snd_pcm_hw_params_get_channels_max( hw_params, &value );
7572 unsigned int deviceChannels = value;
7573 if ( result < 0 || deviceChannels < channels + firstChannel ) {
7574 snd_pcm_close( phandle );
7575 errorStream_ << "RtApiAlsa::probeDeviceOpen: requested channel parameters not supported by device (" << name << "), " << snd_strerror( result ) << ".";
7576 errorText_ = errorStream_.str();
7580 result = snd_pcm_hw_params_get_channels_min( hw_params, &value );
7582 snd_pcm_close( phandle );
7583 errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting minimum channels for device (" << name << "), " << snd_strerror( result ) << ".";
7584 errorText_ = errorStream_.str();
7587 deviceChannels = value;
7588 if ( deviceChannels < channels + firstChannel ) deviceChannels = channels + firstChannel;
7589 stream_.nDeviceChannels[mode] = deviceChannels;
7591 // Set the device channels.
7592 result = snd_pcm_hw_params_set_channels( phandle, hw_params, deviceChannels );
7594 snd_pcm_close( phandle );
7595 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting channels for device (" << name << "), " << snd_strerror( result ) << ".";
7596 errorText_ = errorStream_.str();
7600 // Set the buffer (or period) size.
7602 snd_pcm_uframes_t periodSize = *bufferSize;
7603 result = snd_pcm_hw_params_set_period_size_near( phandle, hw_params, &periodSize, &dir );
7605 snd_pcm_close( phandle );
7606 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting period size for device (" << name << "), " << snd_strerror( result ) << ".";
7607 errorText_ = errorStream_.str();
7610 *bufferSize = periodSize;
7612 // Set the buffer number, which in ALSA is referred to as the "period".
7613 unsigned int periods = 0;
7614 if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2;
7615 if ( options && options->numberOfBuffers > 0 ) periods = options->numberOfBuffers;
7616 if ( periods < 2 ) periods = 4; // a fairly safe default value
7617 result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir );
7619 snd_pcm_close( phandle );
7620 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << ".";
7621 errorText_ = errorStream_.str();
7625 // If attempting to setup a duplex stream, the bufferSize parameter
7626 // MUST be the same in both directions!
7627 if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {
7628 snd_pcm_close( phandle );
7629 errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ").";
7630 errorText_ = errorStream_.str();
7634 stream_.bufferSize = *bufferSize;
7636 // Install the hardware configuration
7637 result = snd_pcm_hw_params( phandle, hw_params );
7639 snd_pcm_close( phandle );
7640 errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing hardware configuration on device (" << name << "), " << snd_strerror( result ) << ".";
7641 errorText_ = errorStream_.str();
7645 #if defined(__RTAUDIO_DEBUG__)
7646 fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n");
7647 snd_pcm_hw_params_dump( hw_params, out );
7650 // Set the software configuration to fill buffers with zeros and prevent device stopping on xruns.
7651 snd_pcm_sw_params_t *sw_params = NULL;
7652 snd_pcm_sw_params_alloca( &sw_params );
7653 snd_pcm_sw_params_current( phandle, sw_params );
7654 snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize );
7655 snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, ULONG_MAX );
7656 snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 );
7658 // The following two settings were suggested by Theo Veenker
7659 //snd_pcm_sw_params_set_avail_min( phandle, sw_params, *bufferSize );
7660 //snd_pcm_sw_params_set_xfer_align( phandle, sw_params, 1 );
7662 // here are two options for a fix
7663 //snd_pcm_sw_params_set_silence_size( phandle, sw_params, ULONG_MAX );
7664 snd_pcm_uframes_t val;
7665 snd_pcm_sw_params_get_boundary( sw_params, &val );
7666 snd_pcm_sw_params_set_silence_size( phandle, sw_params, val );
7668 result = snd_pcm_sw_params( phandle, sw_params );
7670 snd_pcm_close( phandle );
7671 errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing software configuration on device (" << name << "), " << snd_strerror( result ) << ".";
7672 errorText_ = errorStream_.str();
7676 #if defined(__RTAUDIO_DEBUG__)
7677 fprintf(stderr, "\nRtApiAlsa: dump software params after installation:\n\n");
7678 snd_pcm_sw_params_dump( sw_params, out );
7681 // Set flags for buffer conversion
7682 stream_.doConvertBuffer[mode] = false;
7683 if ( stream_.userFormat != stream_.deviceFormat[mode] )
7684 stream_.doConvertBuffer[mode] = true;
7685 if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
7686 stream_.doConvertBuffer[mode] = true;
7687 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
7688 stream_.nUserChannels[mode] > 1 )
7689 stream_.doConvertBuffer[mode] = true;
7691 // Allocate the ApiHandle if necessary and then save.
7692 AlsaHandle *apiInfo = 0;
7693 if ( stream_.apiHandle == 0 ) {
7695 apiInfo = (AlsaHandle *) new AlsaHandle;
7697 catch ( std::bad_alloc& ) {
7698 errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating AlsaHandle memory.";
7702 if ( pthread_cond_init( &apiInfo->runnable_cv, NULL ) ) {
7703 errorText_ = "RtApiAlsa::probeDeviceOpen: error initializing pthread condition variable.";
7707 stream_.apiHandle = (void *) apiInfo;
7708 apiInfo->handles[0] = 0;
7709 apiInfo->handles[1] = 0;
7712 apiInfo = (AlsaHandle *) stream_.apiHandle;
7714 apiInfo->handles[mode] = phandle;
7717 // Allocate necessary internal buffers.
7718 unsigned long bufferBytes;
7719 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
7720 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
7721 if ( stream_.userBuffer[mode] == NULL ) {
7722 errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating user buffer memory.";
7726 if ( stream_.doConvertBuffer[mode] ) {
7728 bool makeBuffer = true;
7729 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
7730 if ( mode == INPUT ) {
7731 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
7732 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
7733 if ( bufferBytes <= bytesOut ) makeBuffer = false;
7738 bufferBytes *= *bufferSize;
7739 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
7740 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
7741 if ( stream_.deviceBuffer == NULL ) {
7742 errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating device buffer memory.";
7748 stream_.sampleRate = sampleRate;
7749 stream_.nBuffers = periods;
7750 stream_.device[mode] = device;
7751 stream_.state = STREAM_STOPPED;
7753 // Setup the buffer conversion information structure.
7754 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
7756 // Setup thread if necessary.
7757 if ( stream_.mode == OUTPUT && mode == INPUT ) {
7758 // We had already set up an output stream.
7759 stream_.mode = DUPLEX;
7760 // Link the streams if possible.
7761 apiInfo->synchronized = false;
7762 if ( snd_pcm_link( apiInfo->handles[0], apiInfo->handles[1] ) == 0 )
7763 apiInfo->synchronized = true;
7765 errorText_ = "RtApiAlsa::probeDeviceOpen: unable to synchronize input and output devices.";
7766 error( RtAudioError::WARNING );
7770 stream_.mode = mode;
7772 // Setup callback thread.
7773 stream_.callbackInfo.object = (void *) this;
7775 // Set the thread attributes for joinable and realtime scheduling
7776 // priority (optional). The higher priority will only take affect
7777 // if the program is run as root or suid. Note, under Linux
7778 // processes with CAP_SYS_NICE privilege, a user can change
7779 // scheduling policy and priority (thus need not be root). See
7780 // POSIX "capabilities".
7781 pthread_attr_t attr;
7782 pthread_attr_init( &attr );
7783 pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
7784 #ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
7785 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
7786 stream_.callbackInfo.doRealtime = true;
7787 struct sched_param param;
7788 int priority = options->priority;
7789 int min = sched_get_priority_min( SCHED_RR );
7790 int max = sched_get_priority_max( SCHED_RR );
7791 if ( priority < min ) priority = min;
7792 else if ( priority > max ) priority = max;
7793 param.sched_priority = priority;
7795 // Set the policy BEFORE the priority. Otherwise it fails.
7796 pthread_attr_setschedpolicy(&attr, SCHED_RR);
7797 pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM);
7798 // This is definitely required. Otherwise it fails.
7799 pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED);
7800 pthread_attr_setschedparam(&attr, ¶m);
7803 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
7805 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
7808 stream_.callbackInfo.isRunning = true;
7809 result = pthread_create( &stream_.callbackInfo.thread, &attr, alsaCallbackHandler, &stream_.callbackInfo );
7810 pthread_attr_destroy( &attr );
7812 // Failed. Try instead with default attributes.
7813 result = pthread_create( &stream_.callbackInfo.thread, NULL, alsaCallbackHandler, &stream_.callbackInfo );
7815 stream_.callbackInfo.isRunning = false;
7816 errorText_ = "RtApiAlsa::error creating callback thread!";
7826 pthread_cond_destroy( &apiInfo->runnable_cv );
7827 if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
7828 if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
7830 stream_.apiHandle = 0;
7833 if ( phandle) snd_pcm_close( phandle );
7835 for ( int i=0; i<2; i++ ) {
7836 if ( stream_.userBuffer[i] ) {
7837 free( stream_.userBuffer[i] );
7838 stream_.userBuffer[i] = 0;
7842 if ( stream_.deviceBuffer ) {
7843 free( stream_.deviceBuffer );
7844 stream_.deviceBuffer = 0;
7847 stream_.state = STREAM_CLOSED;
7851 void RtApiAlsa :: closeStream()
7853 if ( stream_.state == STREAM_CLOSED ) {
7854 errorText_ = "RtApiAlsa::closeStream(): no open stream to close!";
7855 error( RtAudioError::WARNING );
7859 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
7860 stream_.callbackInfo.isRunning = false;
7861 MUTEX_LOCK( &stream_.mutex );
7862 if ( stream_.state == STREAM_STOPPED ) {
7863 apiInfo->runnable = true;
7864 pthread_cond_signal( &apiInfo->runnable_cv );
7866 MUTEX_UNLOCK( &stream_.mutex );
7867 pthread_join( stream_.callbackInfo.thread, NULL );
7869 if ( stream_.state == STREAM_RUNNING ) {
7870 stream_.state = STREAM_STOPPED;
7871 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
7872 snd_pcm_drop( apiInfo->handles[0] );
7873 if ( stream_.mode == INPUT || stream_.mode == DUPLEX )
7874 snd_pcm_drop( apiInfo->handles[1] );
7878 pthread_cond_destroy( &apiInfo->runnable_cv );
7879 if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
7880 if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
7882 stream_.apiHandle = 0;
7885 for ( int i=0; i<2; i++ ) {
7886 if ( stream_.userBuffer[i] ) {
7887 free( stream_.userBuffer[i] );
7888 stream_.userBuffer[i] = 0;
7892 if ( stream_.deviceBuffer ) {
7893 free( stream_.deviceBuffer );
7894 stream_.deviceBuffer = 0;
7897 stream_.mode = UNINITIALIZED;
7898 stream_.state = STREAM_CLOSED;
7901 void RtApiAlsa :: startStream()
7903 // This method calls snd_pcm_prepare if the device isn't already in that state.
7906 if ( stream_.state == STREAM_RUNNING ) {
7907 errorText_ = "RtApiAlsa::startStream(): the stream is already running!";
7908 error( RtAudioError::WARNING );
7912 MUTEX_LOCK( &stream_.mutex );
7915 snd_pcm_state_t state;
7916 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
7917 snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
7918 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
7919 state = snd_pcm_state( handle[0] );
7920 if ( state != SND_PCM_STATE_PREPARED ) {
7921 result = snd_pcm_prepare( handle[0] );
7923 errorStream_ << "RtApiAlsa::startStream: error preparing output pcm device, " << snd_strerror( result ) << ".";
7924 errorText_ = errorStream_.str();
7930 if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
7931 result = snd_pcm_drop(handle[1]); // fix to remove stale data received since device has been open
7932 state = snd_pcm_state( handle[1] );
7933 if ( state != SND_PCM_STATE_PREPARED ) {
7934 result = snd_pcm_prepare( handle[1] );
7936 errorStream_ << "RtApiAlsa::startStream: error preparing input pcm device, " << snd_strerror( result ) << ".";
7937 errorText_ = errorStream_.str();
7943 stream_.state = STREAM_RUNNING;
7946 apiInfo->runnable = true;
7947 pthread_cond_signal( &apiInfo->runnable_cv );
7948 MUTEX_UNLOCK( &stream_.mutex );
7950 if ( result >= 0 ) return;
7951 error( RtAudioError::SYSTEM_ERROR );
7954 void RtApiAlsa :: stopStream()
7957 if ( stream_.state == STREAM_STOPPED ) {
7958 errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!";
7959 error( RtAudioError::WARNING );
7963 stream_.state = STREAM_STOPPED;
7964 MUTEX_LOCK( &stream_.mutex );
7967 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
7968 snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
7969 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
7970 if ( apiInfo->synchronized )
7971 result = snd_pcm_drop( handle[0] );
7973 result = snd_pcm_drain( handle[0] );
7975 errorStream_ << "RtApiAlsa::stopStream: error draining output pcm device, " << snd_strerror( result ) << ".";
7976 errorText_ = errorStream_.str();
7981 if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
7982 result = snd_pcm_drop( handle[1] );
7984 errorStream_ << "RtApiAlsa::stopStream: error stopping input pcm device, " << snd_strerror( result ) << ".";
7985 errorText_ = errorStream_.str();
7991 apiInfo->runnable = false; // fixes high CPU usage when stopped
7992 MUTEX_UNLOCK( &stream_.mutex );
7994 if ( result >= 0 ) return;
7995 error( RtAudioError::SYSTEM_ERROR );
7998 void RtApiAlsa :: abortStream()
8001 if ( stream_.state == STREAM_STOPPED ) {
8002 errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!";
8003 error( RtAudioError::WARNING );
8007 stream_.state = STREAM_STOPPED;
8008 MUTEX_LOCK( &stream_.mutex );
8011 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
8012 snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
8013 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
8014 result = snd_pcm_drop( handle[0] );
8016 errorStream_ << "RtApiAlsa::abortStream: error aborting output pcm device, " << snd_strerror( result ) << ".";
8017 errorText_ = errorStream_.str();
8022 if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
8023 result = snd_pcm_drop( handle[1] );
8025 errorStream_ << "RtApiAlsa::abortStream: error aborting input pcm device, " << snd_strerror( result ) << ".";
8026 errorText_ = errorStream_.str();
8032 apiInfo->runnable = false; // fixes high CPU usage when stopped
8033 MUTEX_UNLOCK( &stream_.mutex );
8035 if ( result >= 0 ) return;
8036 error( RtAudioError::SYSTEM_ERROR );
8039 void RtApiAlsa :: callbackEvent()
8041 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
8042 if ( stream_.state == STREAM_STOPPED ) {
8043 MUTEX_LOCK( &stream_.mutex );
8044 while ( !apiInfo->runnable )
8045 pthread_cond_wait( &apiInfo->runnable_cv, &stream_.mutex );
8047 if ( stream_.state != STREAM_RUNNING ) {
8048 MUTEX_UNLOCK( &stream_.mutex );
8051 MUTEX_UNLOCK( &stream_.mutex );
8054 if ( stream_.state == STREAM_CLOSED ) {
8055 errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!";
8056 error( RtAudioError::WARNING );
8060 int doStopStream = 0;
8061 RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
8062 double streamTime = getStreamTime();
8063 RtAudioStreamStatus status = 0;
8064 if ( stream_.mode != INPUT && apiInfo->xrun[0] == true ) {
8065 status |= RTAUDIO_OUTPUT_UNDERFLOW;
8066 apiInfo->xrun[0] = false;
8068 if ( stream_.mode != OUTPUT && apiInfo->xrun[1] == true ) {
8069 status |= RTAUDIO_INPUT_OVERFLOW;
8070 apiInfo->xrun[1] = false;
8072 doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
8073 stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
8075 if ( doStopStream == 2 ) {
8080 MUTEX_LOCK( &stream_.mutex );
8082 // The state might change while waiting on a mutex.
8083 if ( stream_.state == STREAM_STOPPED ) goto unlock;
8089 snd_pcm_sframes_t frames;
8090 RtAudioFormat format;
8091 handle = (snd_pcm_t **) apiInfo->handles;
8093 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
8095 // Setup parameters.
8096 if ( stream_.doConvertBuffer[1] ) {
8097 buffer = stream_.deviceBuffer;
8098 channels = stream_.nDeviceChannels[1];
8099 format = stream_.deviceFormat[1];
8102 buffer = stream_.userBuffer[1];
8103 channels = stream_.nUserChannels[1];
8104 format = stream_.userFormat;
8107 // Read samples from device in interleaved/non-interleaved format.
8108 if ( stream_.deviceInterleaved[1] )
8109 result = snd_pcm_readi( handle[1], buffer, stream_.bufferSize );
8111 void *bufs[channels];
8112 size_t offset = stream_.bufferSize * formatBytes( format );
8113 for ( int i=0; i<channels; i++ )
8114 bufs[i] = (void *) (buffer + (i * offset));
8115 result = snd_pcm_readn( handle[1], bufs, stream_.bufferSize );
8118 if ( result < (int) stream_.bufferSize ) {
8119 // Either an error or overrun occured.
8120 if ( result == -EPIPE ) {
8121 snd_pcm_state_t state = snd_pcm_state( handle[1] );
8122 if ( state == SND_PCM_STATE_XRUN ) {
8123 apiInfo->xrun[1] = true;
8124 result = snd_pcm_prepare( handle[1] );
8126 errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after overrun, " << snd_strerror( result ) << ".";
8127 errorText_ = errorStream_.str();
8131 errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
8132 errorText_ = errorStream_.str();
8136 errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << ".";
8137 errorText_ = errorStream_.str();
8139 error( RtAudioError::WARNING );
8143 // Do byte swapping if necessary.
8144 if ( stream_.doByteSwap[1] )
8145 byteSwapBuffer( buffer, stream_.bufferSize * channels, format );
8147 // Do buffer conversion if necessary.
8148 if ( stream_.doConvertBuffer[1] )
8149 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
8151 // Check stream latency
8152 result = snd_pcm_delay( handle[1], &frames );
8153 if ( result == 0 && frames > 0 ) stream_.latency[1] = frames;
8158 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
8160 // Setup parameters and do buffer conversion if necessary.
8161 if ( stream_.doConvertBuffer[0] ) {
8162 buffer = stream_.deviceBuffer;
8163 convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
8164 channels = stream_.nDeviceChannels[0];
8165 format = stream_.deviceFormat[0];
8168 buffer = stream_.userBuffer[0];
8169 channels = stream_.nUserChannels[0];
8170 format = stream_.userFormat;
8173 // Do byte swapping if necessary.
8174 if ( stream_.doByteSwap[0] )
8175 byteSwapBuffer(buffer, stream_.bufferSize * channels, format);
8177 // Write samples to device in interleaved/non-interleaved format.
8178 if ( stream_.deviceInterleaved[0] )
8179 result = snd_pcm_writei( handle[0], buffer, stream_.bufferSize );
8181 void *bufs[channels];
8182 size_t offset = stream_.bufferSize * formatBytes( format );
8183 for ( int i=0; i<channels; i++ )
8184 bufs[i] = (void *) (buffer + (i * offset));
8185 result = snd_pcm_writen( handle[0], bufs, stream_.bufferSize );
8188 if ( result < (int) stream_.bufferSize ) {
8189 // Either an error or underrun occured.
8190 if ( result == -EPIPE ) {
8191 snd_pcm_state_t state = snd_pcm_state( handle[0] );
8192 if ( state == SND_PCM_STATE_XRUN ) {
8193 apiInfo->xrun[0] = true;
8194 result = snd_pcm_prepare( handle[0] );
8196 errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after underrun, " << snd_strerror( result ) << ".";
8197 errorText_ = errorStream_.str();
8200 errorText_ = "RtApiAlsa::callbackEvent: audio write error, underrun.";
8203 errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
8204 errorText_ = errorStream_.str();
8208 errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << ".";
8209 errorText_ = errorStream_.str();
8211 error( RtAudioError::WARNING );
8215 // Check stream latency
8216 result = snd_pcm_delay( handle[0], &frames );
8217 if ( result == 0 && frames > 0 ) stream_.latency[0] = frames;
8221 MUTEX_UNLOCK( &stream_.mutex );
8223 RtApi::tickStreamTime();
8224 if ( doStopStream == 1 ) this->stopStream();
8227 static void *alsaCallbackHandler( void *ptr )
8229 CallbackInfo *info = (CallbackInfo *) ptr;
8230 RtApiAlsa *object = (RtApiAlsa *) info->object;
8231 bool *isRunning = &info->isRunning;
8233 #ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
8234 if ( info->doRealtime ) {
8235 std::cerr << "RtAudio alsa: " <<
8236 (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") <<
8237 "running realtime scheduling" << std::endl;
8241 while ( *isRunning == true ) {
8242 pthread_testcancel();
8243 object->callbackEvent();
8246 pthread_exit( NULL );
8249 //******************** End of __LINUX_ALSA__ *********************//
8252 #if defined(__LINUX_PULSE__)
8254 // Code written by Peter Meerwald, pmeerw@pmeerw.net
8255 // and Tristan Matthews.
8257 #include <pulse/error.h>
8258 #include <pulse/simple.h>
8261 static const unsigned int SUPPORTED_SAMPLERATES[] = { 8000, 16000, 22050, 32000,
8262 44100, 48000, 96000, 0};
8264 struct rtaudio_pa_format_mapping_t {
8265 RtAudioFormat rtaudio_format;
8266 pa_sample_format_t pa_format;
8269 static const rtaudio_pa_format_mapping_t supported_sampleformats[] = {
8270 {RTAUDIO_SINT16, PA_SAMPLE_S16LE},
8271 {RTAUDIO_SINT32, PA_SAMPLE_S32LE},
8272 {RTAUDIO_FLOAT32, PA_SAMPLE_FLOAT32LE},
8273 {0, PA_SAMPLE_INVALID}};
8275 struct PulseAudioHandle {
8279 pthread_cond_t runnable_cv;
8281 PulseAudioHandle() : s_play(0), s_rec(0), runnable(false) { }
8284 RtApiPulse::~RtApiPulse()
8286 if ( stream_.state != STREAM_CLOSED )
8290 unsigned int RtApiPulse::getDeviceCount( void )
8295 RtAudio::DeviceInfo RtApiPulse::getDeviceInfo( unsigned int /*device*/ )
8297 RtAudio::DeviceInfo info;
8299 info.name = "PulseAudio";
8300 info.outputChannels = 2;
8301 info.inputChannels = 2;
8302 info.duplexChannels = 2;
8303 info.isDefaultOutput = true;
8304 info.isDefaultInput = true;
8306 for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr )
8307 info.sampleRates.push_back( *sr );
8309 info.preferredSampleRate = 48000;
8310 info.nativeFormats = RTAUDIO_SINT16 | RTAUDIO_SINT32 | RTAUDIO_FLOAT32;
8315 static void *pulseaudio_callback( void * user )
8317 CallbackInfo *cbi = static_cast<CallbackInfo *>( user );
8318 RtApiPulse *context = static_cast<RtApiPulse *>( cbi->object );
8319 volatile bool *isRunning = &cbi->isRunning;
8321 #ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
8322 if (cbi->doRealtime) {
8323 std::cerr << "RtAudio pulse: " <<
8324 (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") <<
8325 "running realtime scheduling" << std::endl;
8329 while ( *isRunning ) {
8330 pthread_testcancel();
8331 context->callbackEvent();
8334 pthread_exit( NULL );
8337 void RtApiPulse::closeStream( void )
8339 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8341 stream_.callbackInfo.isRunning = false;
8343 MUTEX_LOCK( &stream_.mutex );
8344 if ( stream_.state == STREAM_STOPPED ) {
8345 pah->runnable = true;
8346 pthread_cond_signal( &pah->runnable_cv );
8348 MUTEX_UNLOCK( &stream_.mutex );
8350 pthread_join( pah->thread, 0 );
8351 if ( pah->s_play ) {
8352 pa_simple_flush( pah->s_play, NULL );
8353 pa_simple_free( pah->s_play );
8356 pa_simple_free( pah->s_rec );
8358 pthread_cond_destroy( &pah->runnable_cv );
8360 stream_.apiHandle = 0;
8363 if ( stream_.userBuffer[0] ) {
8364 free( stream_.userBuffer[0] );
8365 stream_.userBuffer[0] = 0;
8367 if ( stream_.userBuffer[1] ) {
8368 free( stream_.userBuffer[1] );
8369 stream_.userBuffer[1] = 0;
8372 stream_.state = STREAM_CLOSED;
8373 stream_.mode = UNINITIALIZED;
8376 void RtApiPulse::callbackEvent( void )
8378 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8380 if ( stream_.state == STREAM_STOPPED ) {
8381 MUTEX_LOCK( &stream_.mutex );
8382 while ( !pah->runnable )
8383 pthread_cond_wait( &pah->runnable_cv, &stream_.mutex );
8385 if ( stream_.state != STREAM_RUNNING ) {
8386 MUTEX_UNLOCK( &stream_.mutex );
8389 MUTEX_UNLOCK( &stream_.mutex );
8392 if ( stream_.state == STREAM_CLOSED ) {
8393 errorText_ = "RtApiPulse::callbackEvent(): the stream is closed ... "
8394 "this shouldn't happen!";
8395 error( RtAudioError::WARNING );
8399 RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
8400 double streamTime = getStreamTime();
8401 RtAudioStreamStatus status = 0;
8402 int doStopStream = callback( stream_.userBuffer[OUTPUT], stream_.userBuffer[INPUT],
8403 stream_.bufferSize, streamTime, status,
8404 stream_.callbackInfo.userData );
8406 if ( doStopStream == 2 ) {
8411 MUTEX_LOCK( &stream_.mutex );
8412 void *pulse_in = stream_.doConvertBuffer[INPUT] ? stream_.deviceBuffer : stream_.userBuffer[INPUT];
8413 void *pulse_out = stream_.doConvertBuffer[OUTPUT] ? stream_.deviceBuffer : stream_.userBuffer[OUTPUT];
8415 if ( stream_.state != STREAM_RUNNING )
8420 if (stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
8421 if ( stream_.doConvertBuffer[OUTPUT] ) {
8422 convertBuffer( stream_.deviceBuffer,
8423 stream_.userBuffer[OUTPUT],
8424 stream_.convertInfo[OUTPUT] );
8425 bytes = stream_.nDeviceChannels[OUTPUT] * stream_.bufferSize *
8426 formatBytes( stream_.deviceFormat[OUTPUT] );
8428 bytes = stream_.nUserChannels[OUTPUT] * stream_.bufferSize *
8429 formatBytes( stream_.userFormat );
8431 if ( pa_simple_write( pah->s_play, pulse_out, bytes, &pa_error ) < 0 ) {
8432 errorStream_ << "RtApiPulse::callbackEvent: audio write error, " <<
8433 pa_strerror( pa_error ) << ".";
8434 errorText_ = errorStream_.str();
8435 error( RtAudioError::WARNING );
8439 if ( stream_.mode == INPUT || stream_.mode == DUPLEX) {
8440 if ( stream_.doConvertBuffer[INPUT] )
8441 bytes = stream_.nDeviceChannels[INPUT] * stream_.bufferSize *
8442 formatBytes( stream_.deviceFormat[INPUT] );
8444 bytes = stream_.nUserChannels[INPUT] * stream_.bufferSize *
8445 formatBytes( stream_.userFormat );
8447 if ( pa_simple_read( pah->s_rec, pulse_in, bytes, &pa_error ) < 0 ) {
8448 errorStream_ << "RtApiPulse::callbackEvent: audio read error, " <<
8449 pa_strerror( pa_error ) << ".";
8450 errorText_ = errorStream_.str();
8451 error( RtAudioError::WARNING );
8453 if ( stream_.doConvertBuffer[INPUT] ) {
8454 convertBuffer( stream_.userBuffer[INPUT],
8455 stream_.deviceBuffer,
8456 stream_.convertInfo[INPUT] );
8461 MUTEX_UNLOCK( &stream_.mutex );
8462 RtApi::tickStreamTime();
8464 if ( doStopStream == 1 )
8468 void RtApiPulse::startStream( void )
8470 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8472 if ( stream_.state == STREAM_CLOSED ) {
8473 errorText_ = "RtApiPulse::startStream(): the stream is not open!";
8474 error( RtAudioError::INVALID_USE );
8477 if ( stream_.state == STREAM_RUNNING ) {
8478 errorText_ = "RtApiPulse::startStream(): the stream is already running!";
8479 error( RtAudioError::WARNING );
8483 MUTEX_LOCK( &stream_.mutex );
8485 stream_.state = STREAM_RUNNING;
8487 pah->runnable = true;
8488 pthread_cond_signal( &pah->runnable_cv );
8489 MUTEX_UNLOCK( &stream_.mutex );
8492 void RtApiPulse::stopStream( void )
8494 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8496 if ( stream_.state == STREAM_CLOSED ) {
8497 errorText_ = "RtApiPulse::stopStream(): the stream is not open!";
8498 error( RtAudioError::INVALID_USE );
8501 if ( stream_.state == STREAM_STOPPED ) {
8502 errorText_ = "RtApiPulse::stopStream(): the stream is already stopped!";
8503 error( RtAudioError::WARNING );
8507 stream_.state = STREAM_STOPPED;
8508 MUTEX_LOCK( &stream_.mutex );
8510 if ( pah && pah->s_play ) {
8512 if ( pa_simple_drain( pah->s_play, &pa_error ) < 0 ) {
8513 errorStream_ << "RtApiPulse::stopStream: error draining output device, " <<
8514 pa_strerror( pa_error ) << ".";
8515 errorText_ = errorStream_.str();
8516 MUTEX_UNLOCK( &stream_.mutex );
8517 error( RtAudioError::SYSTEM_ERROR );
8522 stream_.state = STREAM_STOPPED;
8523 MUTEX_UNLOCK( &stream_.mutex );
8526 void RtApiPulse::abortStream( void )
8528 PulseAudioHandle *pah = static_cast<PulseAudioHandle*>( stream_.apiHandle );
8530 if ( stream_.state == STREAM_CLOSED ) {
8531 errorText_ = "RtApiPulse::abortStream(): the stream is not open!";
8532 error( RtAudioError::INVALID_USE );
8535 if ( stream_.state == STREAM_STOPPED ) {
8536 errorText_ = "RtApiPulse::abortStream(): the stream is already stopped!";
8537 error( RtAudioError::WARNING );
8541 stream_.state = STREAM_STOPPED;
8542 MUTEX_LOCK( &stream_.mutex );
8544 if ( pah && pah->s_play ) {
8546 if ( pa_simple_flush( pah->s_play, &pa_error ) < 0 ) {
8547 errorStream_ << "RtApiPulse::abortStream: error flushing output device, " <<
8548 pa_strerror( pa_error ) << ".";
8549 errorText_ = errorStream_.str();
8550 MUTEX_UNLOCK( &stream_.mutex );
8551 error( RtAudioError::SYSTEM_ERROR );
8556 stream_.state = STREAM_STOPPED;
8557 MUTEX_UNLOCK( &stream_.mutex );
8560 bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode,
8561 unsigned int channels, unsigned int firstChannel,
8562 unsigned int sampleRate, RtAudioFormat format,
8563 unsigned int *bufferSize, RtAudio::StreamOptions *options )
8565 PulseAudioHandle *pah = 0;
8566 unsigned long bufferBytes = 0;
8569 if ( device != 0 ) return false;
8570 if ( mode != INPUT && mode != OUTPUT ) return false;
8571 if ( channels != 1 && channels != 2 ) {
8572 errorText_ = "RtApiPulse::probeDeviceOpen: unsupported number of channels.";
8575 ss.channels = channels;
8577 if ( firstChannel != 0 ) return false;
8579 bool sr_found = false;
8580 for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr ) {
8581 if ( sampleRate == *sr ) {
8583 stream_.sampleRate = sampleRate;
8584 ss.rate = sampleRate;
8589 errorText_ = "RtApiPulse::probeDeviceOpen: unsupported sample rate.";
8594 for ( const rtaudio_pa_format_mapping_t *sf = supported_sampleformats;
8595 sf->rtaudio_format && sf->pa_format != PA_SAMPLE_INVALID; ++sf ) {
8596 if ( format == sf->rtaudio_format ) {
8598 stream_.userFormat = sf->rtaudio_format;
8599 stream_.deviceFormat[mode] = stream_.userFormat;
8600 ss.format = sf->pa_format;
8604 if ( !sf_found ) { // Use internal data format conversion.
8605 stream_.userFormat = format;
8606 stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
8607 ss.format = PA_SAMPLE_FLOAT32LE;
8610 // Set other stream parameters.
8611 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
8612 else stream_.userInterleaved = true;
8613 stream_.deviceInterleaved[mode] = true;
8614 stream_.nBuffers = 1;
8615 stream_.doByteSwap[mode] = false;
8616 stream_.nUserChannels[mode] = channels;
8617 stream_.nDeviceChannels[mode] = channels + firstChannel;
8618 stream_.channelOffset[mode] = 0;
8619 std::string streamName = "RtAudio";
8621 // Set flags for buffer conversion.
8622 stream_.doConvertBuffer[mode] = false;
8623 if ( stream_.userFormat != stream_.deviceFormat[mode] )
8624 stream_.doConvertBuffer[mode] = true;
8625 if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
8626 stream_.doConvertBuffer[mode] = true;
8628 // Allocate necessary internal buffers.
8629 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
8630 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
8631 if ( stream_.userBuffer[mode] == NULL ) {
8632 errorText_ = "RtApiPulse::probeDeviceOpen: error allocating user buffer memory.";
8635 stream_.bufferSize = *bufferSize;
8637 if ( stream_.doConvertBuffer[mode] ) {
8639 bool makeBuffer = true;
8640 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
8641 if ( mode == INPUT ) {
8642 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
8643 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
8644 if ( bufferBytes <= bytesOut ) makeBuffer = false;
8649 bufferBytes *= *bufferSize;
8650 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
8651 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
8652 if ( stream_.deviceBuffer == NULL ) {
8653 errorText_ = "RtApiPulse::probeDeviceOpen: error allocating device buffer memory.";
8659 stream_.device[mode] = device;
8661 // Setup the buffer conversion information structure.
8662 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
8664 if ( !stream_.apiHandle ) {
8665 PulseAudioHandle *pah = new PulseAudioHandle;
8667 errorText_ = "RtApiPulse::probeDeviceOpen: error allocating memory for handle.";
8671 stream_.apiHandle = pah;
8672 if ( pthread_cond_init( &pah->runnable_cv, NULL ) != 0 ) {
8673 errorText_ = "RtApiPulse::probeDeviceOpen: error creating condition variable.";
8677 pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
8680 if ( options && !options->streamName.empty() ) streamName = options->streamName;
8683 pa_buffer_attr buffer_attr;
8684 buffer_attr.fragsize = bufferBytes;
8685 buffer_attr.maxlength = -1;
8687 pah->s_rec = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_RECORD, NULL, "Record", &ss, NULL, &buffer_attr, &error );
8688 if ( !pah->s_rec ) {
8689 errorText_ = "RtApiPulse::probeDeviceOpen: error connecting input to PulseAudio server.";
8694 pah->s_play = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_PLAYBACK, NULL, "Playback", &ss, NULL, NULL, &error );
8695 if ( !pah->s_play ) {
8696 errorText_ = "RtApiPulse::probeDeviceOpen: error connecting output to PulseAudio server.";
8704 if ( stream_.mode == UNINITIALIZED )
8705 stream_.mode = mode;
8706 else if ( stream_.mode == mode )
8709 stream_.mode = DUPLEX;
8711 if ( !stream_.callbackInfo.isRunning ) {
8712 stream_.callbackInfo.object = this;
8714 stream_.state = STREAM_STOPPED;
8715 // Set the thread attributes for joinable and realtime scheduling
8716 // priority (optional). The higher priority will only take affect
8717 // if the program is run as root or suid. Note, under Linux
8718 // processes with CAP_SYS_NICE privilege, a user can change
8719 // scheduling policy and priority (thus need not be root). See
8720 // POSIX "capabilities".
8721 pthread_attr_t attr;
8722 pthread_attr_init( &attr );
8723 pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
8724 #ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
8725 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
8726 stream_.callbackInfo.doRealtime = true;
8727 struct sched_param param;
8728 int priority = options->priority;
8729 int min = sched_get_priority_min( SCHED_RR );
8730 int max = sched_get_priority_max( SCHED_RR );
8731 if ( priority < min ) priority = min;
8732 else if ( priority > max ) priority = max;
8733 param.sched_priority = priority;
8735 // Set the policy BEFORE the priority. Otherwise it fails.
8736 pthread_attr_setschedpolicy(&attr, SCHED_RR);
8737 pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM);
8738 // This is definitely required. Otherwise it fails.
8739 pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED);
8740 pthread_attr_setschedparam(&attr, ¶m);
8743 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
8745 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
8748 stream_.callbackInfo.isRunning = true;
8749 int result = pthread_create( &pah->thread, &attr, pulseaudio_callback, (void *)&stream_.callbackInfo);
8750 pthread_attr_destroy(&attr);
8752 // Failed. Try instead with default attributes.
8753 result = pthread_create( &pah->thread, NULL, pulseaudio_callback, (void *)&stream_.callbackInfo);
8755 stream_.callbackInfo.isRunning = false;
8756 errorText_ = "RtApiPulse::probeDeviceOpen: error creating thread.";
8765 if ( pah && stream_.callbackInfo.isRunning ) {
8766 pthread_cond_destroy( &pah->runnable_cv );
8768 stream_.apiHandle = 0;
8771 for ( int i=0; i<2; i++ ) {
8772 if ( stream_.userBuffer[i] ) {
8773 free( stream_.userBuffer[i] );
8774 stream_.userBuffer[i] = 0;
8778 if ( stream_.deviceBuffer ) {
8779 free( stream_.deviceBuffer );
8780 stream_.deviceBuffer = 0;
8783 stream_.state = STREAM_CLOSED;
8787 //******************** End of __LINUX_PULSE__ *********************//
8790 #if defined(__LINUX_OSS__)
8793 #include <sys/ioctl.h>
8796 #include <sys/soundcard.h>
8800 static void *ossCallbackHandler(void * ptr);
8802 // A structure to hold various information related to the OSS API
8805 int id[2]; // device ids
8808 pthread_cond_t runnable;
8811 :triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
8814 RtApiOss :: RtApiOss()
8816 // Nothing to do here.
8819 RtApiOss :: ~RtApiOss()
8821 if ( stream_.state != STREAM_CLOSED ) closeStream();
8824 unsigned int RtApiOss :: getDeviceCount( void )
8826 int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
8827 if ( mixerfd == -1 ) {
8828 errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'.";
8829 error( RtAudioError::WARNING );
8833 oss_sysinfo sysinfo;
8834 if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) {
8836 errorText_ = "RtApiOss::getDeviceCount: error getting sysinfo, OSS version >= 4.0 is required.";
8837 error( RtAudioError::WARNING );
8842 return sysinfo.numaudios;
8845 RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device )
8847 RtAudio::DeviceInfo info;
8848 info.probed = false;
8850 int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
8851 if ( mixerfd == -1 ) {
8852 errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'.";
8853 error( RtAudioError::WARNING );
8857 oss_sysinfo sysinfo;
8858 int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
8859 if ( result == -1 ) {
8861 errorText_ = "RtApiOss::getDeviceInfo: error getting sysinfo, OSS version >= 4.0 is required.";
8862 error( RtAudioError::WARNING );
8866 unsigned nDevices = sysinfo.numaudios;
8867 if ( nDevices == 0 ) {
8869 errorText_ = "RtApiOss::getDeviceInfo: no devices found!";
8870 error( RtAudioError::INVALID_USE );
8874 if ( device >= nDevices ) {
8876 errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!";
8877 error( RtAudioError::INVALID_USE );
8881 oss_audioinfo ainfo;
8883 result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
8885 if ( result == -1 ) {
8886 errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
8887 errorText_ = errorStream_.str();
8888 error( RtAudioError::WARNING );
8893 if ( ainfo.caps & PCM_CAP_OUTPUT ) info.outputChannels = ainfo.max_channels;
8894 if ( ainfo.caps & PCM_CAP_INPUT ) info.inputChannels = ainfo.max_channels;
8895 if ( ainfo.caps & PCM_CAP_DUPLEX ) {
8896 if ( info.outputChannels > 0 && info.inputChannels > 0 && ainfo.caps & PCM_CAP_DUPLEX )
8897 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
8900 // Probe data formats ... do for input
8901 unsigned long mask = ainfo.iformats;
8902 if ( mask & AFMT_S16_LE || mask & AFMT_S16_BE )
8903 info.nativeFormats |= RTAUDIO_SINT16;
8904 if ( mask & AFMT_S8 )
8905 info.nativeFormats |= RTAUDIO_SINT8;
8906 if ( mask & AFMT_S32_LE || mask & AFMT_S32_BE )
8907 info.nativeFormats |= RTAUDIO_SINT32;
8909 if ( mask & AFMT_FLOAT )
8910 info.nativeFormats |= RTAUDIO_FLOAT32;
8912 if ( mask & AFMT_S24_LE || mask & AFMT_S24_BE )
8913 info.nativeFormats |= RTAUDIO_SINT24;
8915 // Check that we have at least one supported format
8916 if ( info.nativeFormats == 0 ) {
8917 errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio.";
8918 errorText_ = errorStream_.str();
8919 error( RtAudioError::WARNING );
8923 // Probe the supported sample rates.
8924 info.sampleRates.clear();
8925 if ( ainfo.nrates ) {
8926 for ( unsigned int i=0; i<ainfo.nrates; i++ ) {
8927 for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
8928 if ( ainfo.rates[i] == SAMPLE_RATES[k] ) {
8929 info.sampleRates.push_back( SAMPLE_RATES[k] );
8931 if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
8932 info.preferredSampleRate = SAMPLE_RATES[k];
8940 // Check min and max rate values;
8941 for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
8942 if ( ainfo.min_rate <= (int) SAMPLE_RATES[k] && ainfo.max_rate >= (int) SAMPLE_RATES[k] ) {
8943 info.sampleRates.push_back( SAMPLE_RATES[k] );
8945 if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
8946 info.preferredSampleRate = SAMPLE_RATES[k];
8951 if ( info.sampleRates.size() == 0 ) {
8952 errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ").";
8953 errorText_ = errorStream_.str();
8954 error( RtAudioError::WARNING );
8958 info.name = ainfo.name;
8965 bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
8966 unsigned int firstChannel, unsigned int sampleRate,
8967 RtAudioFormat format, unsigned int *bufferSize,
8968 RtAudio::StreamOptions *options )
8970 int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
8971 if ( mixerfd == -1 ) {
8972 errorText_ = "RtApiOss::probeDeviceOpen: error opening '/dev/mixer'.";
8976 oss_sysinfo sysinfo;
8977 int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
8978 if ( result == -1 ) {
8980 errorText_ = "RtApiOss::probeDeviceOpen: error getting sysinfo, OSS version >= 4.0 is required.";
8984 unsigned nDevices = sysinfo.numaudios;
8985 if ( nDevices == 0 ) {
8986 // This should not happen because a check is made before this function is called.
8988 errorText_ = "RtApiOss::probeDeviceOpen: no devices found!";
8992 if ( device >= nDevices ) {
8993 // This should not happen because a check is made before this function is called.
8995 errorText_ = "RtApiOss::probeDeviceOpen: device ID is invalid!";
8999 oss_audioinfo ainfo;
9001 result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
9003 if ( result == -1 ) {
9004 errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
9005 errorText_ = errorStream_.str();
9009 // Check if device supports input or output
9010 if ( ( mode == OUTPUT && !( ainfo.caps & PCM_CAP_OUTPUT ) ) ||
9011 ( mode == INPUT && !( ainfo.caps & PCM_CAP_INPUT ) ) ) {
9012 if ( mode == OUTPUT )
9013 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support output.";
9015 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support input.";
9016 errorText_ = errorStream_.str();
9021 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9022 if ( mode == OUTPUT )
9024 else { // mode == INPUT
9025 if (stream_.mode == OUTPUT && stream_.device[0] == device) {
9026 // We just set the same device for playback ... close and reopen for duplex (OSS only).
9027 close( handle->id[0] );
9029 if ( !( ainfo.caps & PCM_CAP_DUPLEX ) ) {
9030 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support duplex mode.";
9031 errorText_ = errorStream_.str();
9034 // Check that the number previously set channels is the same.
9035 if ( stream_.nUserChannels[0] != channels ) {
9036 errorStream_ << "RtApiOss::probeDeviceOpen: input/output channels must be equal for OSS duplex device (" << ainfo.name << ").";
9037 errorText_ = errorStream_.str();
9046 // Set exclusive access if specified.
9047 if ( options && options->flags & RTAUDIO_HOG_DEVICE ) flags |= O_EXCL;
9049 // Try to open the device.
9051 fd = open( ainfo.devnode, flags, 0 );
9053 if ( errno == EBUSY )
9054 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") is busy.";
9056 errorStream_ << "RtApiOss::probeDeviceOpen: error opening device (" << ainfo.name << ").";
9057 errorText_ = errorStream_.str();
9061 // For duplex operation, specifically set this mode (this doesn't seem to work).
9063 if ( flags | O_RDWR ) {
9064 result = ioctl( fd, SNDCTL_DSP_SETDUPLEX, NULL );
9065 if ( result == -1) {
9066 errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ").";
9067 errorText_ = errorStream_.str();
9073 // Check the device channel support.
9074 stream_.nUserChannels[mode] = channels;
9075 if ( ainfo.max_channels < (int)(channels + firstChannel) ) {
9077 errorStream_ << "RtApiOss::probeDeviceOpen: the device (" << ainfo.name << ") does not support requested channel parameters.";
9078 errorText_ = errorStream_.str();
9082 // Set the number of channels.
9083 int deviceChannels = channels + firstChannel;
9084 result = ioctl( fd, SNDCTL_DSP_CHANNELS, &deviceChannels );
9085 if ( result == -1 || deviceChannels < (int)(channels + firstChannel) ) {
9087 errorStream_ << "RtApiOss::probeDeviceOpen: error setting channel parameters on device (" << ainfo.name << ").";
9088 errorText_ = errorStream_.str();
9091 stream_.nDeviceChannels[mode] = deviceChannels;
9093 // Get the data format mask
9095 result = ioctl( fd, SNDCTL_DSP_GETFMTS, &mask );
9096 if ( result == -1 ) {
9098 errorStream_ << "RtApiOss::probeDeviceOpen: error getting device (" << ainfo.name << ") data formats.";
9099 errorText_ = errorStream_.str();
9103 // Determine how to set the device format.
9104 stream_.userFormat = format;
9105 int deviceFormat = -1;
9106 stream_.doByteSwap[mode] = false;
9107 if ( format == RTAUDIO_SINT8 ) {
9108 if ( mask & AFMT_S8 ) {
9109 deviceFormat = AFMT_S8;
9110 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
9113 else if ( format == RTAUDIO_SINT16 ) {
9114 if ( mask & AFMT_S16_NE ) {
9115 deviceFormat = AFMT_S16_NE;
9116 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
9118 else if ( mask & AFMT_S16_OE ) {
9119 deviceFormat = AFMT_S16_OE;
9120 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
9121 stream_.doByteSwap[mode] = true;
9124 else if ( format == RTAUDIO_SINT24 ) {
9125 if ( mask & AFMT_S24_NE ) {
9126 deviceFormat = AFMT_S24_NE;
9127 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
9129 else if ( mask & AFMT_S24_OE ) {
9130 deviceFormat = AFMT_S24_OE;
9131 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
9132 stream_.doByteSwap[mode] = true;
9135 else if ( format == RTAUDIO_SINT32 ) {
9136 if ( mask & AFMT_S32_NE ) {
9137 deviceFormat = AFMT_S32_NE;
9138 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
9140 else if ( mask & AFMT_S32_OE ) {
9141 deviceFormat = AFMT_S32_OE;
9142 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
9143 stream_.doByteSwap[mode] = true;
9147 if ( deviceFormat == -1 ) {
9148 // The user requested format is not natively supported by the device.
9149 if ( mask & AFMT_S16_NE ) {
9150 deviceFormat = AFMT_S16_NE;
9151 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
9153 else if ( mask & AFMT_S32_NE ) {
9154 deviceFormat = AFMT_S32_NE;
9155 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
9157 else if ( mask & AFMT_S24_NE ) {
9158 deviceFormat = AFMT_S24_NE;
9159 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
9161 else if ( mask & AFMT_S16_OE ) {
9162 deviceFormat = AFMT_S16_OE;
9163 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
9164 stream_.doByteSwap[mode] = true;
9166 else if ( mask & AFMT_S32_OE ) {
9167 deviceFormat = AFMT_S32_OE;
9168 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
9169 stream_.doByteSwap[mode] = true;
9171 else if ( mask & AFMT_S24_OE ) {
9172 deviceFormat = AFMT_S24_OE;
9173 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
9174 stream_.doByteSwap[mode] = true;
9176 else if ( mask & AFMT_S8) {
9177 deviceFormat = AFMT_S8;
9178 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
9182 if ( stream_.deviceFormat[mode] == 0 ) {
9183 // This really shouldn't happen ...
9185 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") data format not supported by RtAudio.";
9186 errorText_ = errorStream_.str();
9190 // Set the data format.
9191 int temp = deviceFormat;
9192 result = ioctl( fd, SNDCTL_DSP_SETFMT, &deviceFormat );
9193 if ( result == -1 || deviceFormat != temp ) {
9195 errorStream_ << "RtApiOss::probeDeviceOpen: error setting data format on device (" << ainfo.name << ").";
9196 errorText_ = errorStream_.str();
9200 // Attempt to set the buffer size. According to OSS, the minimum
9201 // number of buffers is two. The supposed minimum buffer size is 16
9202 // bytes, so that will be our lower bound. The argument to this
9203 // call is in the form 0xMMMMSSSS (hex), where the buffer size (in
9204 // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM.
9205 // We'll check the actual value used near the end of the setup
9207 int ossBufferBytes = *bufferSize * formatBytes( stream_.deviceFormat[mode] ) * deviceChannels;
9208 if ( ossBufferBytes < 16 ) ossBufferBytes = 16;
9210 if ( options ) buffers = options->numberOfBuffers;
9211 if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) buffers = 2;
9212 if ( buffers < 2 ) buffers = 3;
9213 temp = ((int) buffers << 16) + (int)( log10( (double)ossBufferBytes ) / log10( 2.0 ) );
9214 result = ioctl( fd, SNDCTL_DSP_SETFRAGMENT, &temp );
9215 if ( result == -1 ) {
9217 errorStream_ << "RtApiOss::probeDeviceOpen: error setting buffer size on device (" << ainfo.name << ").";
9218 errorText_ = errorStream_.str();
9221 stream_.nBuffers = buffers;
9223 // Save buffer size (in sample frames).
9224 *bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels );
9225 stream_.bufferSize = *bufferSize;
9227 // Set the sample rate.
9228 int srate = sampleRate;
9229 result = ioctl( fd, SNDCTL_DSP_SPEED, &srate );
9230 if ( result == -1 ) {
9232 errorStream_ << "RtApiOss::probeDeviceOpen: error setting sample rate (" << sampleRate << ") on device (" << ainfo.name << ").";
9233 errorText_ = errorStream_.str();
9237 // Verify the sample rate setup worked.
9238 if ( abs( srate - (int)sampleRate ) > 100 ) {
9240 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ").";
9241 errorText_ = errorStream_.str();
9244 stream_.sampleRate = sampleRate;
9246 if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device) {
9247 // We're doing duplex setup here.
9248 stream_.deviceFormat[0] = stream_.deviceFormat[1];
9249 stream_.nDeviceChannels[0] = deviceChannels;
9252 // Set interleaving parameters.
9253 stream_.userInterleaved = true;
9254 stream_.deviceInterleaved[mode] = true;
9255 if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
9256 stream_.userInterleaved = false;
9258 // Set flags for buffer conversion
9259 stream_.doConvertBuffer[mode] = false;
9260 if ( stream_.userFormat != stream_.deviceFormat[mode] )
9261 stream_.doConvertBuffer[mode] = true;
9262 if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
9263 stream_.doConvertBuffer[mode] = true;
9264 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
9265 stream_.nUserChannels[mode] > 1 )
9266 stream_.doConvertBuffer[mode] = true;
9268 // Allocate the stream handles if necessary and then save.
9269 if ( stream_.apiHandle == 0 ) {
9271 handle = new OssHandle;
9273 catch ( std::bad_alloc& ) {
9274 errorText_ = "RtApiOss::probeDeviceOpen: error allocating OssHandle memory.";
9278 if ( pthread_cond_init( &handle->runnable, NULL ) ) {
9279 errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable.";
9283 stream_.apiHandle = (void *) handle;
9286 handle = (OssHandle *) stream_.apiHandle;
9288 handle->id[mode] = fd;
9290 // Allocate necessary internal buffers.
9291 unsigned long bufferBytes;
9292 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
9293 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
9294 if ( stream_.userBuffer[mode] == NULL ) {
9295 errorText_ = "RtApiOss::probeDeviceOpen: error allocating user buffer memory.";
9299 if ( stream_.doConvertBuffer[mode] ) {
9301 bool makeBuffer = true;
9302 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
9303 if ( mode == INPUT ) {
9304 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
9305 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
9306 if ( bufferBytes <= bytesOut ) makeBuffer = false;
9311 bufferBytes *= *bufferSize;
9312 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
9313 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
9314 if ( stream_.deviceBuffer == NULL ) {
9315 errorText_ = "RtApiOss::probeDeviceOpen: error allocating device buffer memory.";
9321 stream_.device[mode] = device;
9322 stream_.state = STREAM_STOPPED;
9324 // Setup the buffer conversion information structure.
9325 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
9327 // Setup thread if necessary.
9328 if ( stream_.mode == OUTPUT && mode == INPUT ) {
9329 // We had already set up an output stream.
9330 stream_.mode = DUPLEX;
9331 if ( stream_.device[0] == device ) handle->id[0] = fd;
9334 stream_.mode = mode;
9336 // Setup callback thread.
9337 stream_.callbackInfo.object = (void *) this;
9339 // Set the thread attributes for joinable and realtime scheduling
9340 // priority. The higher priority will only take affect if the
9341 // program is run as root or suid.
9342 pthread_attr_t attr;
9343 pthread_attr_init( &attr );
9344 pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
9345 #ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
9346 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
9347 stream_.callbackInfo.doRealtime = true;
9348 struct sched_param param;
9349 int priority = options->priority;
9350 int min = sched_get_priority_min( SCHED_RR );
9351 int max = sched_get_priority_max( SCHED_RR );
9352 if ( priority < min ) priority = min;
9353 else if ( priority > max ) priority = max;
9354 param.sched_priority = priority;
9356 // Set the policy BEFORE the priority. Otherwise it fails.
9357 pthread_attr_setschedpolicy(&attr, SCHED_RR);
9358 pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM);
9359 // This is definitely required. Otherwise it fails.
9360 pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED);
9361 pthread_attr_setschedparam(&attr, ¶m);
9364 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
9366 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
9369 stream_.callbackInfo.isRunning = true;
9370 result = pthread_create( &stream_.callbackInfo.thread, &attr, ossCallbackHandler, &stream_.callbackInfo );
9371 pthread_attr_destroy( &attr );
9373 // Failed. Try instead with default attributes.
9374 result = pthread_create( &stream_.callbackInfo.thread, NULL, ossCallbackHandler, &stream_.callbackInfo );
9376 stream_.callbackInfo.isRunning = false;
9377 errorText_ = "RtApiOss::error creating callback thread!";
9387 pthread_cond_destroy( &handle->runnable );
9388 if ( handle->id[0] ) close( handle->id[0] );
9389 if ( handle->id[1] ) close( handle->id[1] );
9391 stream_.apiHandle = 0;
9394 for ( int i=0; i<2; i++ ) {
9395 if ( stream_.userBuffer[i] ) {
9396 free( stream_.userBuffer[i] );
9397 stream_.userBuffer[i] = 0;
9401 if ( stream_.deviceBuffer ) {
9402 free( stream_.deviceBuffer );
9403 stream_.deviceBuffer = 0;
9406 stream_.state = STREAM_CLOSED;
9410 void RtApiOss :: closeStream()
9412 if ( stream_.state == STREAM_CLOSED ) {
9413 errorText_ = "RtApiOss::closeStream(): no open stream to close!";
9414 error( RtAudioError::WARNING );
9418 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9419 stream_.callbackInfo.isRunning = false;
9420 MUTEX_LOCK( &stream_.mutex );
9421 if ( stream_.state == STREAM_STOPPED )
9422 pthread_cond_signal( &handle->runnable );
9423 MUTEX_UNLOCK( &stream_.mutex );
9424 pthread_join( stream_.callbackInfo.thread, NULL );
9426 if ( stream_.state == STREAM_RUNNING ) {
9427 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
9428 ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
9430 ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
9431 stream_.state = STREAM_STOPPED;
9435 pthread_cond_destroy( &handle->runnable );
9436 if ( handle->id[0] ) close( handle->id[0] );
9437 if ( handle->id[1] ) close( handle->id[1] );
9439 stream_.apiHandle = 0;
9442 for ( int i=0; i<2; i++ ) {
9443 if ( stream_.userBuffer[i] ) {
9444 free( stream_.userBuffer[i] );
9445 stream_.userBuffer[i] = 0;
9449 if ( stream_.deviceBuffer ) {
9450 free( stream_.deviceBuffer );
9451 stream_.deviceBuffer = 0;
9454 stream_.mode = UNINITIALIZED;
9455 stream_.state = STREAM_CLOSED;
9458 void RtApiOss :: startStream()
9461 if ( stream_.state == STREAM_RUNNING ) {
9462 errorText_ = "RtApiOss::startStream(): the stream is already running!";
9463 error( RtAudioError::WARNING );
9467 MUTEX_LOCK( &stream_.mutex );
9469 stream_.state = STREAM_RUNNING;
9471 // No need to do anything else here ... OSS automatically starts
9472 // when fed samples.
9474 MUTEX_UNLOCK( &stream_.mutex );
9476 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9477 pthread_cond_signal( &handle->runnable );
9480 void RtApiOss :: stopStream()
9483 if ( stream_.state == STREAM_STOPPED ) {
9484 errorText_ = "RtApiOss::stopStream(): the stream is already stopped!";
9485 error( RtAudioError::WARNING );
9489 MUTEX_LOCK( &stream_.mutex );
9491 // The state might change while waiting on a mutex.
9492 if ( stream_.state == STREAM_STOPPED ) {
9493 MUTEX_UNLOCK( &stream_.mutex );
9498 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9499 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
9501 // Flush the output with zeros a few times.
9504 RtAudioFormat format;
9506 if ( stream_.doConvertBuffer[0] ) {
9507 buffer = stream_.deviceBuffer;
9508 samples = stream_.bufferSize * stream_.nDeviceChannels[0];
9509 format = stream_.deviceFormat[0];
9512 buffer = stream_.userBuffer[0];
9513 samples = stream_.bufferSize * stream_.nUserChannels[0];
9514 format = stream_.userFormat;
9517 memset( buffer, 0, samples * formatBytes(format) );
9518 for ( unsigned int i=0; i<stream_.nBuffers+1; i++ ) {
9519 result = write( handle->id[0], buffer, samples * formatBytes(format) );
9520 if ( result == -1 ) {
9521 errorText_ = "RtApiOss::stopStream: audio write error.";
9522 error( RtAudioError::WARNING );
9526 result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
9527 if ( result == -1 ) {
9528 errorStream_ << "RtApiOss::stopStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
9529 errorText_ = errorStream_.str();
9532 handle->triggered = false;
9535 if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
9536 result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
9537 if ( result == -1 ) {
9538 errorStream_ << "RtApiOss::stopStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
9539 errorText_ = errorStream_.str();
9545 stream_.state = STREAM_STOPPED;
9546 MUTEX_UNLOCK( &stream_.mutex );
9548 if ( result != -1 ) return;
9549 error( RtAudioError::SYSTEM_ERROR );
9552 void RtApiOss :: abortStream()
9555 if ( stream_.state == STREAM_STOPPED ) {
9556 errorText_ = "RtApiOss::abortStream(): the stream is already stopped!";
9557 error( RtAudioError::WARNING );
9561 MUTEX_LOCK( &stream_.mutex );
9563 // The state might change while waiting on a mutex.
9564 if ( stream_.state == STREAM_STOPPED ) {
9565 MUTEX_UNLOCK( &stream_.mutex );
9570 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9571 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
9572 result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
9573 if ( result == -1 ) {
9574 errorStream_ << "RtApiOss::abortStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
9575 errorText_ = errorStream_.str();
9578 handle->triggered = false;
9581 if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
9582 result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
9583 if ( result == -1 ) {
9584 errorStream_ << "RtApiOss::abortStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
9585 errorText_ = errorStream_.str();
9591 stream_.state = STREAM_STOPPED;
9592 MUTEX_UNLOCK( &stream_.mutex );
9594 if ( result != -1 ) return;
9595 error( RtAudioError::SYSTEM_ERROR );
9598 void RtApiOss :: callbackEvent()
9600 OssHandle *handle = (OssHandle *) stream_.apiHandle;
9601 if ( stream_.state == STREAM_STOPPED ) {
9602 MUTEX_LOCK( &stream_.mutex );
9603 pthread_cond_wait( &handle->runnable, &stream_.mutex );
9604 if ( stream_.state != STREAM_RUNNING ) {
9605 MUTEX_UNLOCK( &stream_.mutex );
9608 MUTEX_UNLOCK( &stream_.mutex );
9611 if ( stream_.state == STREAM_CLOSED ) {
9612 errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!";
9613 error( RtAudioError::WARNING );
9617 // Invoke user callback to get fresh output data.
9618 int doStopStream = 0;
9619 RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
9620 double streamTime = getStreamTime();
9621 RtAudioStreamStatus status = 0;
9622 if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
9623 status |= RTAUDIO_OUTPUT_UNDERFLOW;
9624 handle->xrun[0] = false;
9626 if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
9627 status |= RTAUDIO_INPUT_OVERFLOW;
9628 handle->xrun[1] = false;
9630 doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
9631 stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
9632 if ( doStopStream == 2 ) {
9633 this->abortStream();
9637 MUTEX_LOCK( &stream_.mutex );
9639 // The state might change while waiting on a mutex.
9640 if ( stream_.state == STREAM_STOPPED ) goto unlock;
9645 RtAudioFormat format;
9647 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
9649 // Setup parameters and do buffer conversion if necessary.
9650 if ( stream_.doConvertBuffer[0] ) {
9651 buffer = stream_.deviceBuffer;
9652 convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
9653 samples = stream_.bufferSize * stream_.nDeviceChannels[0];
9654 format = stream_.deviceFormat[0];
9657 buffer = stream_.userBuffer[0];
9658 samples = stream_.bufferSize * stream_.nUserChannels[0];
9659 format = stream_.userFormat;
9662 // Do byte swapping if necessary.
9663 if ( stream_.doByteSwap[0] )
9664 byteSwapBuffer( buffer, samples, format );
9666 if ( stream_.mode == DUPLEX && handle->triggered == false ) {
9668 ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
9669 result = write( handle->id[0], buffer, samples * formatBytes(format) );
9670 trig = PCM_ENABLE_INPUT|PCM_ENABLE_OUTPUT;
9671 ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
9672 handle->triggered = true;
9675 // Write samples to device.
9676 result = write( handle->id[0], buffer, samples * formatBytes(format) );
9678 if ( result == -1 ) {
9679 // We'll assume this is an underrun, though there isn't a
9680 // specific means for determining that.
9681 handle->xrun[0] = true;
9682 errorText_ = "RtApiOss::callbackEvent: audio write error.";
9683 error( RtAudioError::WARNING );
9684 // Continue on to input section.
9688 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
9690 // Setup parameters.
9691 if ( stream_.doConvertBuffer[1] ) {
9692 buffer = stream_.deviceBuffer;
9693 samples = stream_.bufferSize * stream_.nDeviceChannels[1];
9694 format = stream_.deviceFormat[1];
9697 buffer = stream_.userBuffer[1];
9698 samples = stream_.bufferSize * stream_.nUserChannels[1];
9699 format = stream_.userFormat;
9702 // Read samples from device.
9703 result = read( handle->id[1], buffer, samples * formatBytes(format) );
9705 if ( result == -1 ) {
9706 // We'll assume this is an overrun, though there isn't a
9707 // specific means for determining that.
9708 handle->xrun[1] = true;
9709 errorText_ = "RtApiOss::callbackEvent: audio read error.";
9710 error( RtAudioError::WARNING );
9714 // Do byte swapping if necessary.
9715 if ( stream_.doByteSwap[1] )
9716 byteSwapBuffer( buffer, samples, format );
9718 // Do buffer conversion if necessary.
9719 if ( stream_.doConvertBuffer[1] )
9720 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
9724 MUTEX_UNLOCK( &stream_.mutex );
9726 RtApi::tickStreamTime();
9727 if ( doStopStream == 1 ) this->stopStream();
9730 static void *ossCallbackHandler( void *ptr )
9732 CallbackInfo *info = (CallbackInfo *) ptr;
9733 RtApiOss *object = (RtApiOss *) info->object;
9734 bool *isRunning = &info->isRunning;
9736 #ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
9737 if (info->doRealtime) {
9738 std::cerr << "RtAudio oss: " <<
9739 (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") <<
9740 "running realtime scheduling" << std::endl;
9744 while ( *isRunning == true ) {
9745 pthread_testcancel();
9746 object->callbackEvent();
9749 pthread_exit( NULL );
9752 //******************** End of __LINUX_OSS__ *********************//
9756 // *************************************************** //
9758 // Protected common (OS-independent) RtAudio methods.
9760 // *************************************************** //
9762 // This method can be modified to control the behavior of error
9763 // message printing.
9764 void RtApi :: error( RtAudioError::Type type )
9766 errorStream_.str(""); // clear the ostringstream
9768 RtAudioErrorCallback errorCallback = (RtAudioErrorCallback) stream_.callbackInfo.errorCallback;
9769 if ( errorCallback ) {
9770 // abortStream() can generate new error messages. Ignore them. Just keep original one.
9772 if ( firstErrorOccurred_ )
9775 firstErrorOccurred_ = true;
9776 const std::string errorMessage = errorText_;
9778 if ( type != RtAudioError::WARNING && stream_.state != STREAM_STOPPED) {
9779 stream_.callbackInfo.isRunning = false; // exit from the thread
9783 errorCallback( type, errorMessage );
9784 firstErrorOccurred_ = false;
9788 if ( type == RtAudioError::WARNING && showWarnings_ == true )
9789 std::cerr << '\n' << errorText_ << "\n\n";
9790 else if ( type != RtAudioError::WARNING )
9791 throw( RtAudioError( errorText_, type ) );
9794 void RtApi :: verifyStream()
9796 if ( stream_.state == STREAM_CLOSED ) {
9797 errorText_ = "RtApi:: a stream is not open!";
9798 error( RtAudioError::INVALID_USE );
9802 void RtApi :: clearStreamInfo()
9804 stream_.mode = UNINITIALIZED;
9805 stream_.state = STREAM_CLOSED;
9806 stream_.sampleRate = 0;
9807 stream_.bufferSize = 0;
9808 stream_.nBuffers = 0;
9809 stream_.userFormat = 0;
9810 stream_.userInterleaved = true;
9811 stream_.streamTime = 0.0;
9812 stream_.apiHandle = 0;
9813 stream_.deviceBuffer = 0;
9814 stream_.callbackInfo.callback = 0;
9815 stream_.callbackInfo.userData = 0;
9816 stream_.callbackInfo.isRunning = false;
9817 stream_.callbackInfo.errorCallback = 0;
9818 for ( int i=0; i<2; i++ ) {
9819 stream_.device[i] = 11111;
9820 stream_.doConvertBuffer[i] = false;
9821 stream_.deviceInterleaved[i] = true;
9822 stream_.doByteSwap[i] = false;
9823 stream_.nUserChannels[i] = 0;
9824 stream_.nDeviceChannels[i] = 0;
9825 stream_.channelOffset[i] = 0;
9826 stream_.deviceFormat[i] = 0;
9827 stream_.latency[i] = 0;
9828 stream_.userBuffer[i] = 0;
9829 stream_.convertInfo[i].channels = 0;
9830 stream_.convertInfo[i].inJump = 0;
9831 stream_.convertInfo[i].outJump = 0;
9832 stream_.convertInfo[i].inFormat = 0;
9833 stream_.convertInfo[i].outFormat = 0;
9834 stream_.convertInfo[i].inOffset.clear();
9835 stream_.convertInfo[i].outOffset.clear();
9839 unsigned int RtApi :: formatBytes( RtAudioFormat format )
9841 if ( format == RTAUDIO_SINT16 )
9843 else if ( format == RTAUDIO_SINT32 || format == RTAUDIO_FLOAT32 )
9845 else if ( format == RTAUDIO_FLOAT64 )
9847 else if ( format == RTAUDIO_SINT24 )
9849 else if ( format == RTAUDIO_SINT8 )
9852 errorText_ = "RtApi::formatBytes: undefined format.";
9853 error( RtAudioError::WARNING );
9858 void RtApi :: setConvertInfo( StreamMode mode, unsigned int firstChannel )
9860 if ( mode == INPUT ) { // convert device to user buffer
9861 stream_.convertInfo[mode].inJump = stream_.nDeviceChannels[1];
9862 stream_.convertInfo[mode].outJump = stream_.nUserChannels[1];
9863 stream_.convertInfo[mode].inFormat = stream_.deviceFormat[1];
9864 stream_.convertInfo[mode].outFormat = stream_.userFormat;
9866 else { // convert user to device buffer
9867 stream_.convertInfo[mode].inJump = stream_.nUserChannels[0];
9868 stream_.convertInfo[mode].outJump = stream_.nDeviceChannels[0];
9869 stream_.convertInfo[mode].inFormat = stream_.userFormat;
9870 stream_.convertInfo[mode].outFormat = stream_.deviceFormat[0];
9873 if ( stream_.convertInfo[mode].inJump < stream_.convertInfo[mode].outJump )
9874 stream_.convertInfo[mode].channels = stream_.convertInfo[mode].inJump;
9876 stream_.convertInfo[mode].channels = stream_.convertInfo[mode].outJump;
9878 // Set up the interleave/deinterleave offsets.
9879 if ( stream_.deviceInterleaved[mode] != stream_.userInterleaved ) {
9880 if ( ( mode == OUTPUT && stream_.deviceInterleaved[mode] ) ||
9881 ( mode == INPUT && stream_.userInterleaved ) ) {
9882 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
9883 stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
9884 stream_.convertInfo[mode].outOffset.push_back( k );
9885 stream_.convertInfo[mode].inJump = 1;
9889 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
9890 stream_.convertInfo[mode].inOffset.push_back( k );
9891 stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
9892 stream_.convertInfo[mode].outJump = 1;
9896 else { // no (de)interleaving
9897 if ( stream_.userInterleaved ) {
9898 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
9899 stream_.convertInfo[mode].inOffset.push_back( k );
9900 stream_.convertInfo[mode].outOffset.push_back( k );
9904 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
9905 stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
9906 stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
9907 stream_.convertInfo[mode].inJump = 1;
9908 stream_.convertInfo[mode].outJump = 1;
9913 // Add channel offset.
9914 if ( firstChannel > 0 ) {
9915 if ( stream_.deviceInterleaved[mode] ) {
9916 if ( mode == OUTPUT ) {
9917 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
9918 stream_.convertInfo[mode].outOffset[k] += firstChannel;
9921 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
9922 stream_.convertInfo[mode].inOffset[k] += firstChannel;
9926 if ( mode == OUTPUT ) {
9927 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
9928 stream_.convertInfo[mode].outOffset[k] += ( firstChannel * stream_.bufferSize );
9931 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
9932 stream_.convertInfo[mode].inOffset[k] += ( firstChannel * stream_.bufferSize );
9938 void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info )
9940 // This function does format conversion, input/output channel compensation, and
9941 // data interleaving/deinterleaving. 24-bit integers are assumed to occupy
9942 // the lower three bytes of a 32-bit integer.
9944 // Clear our device buffer when in/out duplex device channels are different
9945 if ( outBuffer == stream_.deviceBuffer && stream_.mode == DUPLEX &&
9946 ( stream_.nDeviceChannels[0] < stream_.nDeviceChannels[1] ) )
9947 memset( outBuffer, 0, stream_.bufferSize * info.outJump * formatBytes( info.outFormat ) );
9950 if (info.outFormat == RTAUDIO_FLOAT64) {
9952 Float64 *out = (Float64 *)outBuffer;
9954 if (info.inFormat == RTAUDIO_SINT8) {
9955 signed char *in = (signed char *)inBuffer;
9956 scale = 1.0 / 127.5;
9957 for (unsigned int i=0; i<stream_.bufferSize; i++) {
9958 for (j=0; j<info.channels; j++) {
9959 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
9960 out[info.outOffset[j]] += 0.5;
9961 out[info.outOffset[j]] *= scale;
9964 out += info.outJump;
9967 else if (info.inFormat == RTAUDIO_SINT16) {
9968 Int16 *in = (Int16 *)inBuffer;
9969 scale = 1.0 / 32767.5;
9970 for (unsigned int i=0; i<stream_.bufferSize; i++) {
9971 for (j=0; j<info.channels; j++) {
9972 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
9973 out[info.outOffset[j]] += 0.5;
9974 out[info.outOffset[j]] *= scale;
9977 out += info.outJump;
9980 else if (info.inFormat == RTAUDIO_SINT24) {
9981 Int24 *in = (Int24 *)inBuffer;
9982 scale = 1.0 / 8388607.5;
9983 for (unsigned int i=0; i<stream_.bufferSize; i++) {
9984 for (j=0; j<info.channels; j++) {
9985 out[info.outOffset[j]] = (Float64) (in[info.inOffset[j]].asInt());
9986 out[info.outOffset[j]] += 0.5;
9987 out[info.outOffset[j]] *= scale;
9990 out += info.outJump;
9993 else if (info.inFormat == RTAUDIO_SINT32) {
9994 Int32 *in = (Int32 *)inBuffer;
9995 scale = 1.0 / 2147483647.5;
9996 for (unsigned int i=0; i<stream_.bufferSize; i++) {
9997 for (j=0; j<info.channels; j++) {
9998 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
9999 out[info.outOffset[j]] += 0.5;
10000 out[info.outOffset[j]] *= scale;
10003 out += info.outJump;
10006 else if (info.inFormat == RTAUDIO_FLOAT32) {
10007 Float32 *in = (Float32 *)inBuffer;
10008 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10009 for (j=0; j<info.channels; j++) {
10010 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
10013 out += info.outJump;
10016 else if (info.inFormat == RTAUDIO_FLOAT64) {
10017 // Channel compensation and/or (de)interleaving only.
10018 Float64 *in = (Float64 *)inBuffer;
10019 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10020 for (j=0; j<info.channels; j++) {
10021 out[info.outOffset[j]] = in[info.inOffset[j]];
10024 out += info.outJump;
10028 else if (info.outFormat == RTAUDIO_FLOAT32) {
10030 Float32 *out = (Float32 *)outBuffer;
10032 if (info.inFormat == RTAUDIO_SINT8) {
10033 signed char *in = (signed char *)inBuffer;
10034 scale = (Float32) ( 1.0 / 127.5 );
10035 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10036 for (j=0; j<info.channels; j++) {
10037 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
10038 out[info.outOffset[j]] += 0.5;
10039 out[info.outOffset[j]] *= scale;
10042 out += info.outJump;
10045 else if (info.inFormat == RTAUDIO_SINT16) {
10046 Int16 *in = (Int16 *)inBuffer;
10047 scale = (Float32) ( 1.0 / 32767.5 );
10048 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10049 for (j=0; j<info.channels; j++) {
10050 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
10051 out[info.outOffset[j]] += 0.5;
10052 out[info.outOffset[j]] *= scale;
10055 out += info.outJump;
10058 else if (info.inFormat == RTAUDIO_SINT24) {
10059 Int24 *in = (Int24 *)inBuffer;
10060 scale = (Float32) ( 1.0 / 8388607.5 );
10061 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10062 for (j=0; j<info.channels; j++) {
10063 out[info.outOffset[j]] = (Float32) (in[info.inOffset[j]].asInt());
10064 out[info.outOffset[j]] += 0.5;
10065 out[info.outOffset[j]] *= scale;
10068 out += info.outJump;
10071 else if (info.inFormat == RTAUDIO_SINT32) {
10072 Int32 *in = (Int32 *)inBuffer;
10073 scale = (Float32) ( 1.0 / 2147483647.5 );
10074 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10075 for (j=0; j<info.channels; j++) {
10076 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
10077 out[info.outOffset[j]] += 0.5;
10078 out[info.outOffset[j]] *= scale;
10081 out += info.outJump;
10084 else if (info.inFormat == RTAUDIO_FLOAT32) {
10085 // Channel compensation and/or (de)interleaving only.
10086 Float32 *in = (Float32 *)inBuffer;
10087 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10088 for (j=0; j<info.channels; j++) {
10089 out[info.outOffset[j]] = in[info.inOffset[j]];
10092 out += info.outJump;
10095 else if (info.inFormat == RTAUDIO_FLOAT64) {
10096 Float64 *in = (Float64 *)inBuffer;
10097 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10098 for (j=0; j<info.channels; j++) {
10099 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
10102 out += info.outJump;
10106 else if (info.outFormat == RTAUDIO_SINT32) {
10107 Int32 *out = (Int32 *)outBuffer;
10108 if (info.inFormat == RTAUDIO_SINT8) {
10109 signed char *in = (signed char *)inBuffer;
10110 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10111 for (j=0; j<info.channels; j++) {
10112 out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
10113 out[info.outOffset[j]] <<= 24;
10116 out += info.outJump;
10119 else if (info.inFormat == RTAUDIO_SINT16) {
10120 Int16 *in = (Int16 *)inBuffer;
10121 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10122 for (j=0; j<info.channels; j++) {
10123 out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
10124 out[info.outOffset[j]] <<= 16;
10127 out += info.outJump;
10130 else if (info.inFormat == RTAUDIO_SINT24) {
10131 Int24 *in = (Int24 *)inBuffer;
10132 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10133 for (j=0; j<info.channels; j++) {
10134 out[info.outOffset[j]] = (Int32) in[info.inOffset[j]].asInt();
10135 out[info.outOffset[j]] <<= 8;
10138 out += info.outJump;
10141 else if (info.inFormat == RTAUDIO_SINT32) {
10142 // Channel compensation and/or (de)interleaving only.
10143 Int32 *in = (Int32 *)inBuffer;
10144 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10145 for (j=0; j<info.channels; j++) {
10146 out[info.outOffset[j]] = in[info.inOffset[j]];
10149 out += info.outJump;
10152 else if (info.inFormat == RTAUDIO_FLOAT32) {
10153 Float32 *in = (Float32 *)inBuffer;
10154 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10155 for (j=0; j<info.channels; j++) {
10156 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
10159 out += info.outJump;
10162 else if (info.inFormat == RTAUDIO_FLOAT64) {
10163 Float64 *in = (Float64 *)inBuffer;
10164 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10165 for (j=0; j<info.channels; j++) {
10166 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
10169 out += info.outJump;
10173 else if (info.outFormat == RTAUDIO_SINT24) {
10174 Int24 *out = (Int24 *)outBuffer;
10175 if (info.inFormat == RTAUDIO_SINT8) {
10176 signed char *in = (signed char *)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]] << 16);
10180 //out[info.outOffset[j]] <<= 16;
10183 out += info.outJump;
10186 else if (info.inFormat == RTAUDIO_SINT16) {
10187 Int16 *in = (Int16 *)inBuffer;
10188 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10189 for (j=0; j<info.channels; j++) {
10190 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] << 8);
10191 //out[info.outOffset[j]] <<= 8;
10194 out += info.outJump;
10197 else if (info.inFormat == RTAUDIO_SINT24) {
10198 // Channel compensation and/or (de)interleaving only.
10199 Int24 *in = (Int24 *)inBuffer;
10200 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10201 for (j=0; j<info.channels; j++) {
10202 out[info.outOffset[j]] = in[info.inOffset[j]];
10205 out += info.outJump;
10208 else if (info.inFormat == RTAUDIO_SINT32) {
10209 Int32 *in = (Int32 *)inBuffer;
10210 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10211 for (j=0; j<info.channels; j++) {
10212 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] >> 8);
10213 //out[info.outOffset[j]] >>= 8;
10216 out += info.outJump;
10219 else if (info.inFormat == RTAUDIO_FLOAT32) {
10220 Float32 *in = (Float32 *)inBuffer;
10221 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10222 for (j=0; j<info.channels; j++) {
10223 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
10226 out += info.outJump;
10229 else if (info.inFormat == RTAUDIO_FLOAT64) {
10230 Float64 *in = (Float64 *)inBuffer;
10231 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10232 for (j=0; j<info.channels; j++) {
10233 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
10236 out += info.outJump;
10240 else if (info.outFormat == RTAUDIO_SINT16) {
10241 Int16 *out = (Int16 *)outBuffer;
10242 if (info.inFormat == RTAUDIO_SINT8) {
10243 signed char *in = (signed char *)inBuffer;
10244 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10245 for (j=0; j<info.channels; j++) {
10246 out[info.outOffset[j]] = (Int16) in[info.inOffset[j]];
10247 out[info.outOffset[j]] <<= 8;
10250 out += info.outJump;
10253 else if (info.inFormat == RTAUDIO_SINT16) {
10254 // Channel compensation and/or (de)interleaving only.
10255 Int16 *in = (Int16 *)inBuffer;
10256 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10257 for (j=0; j<info.channels; j++) {
10258 out[info.outOffset[j]] = in[info.inOffset[j]];
10261 out += info.outJump;
10264 else if (info.inFormat == RTAUDIO_SINT24) {
10265 Int24 *in = (Int24 *)inBuffer;
10266 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10267 for (j=0; j<info.channels; j++) {
10268 out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]].asInt() >> 8);
10271 out += info.outJump;
10274 else if (info.inFormat == RTAUDIO_SINT32) {
10275 Int32 *in = (Int32 *)inBuffer;
10276 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10277 for (j=0; j<info.channels; j++) {
10278 out[info.outOffset[j]] = (Int16) ((in[info.inOffset[j]] >> 16) & 0x0000ffff);
10281 out += info.outJump;
10284 else if (info.inFormat == RTAUDIO_FLOAT32) {
10285 Float32 *in = (Float32 *)inBuffer;
10286 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10287 for (j=0; j<info.channels; j++) {
10288 out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
10291 out += info.outJump;
10294 else if (info.inFormat == RTAUDIO_FLOAT64) {
10295 Float64 *in = (Float64 *)inBuffer;
10296 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10297 for (j=0; j<info.channels; j++) {
10298 out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
10301 out += info.outJump;
10305 else if (info.outFormat == RTAUDIO_SINT8) {
10306 signed char *out = (signed char *)outBuffer;
10307 if (info.inFormat == RTAUDIO_SINT8) {
10308 // Channel compensation and/or (de)interleaving only.
10309 signed char *in = (signed char *)inBuffer;
10310 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10311 for (j=0; j<info.channels; j++) {
10312 out[info.outOffset[j]] = in[info.inOffset[j]];
10315 out += info.outJump;
10318 if (info.inFormat == RTAUDIO_SINT16) {
10319 Int16 *in = (Int16 *)inBuffer;
10320 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10321 for (j=0; j<info.channels; j++) {
10322 out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 8) & 0x00ff);
10325 out += info.outJump;
10328 else if (info.inFormat == RTAUDIO_SINT24) {
10329 Int24 *in = (Int24 *)inBuffer;
10330 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10331 for (j=0; j<info.channels; j++) {
10332 out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]].asInt() >> 16);
10335 out += info.outJump;
10338 else if (info.inFormat == RTAUDIO_SINT32) {
10339 Int32 *in = (Int32 *)inBuffer;
10340 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10341 for (j=0; j<info.channels; j++) {
10342 out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 24) & 0x000000ff);
10345 out += info.outJump;
10348 else if (info.inFormat == RTAUDIO_FLOAT32) {
10349 Float32 *in = (Float32 *)inBuffer;
10350 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10351 for (j=0; j<info.channels; j++) {
10352 out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
10355 out += info.outJump;
10358 else if (info.inFormat == RTAUDIO_FLOAT64) {
10359 Float64 *in = (Float64 *)inBuffer;
10360 for (unsigned int i=0; i<stream_.bufferSize; i++) {
10361 for (j=0; j<info.channels; j++) {
10362 out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
10365 out += info.outJump;
10371 //static inline uint16_t bswap_16(uint16_t x) { return (x>>8) | (x<<8); }
10372 //static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); }
10373 //static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); }
10375 void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format )
10381 if ( format == RTAUDIO_SINT16 ) {
10382 for ( unsigned int i=0; i<samples; i++ ) {
10383 // Swap 1st and 2nd bytes.
10388 // Increment 2 bytes.
10392 else if ( format == RTAUDIO_SINT32 ||
10393 format == RTAUDIO_FLOAT32 ) {
10394 for ( unsigned int i=0; i<samples; i++ ) {
10395 // Swap 1st and 4th bytes.
10400 // Swap 2nd and 3rd bytes.
10406 // Increment 3 more bytes.
10410 else if ( format == RTAUDIO_SINT24 ) {
10411 for ( unsigned int i=0; i<samples; i++ ) {
10412 // Swap 1st and 3rd bytes.
10417 // Increment 2 more bytes.
10421 else if ( format == RTAUDIO_FLOAT64 ) {
10422 for ( unsigned int i=0; i<samples; i++ ) {
10423 // Swap 1st and 8th bytes
10428 // Swap 2nd and 7th bytes
10434 // Swap 3rd and 6th bytes
10440 // Swap 4th and 5th bytes
10446 // Increment 5 more bytes.
10452 // Indentation settings for Vim and Emacs
10454 // Local Variables:
10455 // c-basic-offset: 2
10456 // indent-tabs-mode: nil
10459 // vim: et sts=2 sw=2