1 /************************************************************************/
\r
3 \brief Realtime audio i/o C++ classes.
\r
5 RtAudio provides a common API (Application Programming Interface)
\r
6 for realtime audio input/output across Linux (native ALSA, Jack,
\r
7 and OSS), Macintosh OS X (CoreAudio and Jack), and Windows
\r
8 (DirectSound, ASIO and WASAPI) operating systems.
\r
10 RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/
\r
12 RtAudio: realtime audio i/o C++ classes
\r
13 Copyright (c) 2001-2014 Gary P. Scavone
\r
15 Permission is hereby granted, free of charge, to any person
\r
16 obtaining a copy of this software and associated documentation files
\r
17 (the "Software"), to deal in the Software without restriction,
\r
18 including without limitation the rights to use, copy, modify, merge,
\r
19 publish, distribute, sublicense, and/or sell copies of the Software,
\r
20 and to permit persons to whom the Software is furnished to do so,
\r
21 subject to the following conditions:
\r
23 The above copyright notice and this permission notice shall be
\r
24 included in all copies or substantial portions of the Software.
\r
26 Any person wishing to distribute modifications to the Software is
\r
27 asked to send the modifications to the original developer so that
\r
28 they can be incorporated into the canonical version. This is,
\r
29 however, not a binding provision of this license.
\r
31 THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
\r
32 EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
\r
33 MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
\r
34 IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
\r
35 ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
\r
36 CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
\r
37 WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
\r
39 /************************************************************************/
\r
41 // RtAudio: Version 4.1.1pre
\r
43 #include "RtAudio.h"
\r
49 // Static variable definitions.
\r
50 const unsigned int RtApi::MAX_SAMPLE_RATES = 14;
\r
51 const unsigned int RtApi::SAMPLE_RATES[] = {
\r
52 4000, 5512, 8000, 9600, 11025, 16000, 22050,
\r
53 32000, 44100, 48000, 88200, 96000, 176400, 192000
\r
56 #if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) || defined(__WINDOWS_WASAPI__)
\r
57 #define MUTEX_INITIALIZE(A) InitializeCriticalSection(A)
\r
58 #define MUTEX_DESTROY(A) DeleteCriticalSection(A)
\r
59 #define MUTEX_LOCK(A) EnterCriticalSection(A)
\r
60 #define MUTEX_UNLOCK(A) LeaveCriticalSection(A)
\r
61 #elif defined(__LINUX_ALSA__) || defined(__LINUX_PULSE__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__)
\r
63 #define MUTEX_INITIALIZE(A) pthread_mutex_init(A, NULL)
\r
64 #define MUTEX_DESTROY(A) pthread_mutex_destroy(A)
\r
65 #define MUTEX_LOCK(A) pthread_mutex_lock(A)
\r
66 #define MUTEX_UNLOCK(A) pthread_mutex_unlock(A)
\r
68 #define MUTEX_INITIALIZE(A) abs(*A) // dummy definitions
\r
69 #define MUTEX_DESTROY(A) abs(*A) // dummy definitions
\r
72 // *************************************************** //
\r
74 // RtAudio definitions.
\r
76 // *************************************************** //
\r
78 std::string RtAudio :: getVersion( void ) throw()
\r
80 return RTAUDIO_VERSION;
\r
83 void RtAudio :: getCompiledApi( std::vector<RtAudio::Api> &apis ) throw()
\r
87 // The order here will control the order of RtAudio's API search in
\r
89 #if defined(__UNIX_JACK__)
\r
90 apis.push_back( UNIX_JACK );
\r
92 #if defined(__LINUX_ALSA__)
\r
93 apis.push_back( LINUX_ALSA );
\r
95 #if defined(__LINUX_PULSE__)
\r
96 apis.push_back( LINUX_PULSE );
\r
98 #if defined(__LINUX_OSS__)
\r
99 apis.push_back( LINUX_OSS );
\r
101 #if defined(__WINDOWS_ASIO__)
\r
102 apis.push_back( WINDOWS_ASIO );
\r
104 #if defined(__WINDOWS_WASAPI__)
\r
105 apis.push_back( WINDOWS_WASAPI );
\r
107 #if defined(__WINDOWS_DS__)
\r
108 apis.push_back( WINDOWS_DS );
\r
110 #if defined(__MACOSX_CORE__)
\r
111 apis.push_back( MACOSX_CORE );
\r
113 #if defined(__RTAUDIO_DUMMY__)
\r
114 apis.push_back( RTAUDIO_DUMMY );
\r
118 void RtAudio :: openRtApi( RtAudio::Api api )
\r
124 #if defined(__UNIX_JACK__)
\r
125 if ( api == UNIX_JACK )
\r
126 rtapi_ = new RtApiJack();
\r
128 #if defined(__LINUX_ALSA__)
\r
129 if ( api == LINUX_ALSA )
\r
130 rtapi_ = new RtApiAlsa();
\r
132 #if defined(__LINUX_PULSE__)
\r
133 if ( api == LINUX_PULSE )
\r
134 rtapi_ = new RtApiPulse();
\r
136 #if defined(__LINUX_OSS__)
\r
137 if ( api == LINUX_OSS )
\r
138 rtapi_ = new RtApiOss();
\r
140 #if defined(__WINDOWS_ASIO__)
\r
141 if ( api == WINDOWS_ASIO )
\r
142 rtapi_ = new RtApiAsio();
\r
144 #if defined(__WINDOWS_WASAPI__)
\r
145 if ( api == WINDOWS_WASAPI )
\r
146 rtapi_ = new RtApiWasapi();
\r
148 #if defined(__WINDOWS_DS__)
\r
149 if ( api == WINDOWS_DS )
\r
150 rtapi_ = new RtApiDs();
\r
152 #if defined(__MACOSX_CORE__)
\r
153 if ( api == MACOSX_CORE )
\r
154 rtapi_ = new RtApiCore();
\r
156 #if defined(__RTAUDIO_DUMMY__)
\r
157 if ( api == RTAUDIO_DUMMY )
\r
158 rtapi_ = new RtApiDummy();
\r
162 RtAudio :: RtAudio( RtAudio::Api api )
\r
166 if ( api != UNSPECIFIED ) {
\r
167 // Attempt to open the specified API.
\r
169 if ( rtapi_ ) return;
\r
171 // No compiled support for specified API value. Issue a debug
\r
172 // warning and continue as if no API was specified.
\r
173 std::cerr << "\nRtAudio: no compiled support for specified API argument!\n" << std::endl;
\r
176 // Iterate through the compiled APIs and return as soon as we find
\r
177 // one with at least one device or we reach the end of the list.
\r
178 std::vector< RtAudio::Api > apis;
\r
179 getCompiledApi( apis );
\r
180 for ( unsigned int i=0; i<apis.size(); i++ ) {
\r
181 openRtApi( apis[i] );
\r
182 if ( rtapi_->getDeviceCount() ) break;
\r
185 if ( rtapi_ ) return;
\r
187 // It should not be possible to get here because the preprocessor
\r
188 // definition __RTAUDIO_DUMMY__ is automatically defined if no
\r
189 // API-specific definitions are passed to the compiler. But just in
\r
190 // case something weird happens, we'll thow an error.
\r
191 std::string errorText = "\nRtAudio: no compiled API support found ... critical error!!\n\n";
\r
192 throw( RtAudioError( errorText, RtAudioError::UNSPECIFIED ) );
\r
195 RtAudio :: ~RtAudio() throw()
\r
201 void RtAudio :: openStream( RtAudio::StreamParameters *outputParameters,
\r
202 RtAudio::StreamParameters *inputParameters,
\r
203 RtAudioFormat format, unsigned int sampleRate,
\r
204 unsigned int *bufferFrames,
\r
205 RtAudioCallback callback, void *userData,
\r
206 RtAudio::StreamOptions *options,
\r
207 RtAudioErrorCallback errorCallback )
\r
209 return rtapi_->openStream( outputParameters, inputParameters, format,
\r
210 sampleRate, bufferFrames, callback,
\r
211 userData, options, errorCallback );
\r
214 // *************************************************** //
\r
216 // Public RtApi definitions (see end of file for
\r
217 // private or protected utility functions).
\r
219 // *************************************************** //
\r
223 stream_.state = STREAM_CLOSED;
\r
224 stream_.mode = UNINITIALIZED;
\r
225 stream_.apiHandle = 0;
\r
226 stream_.userBuffer[0] = 0;
\r
227 stream_.userBuffer[1] = 0;
\r
228 MUTEX_INITIALIZE( &stream_.mutex );
\r
229 showWarnings_ = true;
\r
230 firstErrorOccurred_ = false;
\r
235 MUTEX_DESTROY( &stream_.mutex );
\r
238 void RtApi :: openStream( RtAudio::StreamParameters *oParams,
\r
239 RtAudio::StreamParameters *iParams,
\r
240 RtAudioFormat format, unsigned int sampleRate,
\r
241 unsigned int *bufferFrames,
\r
242 RtAudioCallback callback, void *userData,
\r
243 RtAudio::StreamOptions *options,
\r
244 RtAudioErrorCallback errorCallback )
\r
246 if ( stream_.state != STREAM_CLOSED ) {
\r
247 errorText_ = "RtApi::openStream: a stream is already open!";
\r
248 error( RtAudioError::INVALID_USE );
\r
252 // Clear stream information potentially left from a previously open stream.
\r
255 if ( oParams && oParams->nChannels < 1 ) {
\r
256 errorText_ = "RtApi::openStream: a non-NULL output StreamParameters structure cannot have an nChannels value less than one.";
\r
257 error( RtAudioError::INVALID_USE );
\r
261 if ( iParams && iParams->nChannels < 1 ) {
\r
262 errorText_ = "RtApi::openStream: a non-NULL input StreamParameters structure cannot have an nChannels value less than one.";
\r
263 error( RtAudioError::INVALID_USE );
\r
267 if ( oParams == NULL && iParams == NULL ) {
\r
268 errorText_ = "RtApi::openStream: input and output StreamParameters structures are both NULL!";
\r
269 error( RtAudioError::INVALID_USE );
\r
273 if ( formatBytes(format) == 0 ) {
\r
274 errorText_ = "RtApi::openStream: 'format' parameter value is undefined.";
\r
275 error( RtAudioError::INVALID_USE );
\r
279 unsigned int nDevices = getDeviceCount();
\r
280 unsigned int oChannels = 0;
\r
282 oChannels = oParams->nChannels;
\r
283 if ( oParams->deviceId >= nDevices ) {
\r
284 errorText_ = "RtApi::openStream: output device parameter value is invalid.";
\r
285 error( RtAudioError::INVALID_USE );
\r
290 unsigned int iChannels = 0;
\r
292 iChannels = iParams->nChannels;
\r
293 if ( iParams->deviceId >= nDevices ) {
\r
294 errorText_ = "RtApi::openStream: input device parameter value is invalid.";
\r
295 error( RtAudioError::INVALID_USE );
\r
302 if ( oChannels > 0 ) {
\r
304 result = probeDeviceOpen( oParams->deviceId, OUTPUT, oChannels, oParams->firstChannel,
\r
305 sampleRate, format, bufferFrames, options );
\r
306 if ( result == false ) {
\r
307 error( RtAudioError::SYSTEM_ERROR );
\r
312 if ( iChannels > 0 ) {
\r
314 result = probeDeviceOpen( iParams->deviceId, INPUT, iChannels, iParams->firstChannel,
\r
315 sampleRate, format, bufferFrames, options );
\r
316 if ( result == false ) {
\r
317 if ( oChannels > 0 ) closeStream();
\r
318 error( RtAudioError::SYSTEM_ERROR );
\r
323 stream_.callbackInfo.callback = (void *) callback;
\r
324 stream_.callbackInfo.userData = userData;
\r
325 stream_.callbackInfo.errorCallback = (void *) errorCallback;
\r
327 if ( options ) options->numberOfBuffers = stream_.nBuffers;
\r
328 stream_.state = STREAM_STOPPED;
\r
331 unsigned int RtApi :: getDefaultInputDevice( void )
\r
333 // Should be implemented in subclasses if possible.
\r
337 unsigned int RtApi :: getDefaultOutputDevice( void )
\r
339 // Should be implemented in subclasses if possible.
\r
343 void RtApi :: closeStream( void )
\r
345 // MUST be implemented in subclasses!
\r
349 bool RtApi :: probeDeviceOpen( unsigned int /*device*/, StreamMode /*mode*/, unsigned int /*channels*/,
\r
350 unsigned int /*firstChannel*/, unsigned int /*sampleRate*/,
\r
351 RtAudioFormat /*format*/, unsigned int * /*bufferSize*/,
\r
352 RtAudio::StreamOptions * /*options*/ )
\r
354 // MUST be implemented in subclasses!
\r
358 void RtApi :: tickStreamTime( void )
\r
360 // Subclasses that do not provide their own implementation of
\r
361 // getStreamTime should call this function once per buffer I/O to
\r
362 // provide basic stream time support.
\r
364 stream_.streamTime += ( stream_.bufferSize * 1.0 / stream_.sampleRate );
\r
366 #if defined( HAVE_GETTIMEOFDAY )
\r
367 gettimeofday( &stream_.lastTickTimestamp, NULL );
\r
371 long RtApi :: getStreamLatency( void )
\r
375 long totalLatency = 0;
\r
376 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
\r
377 totalLatency = stream_.latency[0];
\r
378 if ( stream_.mode == INPUT || stream_.mode == DUPLEX )
\r
379 totalLatency += stream_.latency[1];
\r
381 return totalLatency;
\r
384 double RtApi :: getStreamTime( void )
\r
388 #if defined( HAVE_GETTIMEOFDAY )
\r
389 // Return a very accurate estimate of the stream time by
\r
390 // adding in the elapsed time since the last tick.
\r
391 struct timeval then;
\r
392 struct timeval now;
\r
394 if ( stream_.state != STREAM_RUNNING || stream_.streamTime == 0.0 )
\r
395 return stream_.streamTime;
\r
397 gettimeofday( &now, NULL );
\r
398 then = stream_.lastTickTimestamp;
\r
399 return stream_.streamTime +
\r
400 ((now.tv_sec + 0.000001 * now.tv_usec) -
\r
401 (then.tv_sec + 0.000001 * then.tv_usec));
\r
403 return stream_.streamTime;
\r
407 unsigned int RtApi :: getStreamSampleRate( void )
\r
411 return stream_.sampleRate;
\r
415 // *************************************************** //
\r
417 // OS/API-specific methods.
\r
419 // *************************************************** //
\r
421 #if defined(__MACOSX_CORE__)
\r
423 // The OS X CoreAudio API is designed to use a separate callback
\r
424 // procedure for each of its audio devices. A single RtAudio duplex
\r
425 // stream using two different devices is supported here, though it
\r
426 // cannot be guaranteed to always behave correctly because we cannot
\r
427 // synchronize these two callbacks.
\r
429 // A property listener is installed for over/underrun information.
\r
430 // However, no functionality is currently provided to allow property
\r
431 // listeners to trigger user handlers because it is unclear what could
\r
432 // be done if a critical stream parameter (buffer size, sample rate,
\r
433 // device disconnect) notification arrived. The listeners entail
\r
434 // quite a bit of extra code and most likely, a user program wouldn't
\r
435 // be prepared for the result anyway. However, we do provide a flag
\r
436 // to the client callback function to inform of an over/underrun.
\r
438 // A structure to hold various information related to the CoreAudio API
\r
440 struct CoreHandle {
\r
441 AudioDeviceID id[2]; // device ids
\r
442 #if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
\r
443 AudioDeviceIOProcID procId[2];
\r
445 UInt32 iStream[2]; // device stream index (or first if using multiple)
\r
446 UInt32 nStreams[2]; // number of streams to use
\r
448 char *deviceBuffer;
\r
449 pthread_cond_t condition;
\r
450 int drainCounter; // Tracks callback counts when draining
\r
451 bool internalDrain; // Indicates if stop is initiated from callback or not.
\r
454 :deviceBuffer(0), drainCounter(0), internalDrain(false) { nStreams[0] = 1; nStreams[1] = 1; id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
\r
457 RtApiCore:: RtApiCore()
\r
459 #if defined( AVAILABLE_MAC_OS_X_VERSION_10_6_AND_LATER )
\r
460 // This is a largely undocumented but absolutely necessary
\r
461 // requirement starting with OS-X 10.6. If not called, queries and
\r
462 // updates to various audio device properties are not handled
\r
464 CFRunLoopRef theRunLoop = NULL;
\r
465 AudioObjectPropertyAddress property = { kAudioHardwarePropertyRunLoop,
\r
466 kAudioObjectPropertyScopeGlobal,
\r
467 kAudioObjectPropertyElementMaster };
\r
468 OSStatus result = AudioObjectSetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, sizeof(CFRunLoopRef), &theRunLoop);
\r
469 if ( result != noErr ) {
\r
470 errorText_ = "RtApiCore::RtApiCore: error setting run loop property!";
\r
471 error( RtAudioError::WARNING );
\r
476 RtApiCore :: ~RtApiCore()
\r
478 // The subclass destructor gets called before the base class
\r
479 // destructor, so close an existing stream before deallocating
\r
480 // apiDeviceId memory.
\r
481 if ( stream_.state != STREAM_CLOSED ) closeStream();
\r
484 unsigned int RtApiCore :: getDeviceCount( void )
\r
486 // Find out how many audio devices there are, if any.
\r
488 AudioObjectPropertyAddress propertyAddress = { kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
\r
489 OSStatus result = AudioObjectGetPropertyDataSize( kAudioObjectSystemObject, &propertyAddress, 0, NULL, &dataSize );
\r
490 if ( result != noErr ) {
\r
491 errorText_ = "RtApiCore::getDeviceCount: OS-X error getting device info!";
\r
492 error( RtAudioError::WARNING );
\r
496 return dataSize / sizeof( AudioDeviceID );
\r
499 unsigned int RtApiCore :: getDefaultInputDevice( void )
\r
501 unsigned int nDevices = getDeviceCount();
\r
502 if ( nDevices <= 1 ) return 0;
\r
505 UInt32 dataSize = sizeof( AudioDeviceID );
\r
506 AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultInputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
\r
507 OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id );
\r
508 if ( result != noErr ) {
\r
509 errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device.";
\r
510 error( RtAudioError::WARNING );
\r
514 dataSize *= nDevices;
\r
515 AudioDeviceID deviceList[ nDevices ];
\r
516 property.mSelector = kAudioHardwarePropertyDevices;
\r
517 result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList );
\r
518 if ( result != noErr ) {
\r
519 errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device IDs.";
\r
520 error( RtAudioError::WARNING );
\r
524 for ( unsigned int i=0; i<nDevices; i++ )
\r
525 if ( id == deviceList[i] ) return i;
\r
527 errorText_ = "RtApiCore::getDefaultInputDevice: No default device found!";
\r
528 error( RtAudioError::WARNING );
\r
532 unsigned int RtApiCore :: getDefaultOutputDevice( void )
\r
534 unsigned int nDevices = getDeviceCount();
\r
535 if ( nDevices <= 1 ) return 0;
\r
538 UInt32 dataSize = sizeof( AudioDeviceID );
\r
539 AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultOutputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
\r
540 OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id );
\r
541 if ( result != noErr ) {
\r
542 errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device.";
\r
543 error( RtAudioError::WARNING );
\r
547 dataSize = sizeof( AudioDeviceID ) * nDevices;
\r
548 AudioDeviceID deviceList[ nDevices ];
\r
549 property.mSelector = kAudioHardwarePropertyDevices;
\r
550 result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList );
\r
551 if ( result != noErr ) {
\r
552 errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device IDs.";
\r
553 error( RtAudioError::WARNING );
\r
557 for ( unsigned int i=0; i<nDevices; i++ )
\r
558 if ( id == deviceList[i] ) return i;
\r
560 errorText_ = "RtApiCore::getDefaultOutputDevice: No default device found!";
\r
561 error( RtAudioError::WARNING );
\r
565 RtAudio::DeviceInfo RtApiCore :: getDeviceInfo( unsigned int device )
\r
567 RtAudio::DeviceInfo info;
\r
568 info.probed = false;
\r
571 unsigned int nDevices = getDeviceCount();
\r
572 if ( nDevices == 0 ) {
\r
573 errorText_ = "RtApiCore::getDeviceInfo: no devices found!";
\r
574 error( RtAudioError::INVALID_USE );
\r
578 if ( device >= nDevices ) {
\r
579 errorText_ = "RtApiCore::getDeviceInfo: device ID is invalid!";
\r
580 error( RtAudioError::INVALID_USE );
\r
584 AudioDeviceID deviceList[ nDevices ];
\r
585 UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;
\r
586 AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
\r
587 kAudioObjectPropertyScopeGlobal,
\r
588 kAudioObjectPropertyElementMaster };
\r
589 OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property,
\r
590 0, NULL, &dataSize, (void *) &deviceList );
\r
591 if ( result != noErr ) {
\r
592 errorText_ = "RtApiCore::getDeviceInfo: OS-X system error getting device IDs.";
\r
593 error( RtAudioError::WARNING );
\r
597 AudioDeviceID id = deviceList[ device ];
\r
599 // Get the device name.
\r
601 CFStringRef cfname;
\r
602 dataSize = sizeof( CFStringRef );
\r
603 property.mSelector = kAudioObjectPropertyManufacturer;
\r
604 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname );
\r
605 if ( result != noErr ) {
\r
606 errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device manufacturer.";
\r
607 errorText_ = errorStream_.str();
\r
608 error( RtAudioError::WARNING );
\r
612 //const char *mname = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() );
\r
613 int length = CFStringGetLength(cfname);
\r
614 char *mname = (char *)malloc(length * 3 + 1);
\r
615 #if defined( UNICODE ) || defined( _UNICODE )
\r
616 CFStringGetCString(cfname, mname, length * 3 + 1, kCFStringEncodingUTF8);
\r
618 CFStringGetCString(cfname, mname, length * 3 + 1, CFStringGetSystemEncoding());
\r
620 info.name.append( (const char *)mname, strlen(mname) );
\r
621 info.name.append( ": " );
\r
622 CFRelease( cfname );
\r
625 property.mSelector = kAudioObjectPropertyName;
\r
626 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname );
\r
627 if ( result != noErr ) {
\r
628 errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device name.";
\r
629 errorText_ = errorStream_.str();
\r
630 error( RtAudioError::WARNING );
\r
634 //const char *name = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() );
\r
635 length = CFStringGetLength(cfname);
\r
636 char *name = (char *)malloc(length * 3 + 1);
\r
637 #if defined( UNICODE ) || defined( _UNICODE )
\r
638 CFStringGetCString(cfname, name, length * 3 + 1, kCFStringEncodingUTF8);
\r
640 CFStringGetCString(cfname, name, length * 3 + 1, CFStringGetSystemEncoding());
\r
642 info.name.append( (const char *)name, strlen(name) );
\r
643 CFRelease( cfname );
\r
646 // Get the output stream "configuration".
\r
647 AudioBufferList *bufferList = nil;
\r
648 property.mSelector = kAudioDevicePropertyStreamConfiguration;
\r
649 property.mScope = kAudioDevicePropertyScopeOutput;
\r
650 // property.mElement = kAudioObjectPropertyElementWildcard;
\r
652 result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
\r
653 if ( result != noErr || dataSize == 0 ) {
\r
654 errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration info for device (" << device << ").";
\r
655 errorText_ = errorStream_.str();
\r
656 error( RtAudioError::WARNING );
\r
660 // Allocate the AudioBufferList.
\r
661 bufferList = (AudioBufferList *) malloc( dataSize );
\r
662 if ( bufferList == NULL ) {
\r
663 errorText_ = "RtApiCore::getDeviceInfo: memory error allocating output AudioBufferList.";
\r
664 error( RtAudioError::WARNING );
\r
668 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
\r
669 if ( result != noErr || dataSize == 0 ) {
\r
670 free( bufferList );
\r
671 errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration for device (" << device << ").";
\r
672 errorText_ = errorStream_.str();
\r
673 error( RtAudioError::WARNING );
\r
677 // Get output channel information.
\r
678 unsigned int i, nStreams = bufferList->mNumberBuffers;
\r
679 for ( i=0; i<nStreams; i++ )
\r
680 info.outputChannels += bufferList->mBuffers[i].mNumberChannels;
\r
681 free( bufferList );
\r
683 // Get the input stream "configuration".
\r
684 property.mScope = kAudioDevicePropertyScopeInput;
\r
685 result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
\r
686 if ( result != noErr || dataSize == 0 ) {
\r
687 errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration info for device (" << device << ").";
\r
688 errorText_ = errorStream_.str();
\r
689 error( RtAudioError::WARNING );
\r
693 // Allocate the AudioBufferList.
\r
694 bufferList = (AudioBufferList *) malloc( dataSize );
\r
695 if ( bufferList == NULL ) {
\r
696 errorText_ = "RtApiCore::getDeviceInfo: memory error allocating input AudioBufferList.";
\r
697 error( RtAudioError::WARNING );
\r
701 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
\r
702 if (result != noErr || dataSize == 0) {
\r
703 free( bufferList );
\r
704 errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration for device (" << device << ").";
\r
705 errorText_ = errorStream_.str();
\r
706 error( RtAudioError::WARNING );
\r
710 // Get input channel information.
\r
711 nStreams = bufferList->mNumberBuffers;
\r
712 for ( i=0; i<nStreams; i++ )
\r
713 info.inputChannels += bufferList->mBuffers[i].mNumberChannels;
\r
714 free( bufferList );
\r
716 // If device opens for both playback and capture, we determine the channels.
\r
717 if ( info.outputChannels > 0 && info.inputChannels > 0 )
\r
718 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
\r
720 // Probe the device sample rates.
\r
721 bool isInput = false;
\r
722 if ( info.outputChannels == 0 ) isInput = true;
\r
724 // Determine the supported sample rates.
\r
725 property.mSelector = kAudioDevicePropertyAvailableNominalSampleRates;
\r
726 if ( isInput == false ) property.mScope = kAudioDevicePropertyScopeOutput;
\r
727 result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
\r
728 if ( result != kAudioHardwareNoError || dataSize == 0 ) {
\r
729 errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rate info.";
\r
730 errorText_ = errorStream_.str();
\r
731 error( RtAudioError::WARNING );
\r
735 UInt32 nRanges = dataSize / sizeof( AudioValueRange );
\r
736 AudioValueRange rangeList[ nRanges ];
\r
737 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &rangeList );
\r
738 if ( result != kAudioHardwareNoError ) {
\r
739 errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rates.";
\r
740 errorText_ = errorStream_.str();
\r
741 error( RtAudioError::WARNING );
\r
745 // The sample rate reporting mechanism is a bit of a mystery. It
\r
746 // seems that it can either return individual rates or a range of
\r
747 // rates. I assume that if the min / max range values are the same,
\r
748 // then that represents a single supported rate and if the min / max
\r
749 // range values are different, the device supports an arbitrary
\r
750 // range of values (though there might be multiple ranges, so we'll
\r
751 // use the most conservative range).
\r
752 Float64 minimumRate = 1.0, maximumRate = 10000000000.0;
\r
753 bool haveValueRange = false;
\r
754 info.sampleRates.clear();
\r
755 for ( UInt32 i=0; i<nRanges; i++ ) {
\r
756 if ( rangeList[i].mMinimum == rangeList[i].mMaximum )
\r
757 info.sampleRates.push_back( (unsigned int) rangeList[i].mMinimum );
\r
759 haveValueRange = true;
\r
760 if ( rangeList[i].mMinimum > minimumRate ) minimumRate = rangeList[i].mMinimum;
\r
761 if ( rangeList[i].mMaximum < maximumRate ) maximumRate = rangeList[i].mMaximum;
\r
765 if ( haveValueRange ) {
\r
766 for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
\r
767 if ( SAMPLE_RATES[k] >= (unsigned int) minimumRate && SAMPLE_RATES[k] <= (unsigned int) maximumRate )
\r
768 info.sampleRates.push_back( SAMPLE_RATES[k] );
\r
772 // Sort and remove any redundant values
\r
773 std::sort( info.sampleRates.begin(), info.sampleRates.end() );
\r
774 info.sampleRates.erase( unique( info.sampleRates.begin(), info.sampleRates.end() ), info.sampleRates.end() );
\r
776 if ( info.sampleRates.size() == 0 ) {
\r
777 errorStream_ << "RtApiCore::probeDeviceInfo: No supported sample rates found for device (" << device << ").";
\r
778 errorText_ = errorStream_.str();
\r
779 error( RtAudioError::WARNING );
\r
783 // CoreAudio always uses 32-bit floating point data for PCM streams.
\r
784 // Thus, any other "physical" formats supported by the device are of
\r
785 // no interest to the client.
\r
786 info.nativeFormats = RTAUDIO_FLOAT32;
\r
788 if ( info.outputChannels > 0 )
\r
789 if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true;
\r
790 if ( info.inputChannels > 0 )
\r
791 if ( getDefaultInputDevice() == device ) info.isDefaultInput = true;
\r
793 info.probed = true;
\r
797 static OSStatus callbackHandler( AudioDeviceID inDevice,
\r
798 const AudioTimeStamp* /*inNow*/,
\r
799 const AudioBufferList* inInputData,
\r
800 const AudioTimeStamp* /*inInputTime*/,
\r
801 AudioBufferList* outOutputData,
\r
802 const AudioTimeStamp* /*inOutputTime*/,
\r
803 void* infoPointer )
\r
805 CallbackInfo *info = (CallbackInfo *) infoPointer;
\r
807 RtApiCore *object = (RtApiCore *) info->object;
\r
808 if ( object->callbackEvent( inDevice, inInputData, outOutputData ) == false )
\r
809 return kAudioHardwareUnspecifiedError;
\r
811 return kAudioHardwareNoError;
\r
814 static OSStatus xrunListener( AudioObjectID /*inDevice*/,
\r
816 const AudioObjectPropertyAddress properties[],
\r
817 void* handlePointer )
\r
819 CoreHandle *handle = (CoreHandle *) handlePointer;
\r
820 for ( UInt32 i=0; i<nAddresses; i++ ) {
\r
821 if ( properties[i].mSelector == kAudioDeviceProcessorOverload ) {
\r
822 if ( properties[i].mScope == kAudioDevicePropertyScopeInput )
\r
823 handle->xrun[1] = true;
\r
825 handle->xrun[0] = true;
\r
829 return kAudioHardwareNoError;
\r
832 static OSStatus rateListener( AudioObjectID inDevice,
\r
833 UInt32 /*nAddresses*/,
\r
834 const AudioObjectPropertyAddress /*properties*/[],
\r
835 void* ratePointer )
\r
837 Float64 *rate = (Float64 *) ratePointer;
\r
838 UInt32 dataSize = sizeof( Float64 );
\r
839 AudioObjectPropertyAddress property = { kAudioDevicePropertyNominalSampleRate,
\r
840 kAudioObjectPropertyScopeGlobal,
\r
841 kAudioObjectPropertyElementMaster };
\r
842 AudioObjectGetPropertyData( inDevice, &property, 0, NULL, &dataSize, rate );
\r
843 return kAudioHardwareNoError;
\r
846 bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
\r
847 unsigned int firstChannel, unsigned int sampleRate,
\r
848 RtAudioFormat format, unsigned int *bufferSize,
\r
849 RtAudio::StreamOptions *options )
\r
852 unsigned int nDevices = getDeviceCount();
\r
853 if ( nDevices == 0 ) {
\r
854 // This should not happen because a check is made before this function is called.
\r
855 errorText_ = "RtApiCore::probeDeviceOpen: no devices found!";
\r
859 if ( device >= nDevices ) {
\r
860 // This should not happen because a check is made before this function is called.
\r
861 errorText_ = "RtApiCore::probeDeviceOpen: device ID is invalid!";
\r
865 AudioDeviceID deviceList[ nDevices ];
\r
866 UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;
\r
867 AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
\r
868 kAudioObjectPropertyScopeGlobal,
\r
869 kAudioObjectPropertyElementMaster };
\r
870 OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property,
\r
871 0, NULL, &dataSize, (void *) &deviceList );
\r
872 if ( result != noErr ) {
\r
873 errorText_ = "RtApiCore::probeDeviceOpen: OS-X system error getting device IDs.";
\r
877 AudioDeviceID id = deviceList[ device ];
\r
879 // Setup for stream mode.
\r
880 bool isInput = false;
\r
881 if ( mode == INPUT ) {
\r
883 property.mScope = kAudioDevicePropertyScopeInput;
\r
886 property.mScope = kAudioDevicePropertyScopeOutput;
\r
888 // Get the stream "configuration".
\r
889 AudioBufferList *bufferList = nil;
\r
891 property.mSelector = kAudioDevicePropertyStreamConfiguration;
\r
892 result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
\r
893 if ( result != noErr || dataSize == 0 ) {
\r
894 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration info for device (" << device << ").";
\r
895 errorText_ = errorStream_.str();
\r
899 // Allocate the AudioBufferList.
\r
900 bufferList = (AudioBufferList *) malloc( dataSize );
\r
901 if ( bufferList == NULL ) {
\r
902 errorText_ = "RtApiCore::probeDeviceOpen: memory error allocating AudioBufferList.";
\r
906 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
\r
907 if (result != noErr || dataSize == 0) {
\r
908 free( bufferList );
\r
909 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration for device (" << device << ").";
\r
910 errorText_ = errorStream_.str();
\r
914 // Search for one or more streams that contain the desired number of
\r
915 // channels. CoreAudio devices can have an arbitrary number of
\r
916 // streams and each stream can have an arbitrary number of channels.
\r
917 // For each stream, a single buffer of interleaved samples is
\r
918 // provided. RtAudio prefers the use of one stream of interleaved
\r
919 // data or multiple consecutive single-channel streams. However, we
\r
920 // now support multiple consecutive multi-channel streams of
\r
921 // interleaved data as well.
\r
922 UInt32 iStream, offsetCounter = firstChannel;
\r
923 UInt32 nStreams = bufferList->mNumberBuffers;
\r
924 bool monoMode = false;
\r
925 bool foundStream = false;
\r
927 // First check that the device supports the requested number of
\r
929 UInt32 deviceChannels = 0;
\r
930 for ( iStream=0; iStream<nStreams; iStream++ )
\r
931 deviceChannels += bufferList->mBuffers[iStream].mNumberChannels;
\r
933 if ( deviceChannels < ( channels + firstChannel ) ) {
\r
934 free( bufferList );
\r
935 errorStream_ << "RtApiCore::probeDeviceOpen: the device (" << device << ") does not support the requested channel count.";
\r
936 errorText_ = errorStream_.str();
\r
940 // Look for a single stream meeting our needs.
\r
941 UInt32 firstStream, streamCount = 1, streamChannels = 0, channelOffset = 0;
\r
942 for ( iStream=0; iStream<nStreams; iStream++ ) {
\r
943 streamChannels = bufferList->mBuffers[iStream].mNumberChannels;
\r
944 if ( streamChannels >= channels + offsetCounter ) {
\r
945 firstStream = iStream;
\r
946 channelOffset = offsetCounter;
\r
947 foundStream = true;
\r
950 if ( streamChannels > offsetCounter ) break;
\r
951 offsetCounter -= streamChannels;
\r
954 // If we didn't find a single stream above, then we should be able
\r
955 // to meet the channel specification with multiple streams.
\r
956 if ( foundStream == false ) {
\r
958 offsetCounter = firstChannel;
\r
959 for ( iStream=0; iStream<nStreams; iStream++ ) {
\r
960 streamChannels = bufferList->mBuffers[iStream].mNumberChannels;
\r
961 if ( streamChannels > offsetCounter ) break;
\r
962 offsetCounter -= streamChannels;
\r
965 firstStream = iStream;
\r
966 channelOffset = offsetCounter;
\r
967 Int32 channelCounter = channels + offsetCounter - streamChannels;
\r
969 if ( streamChannels > 1 ) monoMode = false;
\r
970 while ( channelCounter > 0 ) {
\r
971 streamChannels = bufferList->mBuffers[++iStream].mNumberChannels;
\r
972 if ( streamChannels > 1 ) monoMode = false;
\r
973 channelCounter -= streamChannels;
\r
978 free( bufferList );
\r
980 // Determine the buffer size.
\r
981 AudioValueRange bufferRange;
\r
982 dataSize = sizeof( AudioValueRange );
\r
983 property.mSelector = kAudioDevicePropertyBufferFrameSizeRange;
\r
984 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &bufferRange );
\r
986 if ( result != noErr ) {
\r
987 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting buffer size range for device (" << device << ").";
\r
988 errorText_ = errorStream_.str();
\r
992 if ( bufferRange.mMinimum > *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMinimum;
\r
993 else if ( bufferRange.mMaximum < *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMaximum;
\r
994 if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) *bufferSize = (unsigned long) bufferRange.mMinimum;
\r
996 // Set the buffer size. For multiple streams, I'm assuming we only
\r
997 // need to make this setting for the master channel.
\r
998 UInt32 theSize = (UInt32) *bufferSize;
\r
999 dataSize = sizeof( UInt32 );
\r
1000 property.mSelector = kAudioDevicePropertyBufferFrameSize;
\r
1001 result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &theSize );
\r
1003 if ( result != noErr ) {
\r
1004 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting the buffer size for device (" << device << ").";
\r
1005 errorText_ = errorStream_.str();
\r
1009 // If attempting to setup a duplex stream, the bufferSize parameter
\r
1010 // MUST be the same in both directions!
\r
1011 *bufferSize = theSize;
\r
1012 if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {
\r
1013 errorStream_ << "RtApiCore::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << device << ").";
\r
1014 errorText_ = errorStream_.str();
\r
1018 stream_.bufferSize = *bufferSize;
\r
1019 stream_.nBuffers = 1;
\r
1021 // Try to set "hog" mode ... it's not clear to me this is working.
\r
1022 if ( options && options->flags & RTAUDIO_HOG_DEVICE ) {
\r
1024 dataSize = sizeof( hog_pid );
\r
1025 property.mSelector = kAudioDevicePropertyHogMode;
\r
1026 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &hog_pid );
\r
1027 if ( result != noErr ) {
\r
1028 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting 'hog' state!";
\r
1029 errorText_ = errorStream_.str();
\r
1033 if ( hog_pid != getpid() ) {
\r
1034 hog_pid = getpid();
\r
1035 result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &hog_pid );
\r
1036 if ( result != noErr ) {
\r
1037 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting 'hog' state!";
\r
1038 errorText_ = errorStream_.str();
\r
1044 // Check and if necessary, change the sample rate for the device.
\r
1045 Float64 nominalRate;
\r
1046 dataSize = sizeof( Float64 );
\r
1047 property.mSelector = kAudioDevicePropertyNominalSampleRate;
\r
1048 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &nominalRate );
\r
1049 if ( result != noErr ) {
\r
1050 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting current sample rate.";
\r
1051 errorText_ = errorStream_.str();
\r
1055 // Only change the sample rate if off by more than 1 Hz.
\r
1056 if ( fabs( nominalRate - (double)sampleRate ) > 1.0 ) {
\r
1058 // Set a property listener for the sample rate change
\r
1059 Float64 reportedRate = 0.0;
\r
1060 AudioObjectPropertyAddress tmp = { kAudioDevicePropertyNominalSampleRate, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
\r
1061 result = AudioObjectAddPropertyListener( id, &tmp, rateListener, (void *) &reportedRate );
\r
1062 if ( result != noErr ) {
\r
1063 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate property listener for device (" << device << ").";
\r
1064 errorText_ = errorStream_.str();
\r
1068 nominalRate = (Float64) sampleRate;
\r
1069 result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &nominalRate );
\r
1070 if ( result != noErr ) {
\r
1071 AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate );
\r
1072 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate for device (" << device << ").";
\r
1073 errorText_ = errorStream_.str();
\r
1077 // Now wait until the reported nominal rate is what we just set.
\r
1078 UInt32 microCounter = 0;
\r
1079 while ( reportedRate != nominalRate ) {
\r
1080 microCounter += 5000;
\r
1081 if ( microCounter > 5000000 ) break;
\r
1085 // Remove the property listener.
\r
1086 AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate );
\r
1088 if ( microCounter > 5000000 ) {
\r
1089 errorStream_ << "RtApiCore::probeDeviceOpen: timeout waiting for sample rate update for device (" << device << ").";
\r
1090 errorText_ = errorStream_.str();
\r
1095 // Now set the stream format for all streams. Also, check the
\r
1096 // physical format of the device and change that if necessary.
\r
1097 AudioStreamBasicDescription description;
\r
1098 dataSize = sizeof( AudioStreamBasicDescription );
\r
1099 property.mSelector = kAudioStreamPropertyVirtualFormat;
\r
1100 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &description );
\r
1101 if ( result != noErr ) {
\r
1102 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream format for device (" << device << ").";
\r
1103 errorText_ = errorStream_.str();
\r
1107 // Set the sample rate and data format id. However, only make the
\r
1108 // change if the sample rate is not within 1.0 of the desired
\r
1109 // rate and the format is not linear pcm.
\r
1110 bool updateFormat = false;
\r
1111 if ( fabs( description.mSampleRate - (Float64)sampleRate ) > 1.0 ) {
\r
1112 description.mSampleRate = (Float64) sampleRate;
\r
1113 updateFormat = true;
\r
1116 if ( description.mFormatID != kAudioFormatLinearPCM ) {
\r
1117 description.mFormatID = kAudioFormatLinearPCM;
\r
1118 updateFormat = true;
\r
1121 if ( updateFormat ) {
\r
1122 result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &description );
\r
1123 if ( result != noErr ) {
\r
1124 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate or data format for device (" << device << ").";
\r
1125 errorText_ = errorStream_.str();
\r
1130 // Now check the physical format.
\r
1131 property.mSelector = kAudioStreamPropertyPhysicalFormat;
\r
1132 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &description );
\r
1133 if ( result != noErr ) {
\r
1134 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream physical format for device (" << device << ").";
\r
1135 errorText_ = errorStream_.str();
\r
1139 //std::cout << "Current physical stream format:" << std::endl;
\r
1140 //std::cout << " mBitsPerChan = " << description.mBitsPerChannel << std::endl;
\r
1141 //std::cout << " aligned high = " << (description.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (description.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl;
\r
1142 //std::cout << " bytesPerFrame = " << description.mBytesPerFrame << std::endl;
\r
1143 //std::cout << " sample rate = " << description.mSampleRate << std::endl;
\r
1145 if ( description.mFormatID != kAudioFormatLinearPCM || description.mBitsPerChannel < 16 ) {
\r
1146 description.mFormatID = kAudioFormatLinearPCM;
\r
1147 //description.mSampleRate = (Float64) sampleRate;
\r
1148 AudioStreamBasicDescription testDescription = description;
\r
1149 UInt32 formatFlags;
\r
1151 // We'll try higher bit rates first and then work our way down.
\r
1152 std::vector< std::pair<UInt32, UInt32> > physicalFormats;
\r
1153 formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsFloat) & ~kLinearPCMFormatFlagIsSignedInteger;
\r
1154 physicalFormats.push_back( std::pair<Float32, UInt32>( 32, formatFlags ) );
\r
1155 formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat;
\r
1156 physicalFormats.push_back( std::pair<Float32, UInt32>( 32, formatFlags ) );
\r
1157 physicalFormats.push_back( std::pair<Float32, UInt32>( 24, formatFlags ) ); // 24-bit packed
\r
1158 formatFlags &= ~( kAudioFormatFlagIsPacked | kAudioFormatFlagIsAlignedHigh );
\r
1159 physicalFormats.push_back( std::pair<Float32, UInt32>( 24.2, formatFlags ) ); // 24-bit in 4 bytes, aligned low
\r
1160 formatFlags |= kAudioFormatFlagIsAlignedHigh;
\r
1161 physicalFormats.push_back( std::pair<Float32, UInt32>( 24.4, formatFlags ) ); // 24-bit in 4 bytes, aligned high
\r
1162 formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat;
\r
1163 physicalFormats.push_back( std::pair<Float32, UInt32>( 16, formatFlags ) );
\r
1164 physicalFormats.push_back( std::pair<Float32, UInt32>( 8, formatFlags ) );
\r
1166 bool setPhysicalFormat = false;
\r
1167 for( unsigned int i=0; i<physicalFormats.size(); i++ ) {
\r
1168 testDescription = description;
\r
1169 testDescription.mBitsPerChannel = (UInt32) physicalFormats[i].first;
\r
1170 testDescription.mFormatFlags = physicalFormats[i].second;
\r
1171 if ( (24 == (UInt32)physicalFormats[i].first) && ~( physicalFormats[i].second & kAudioFormatFlagIsPacked ) )
\r
1172 testDescription.mBytesPerFrame = 4 * testDescription.mChannelsPerFrame;
\r
1174 testDescription.mBytesPerFrame = testDescription.mBitsPerChannel/8 * testDescription.mChannelsPerFrame;
\r
1175 testDescription.mBytesPerPacket = testDescription.mBytesPerFrame * testDescription.mFramesPerPacket;
\r
1176 result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &testDescription );
\r
1177 if ( result == noErr ) {
\r
1178 setPhysicalFormat = true;
\r
1179 //std::cout << "Updated physical stream format:" << std::endl;
\r
1180 //std::cout << " mBitsPerChan = " << testDescription.mBitsPerChannel << std::endl;
\r
1181 //std::cout << " aligned high = " << (testDescription.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (testDescription.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl;
\r
1182 //std::cout << " bytesPerFrame = " << testDescription.mBytesPerFrame << std::endl;
\r
1183 //std::cout << " sample rate = " << testDescription.mSampleRate << std::endl;
\r
1188 if ( !setPhysicalFormat ) {
\r
1189 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting physical data format for device (" << device << ").";
\r
1190 errorText_ = errorStream_.str();
\r
1193 } // done setting virtual/physical formats.
\r
1195 // Get the stream / device latency.
\r
1197 dataSize = sizeof( UInt32 );
\r
1198 property.mSelector = kAudioDevicePropertyLatency;
\r
1199 if ( AudioObjectHasProperty( id, &property ) == true ) {
\r
1200 result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &latency );
\r
1201 if ( result == kAudioHardwareNoError ) stream_.latency[ mode ] = latency;
\r
1203 errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting device latency for device (" << device << ").";
\r
1204 errorText_ = errorStream_.str();
\r
1205 error( RtAudioError::WARNING );
\r
1209 // Byte-swapping: According to AudioHardware.h, the stream data will
\r
1210 // always be presented in native-endian format, so we should never
\r
1211 // need to byte swap.
\r
1212 stream_.doByteSwap[mode] = false;
\r
1214 // From the CoreAudio documentation, PCM data must be supplied as
\r
1216 stream_.userFormat = format;
\r
1217 stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
\r
1219 if ( streamCount == 1 )
\r
1220 stream_.nDeviceChannels[mode] = description.mChannelsPerFrame;
\r
1221 else // multiple streams
\r
1222 stream_.nDeviceChannels[mode] = channels;
\r
1223 stream_.nUserChannels[mode] = channels;
\r
1224 stream_.channelOffset[mode] = channelOffset; // offset within a CoreAudio stream
\r
1225 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
\r
1226 else stream_.userInterleaved = true;
\r
1227 stream_.deviceInterleaved[mode] = true;
\r
1228 if ( monoMode == true ) stream_.deviceInterleaved[mode] = false;
\r
1230 // Set flags for buffer conversion.
\r
1231 stream_.doConvertBuffer[mode] = false;
\r
1232 if ( stream_.userFormat != stream_.deviceFormat[mode] )
\r
1233 stream_.doConvertBuffer[mode] = true;
\r
1234 if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
\r
1235 stream_.doConvertBuffer[mode] = true;
\r
1236 if ( streamCount == 1 ) {
\r
1237 if ( stream_.nUserChannels[mode] > 1 &&
\r
1238 stream_.userInterleaved != stream_.deviceInterleaved[mode] )
\r
1239 stream_.doConvertBuffer[mode] = true;
\r
1241 else if ( monoMode && stream_.userInterleaved )
\r
1242 stream_.doConvertBuffer[mode] = true;
\r
1244 // Allocate our CoreHandle structure for the stream.
\r
1245 CoreHandle *handle = 0;
\r
1246 if ( stream_.apiHandle == 0 ) {
\r
1248 handle = new CoreHandle;
\r
1250 catch ( std::bad_alloc& ) {
\r
1251 errorText_ = "RtApiCore::probeDeviceOpen: error allocating CoreHandle memory.";
\r
1255 if ( pthread_cond_init( &handle->condition, NULL ) ) {
\r
1256 errorText_ = "RtApiCore::probeDeviceOpen: error initializing pthread condition variable.";
\r
1259 stream_.apiHandle = (void *) handle;
\r
1262 handle = (CoreHandle *) stream_.apiHandle;
\r
1263 handle->iStream[mode] = firstStream;
\r
1264 handle->nStreams[mode] = streamCount;
\r
1265 handle->id[mode] = id;
\r
1267 // Allocate necessary internal buffers.
\r
1268 unsigned long bufferBytes;
\r
1269 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
\r
1270 // stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
\r
1271 stream_.userBuffer[mode] = (char *) malloc( bufferBytes * sizeof(char) );
\r
1272 memset( stream_.userBuffer[mode], 0, bufferBytes * sizeof(char) );
\r
1273 if ( stream_.userBuffer[mode] == NULL ) {
\r
1274 errorText_ = "RtApiCore::probeDeviceOpen: error allocating user buffer memory.";
\r
1278 // If possible, we will make use of the CoreAudio stream buffers as
\r
1279 // "device buffers". However, we can't do this if using multiple
\r
1281 if ( stream_.doConvertBuffer[mode] && handle->nStreams[mode] > 1 ) {
\r
1283 bool makeBuffer = true;
\r
1284 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
\r
1285 if ( mode == INPUT ) {
\r
1286 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
\r
1287 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
\r
1288 if ( bufferBytes <= bytesOut ) makeBuffer = false;
\r
1292 if ( makeBuffer ) {
\r
1293 bufferBytes *= *bufferSize;
\r
1294 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
\r
1295 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
\r
1296 if ( stream_.deviceBuffer == NULL ) {
\r
1297 errorText_ = "RtApiCore::probeDeviceOpen: error allocating device buffer memory.";
\r
1303 stream_.sampleRate = sampleRate;
\r
1304 stream_.device[mode] = device;
\r
1305 stream_.state = STREAM_STOPPED;
\r
1306 stream_.callbackInfo.object = (void *) this;
\r
1308 // Setup the buffer conversion information structure.
\r
1309 if ( stream_.doConvertBuffer[mode] ) {
\r
1310 if ( streamCount > 1 ) setConvertInfo( mode, 0 );
\r
1311 else setConvertInfo( mode, channelOffset );
\r
1314 if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device )
\r
1315 // Only one callback procedure per device.
\r
1316 stream_.mode = DUPLEX;
\r
1318 #if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
\r
1319 result = AudioDeviceCreateIOProcID( id, callbackHandler, (void *) &stream_.callbackInfo, &handle->procId[mode] );
\r
1321 // deprecated in favor of AudioDeviceCreateIOProcID()
\r
1322 result = AudioDeviceAddIOProc( id, callbackHandler, (void *) &stream_.callbackInfo );
\r
1324 if ( result != noErr ) {
\r
1325 errorStream_ << "RtApiCore::probeDeviceOpen: system error setting callback for device (" << device << ").";
\r
1326 errorText_ = errorStream_.str();
\r
1329 if ( stream_.mode == OUTPUT && mode == INPUT )
\r
1330 stream_.mode = DUPLEX;
\r
1332 stream_.mode = mode;
\r
1335 // Setup the device property listener for over/underload.
\r
1336 property.mSelector = kAudioDeviceProcessorOverload;
\r
1337 property.mScope = kAudioObjectPropertyScopeGlobal;
\r
1338 result = AudioObjectAddPropertyListener( id, &property, xrunListener, (void *) handle );
\r
1344 pthread_cond_destroy( &handle->condition );
\r
1346 stream_.apiHandle = 0;
\r
1349 for ( int i=0; i<2; i++ ) {
\r
1350 if ( stream_.userBuffer[i] ) {
\r
1351 free( stream_.userBuffer[i] );
\r
1352 stream_.userBuffer[i] = 0;
\r
1356 if ( stream_.deviceBuffer ) {
\r
1357 free( stream_.deviceBuffer );
\r
1358 stream_.deviceBuffer = 0;
\r
1361 stream_.state = STREAM_CLOSED;
\r
1365 void RtApiCore :: closeStream( void )
\r
1367 if ( stream_.state == STREAM_CLOSED ) {
\r
1368 errorText_ = "RtApiCore::closeStream(): no open stream to close!";
\r
1369 error( RtAudioError::WARNING );
\r
1373 CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
\r
1374 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
\r
1375 if ( stream_.state == STREAM_RUNNING )
\r
1376 AudioDeviceStop( handle->id[0], callbackHandler );
\r
1377 #if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
\r
1378 AudioDeviceDestroyIOProcID( handle->id[0], handle->procId[0] );
\r
1380 // deprecated in favor of AudioDeviceDestroyIOProcID()
\r
1381 AudioDeviceRemoveIOProc( handle->id[0], callbackHandler );
\r
1385 if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
\r
1386 if ( stream_.state == STREAM_RUNNING )
\r
1387 AudioDeviceStop( handle->id[1], callbackHandler );
\r
1388 #if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
\r
1389 AudioDeviceDestroyIOProcID( handle->id[1], handle->procId[1] );
\r
1391 // deprecated in favor of AudioDeviceDestroyIOProcID()
\r
1392 AudioDeviceRemoveIOProc( handle->id[1], callbackHandler );
\r
1396 for ( int i=0; i<2; i++ ) {
\r
1397 if ( stream_.userBuffer[i] ) {
\r
1398 free( stream_.userBuffer[i] );
\r
1399 stream_.userBuffer[i] = 0;
\r
1403 if ( stream_.deviceBuffer ) {
\r
1404 free( stream_.deviceBuffer );
\r
1405 stream_.deviceBuffer = 0;
\r
1408 // Destroy pthread condition variable.
\r
1409 pthread_cond_destroy( &handle->condition );
\r
1411 stream_.apiHandle = 0;
\r
1413 stream_.mode = UNINITIALIZED;
\r
1414 stream_.state = STREAM_CLOSED;
\r
1417 void RtApiCore :: startStream( void )
\r
1420 if ( stream_.state == STREAM_RUNNING ) {
\r
1421 errorText_ = "RtApiCore::startStream(): the stream is already running!";
\r
1422 error( RtAudioError::WARNING );
\r
1426 OSStatus result = noErr;
\r
1427 CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
\r
1428 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
\r
1430 result = AudioDeviceStart( handle->id[0], callbackHandler );
\r
1431 if ( result != noErr ) {
\r
1432 errorStream_ << "RtApiCore::startStream: system error (" << getErrorCode( result ) << ") starting callback procedure on device (" << stream_.device[0] << ").";
\r
1433 errorText_ = errorStream_.str();
\r
1438 if ( stream_.mode == INPUT ||
\r
1439 ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
\r
1441 result = AudioDeviceStart( handle->id[1], callbackHandler );
\r
1442 if ( result != noErr ) {
\r
1443 errorStream_ << "RtApiCore::startStream: system error starting input callback procedure on device (" << stream_.device[1] << ").";
\r
1444 errorText_ = errorStream_.str();
\r
1449 handle->drainCounter = 0;
\r
1450 handle->internalDrain = false;
\r
1451 stream_.state = STREAM_RUNNING;
\r
1454 if ( result == noErr ) return;
\r
1455 error( RtAudioError::SYSTEM_ERROR );
\r
1458 void RtApiCore :: stopStream( void )
\r
1461 if ( stream_.state == STREAM_STOPPED ) {
\r
1462 errorText_ = "RtApiCore::stopStream(): the stream is already stopped!";
\r
1463 error( RtAudioError::WARNING );
\r
1467 OSStatus result = noErr;
\r
1468 CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
\r
1469 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
\r
1471 if ( handle->drainCounter == 0 ) {
\r
1472 handle->drainCounter = 2;
\r
1473 pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
\r
1476 result = AudioDeviceStop( handle->id[0], callbackHandler );
\r
1477 if ( result != noErr ) {
\r
1478 errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping callback procedure on device (" << stream_.device[0] << ").";
\r
1479 errorText_ = errorStream_.str();
\r
1484 if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
\r
1486 result = AudioDeviceStop( handle->id[1], callbackHandler );
\r
1487 if ( result != noErr ) {
\r
1488 errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping input callback procedure on device (" << stream_.device[1] << ").";
\r
1489 errorText_ = errorStream_.str();
\r
1494 stream_.state = STREAM_STOPPED;
\r
1497 if ( result == noErr ) return;
\r
1498 error( RtAudioError::SYSTEM_ERROR );
\r
1501 void RtApiCore :: abortStream( void )
\r
1504 if ( stream_.state == STREAM_STOPPED ) {
\r
1505 errorText_ = "RtApiCore::abortStream(): the stream is already stopped!";
\r
1506 error( RtAudioError::WARNING );
\r
1510 CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
\r
1511 handle->drainCounter = 2;
\r
1516 // This function will be called by a spawned thread when the user
\r
1517 // callback function signals that the stream should be stopped or
\r
1518 // aborted. It is better to handle it this way because the
\r
1519 // callbackEvent() function probably should return before the AudioDeviceStop()
\r
1520 // function is called.
\r
1521 static void *coreStopStream( void *ptr )
\r
1523 CallbackInfo *info = (CallbackInfo *) ptr;
\r
1524 RtApiCore *object = (RtApiCore *) info->object;
\r
1526 object->stopStream();
\r
1527 pthread_exit( NULL );
\r
1530 bool RtApiCore :: callbackEvent( AudioDeviceID deviceId,
\r
1531 const AudioBufferList *inBufferList,
\r
1532 const AudioBufferList *outBufferList )
\r
1534 if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;
\r
1535 if ( stream_.state == STREAM_CLOSED ) {
\r
1536 errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";
\r
1537 error( RtAudioError::WARNING );
\r
1541 CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
\r
1542 CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
\r
1544 // Check if we were draining the stream and signal is finished.
\r
1545 if ( handle->drainCounter > 3 ) {
\r
1546 ThreadHandle threadId;
\r
1548 stream_.state = STREAM_STOPPING;
\r
1549 if ( handle->internalDrain == true )
\r
1550 pthread_create( &threadId, NULL, coreStopStream, info );
\r
1551 else // external call to stopStream()
\r
1552 pthread_cond_signal( &handle->condition );
\r
1556 AudioDeviceID outputDevice = handle->id[0];
\r
1558 // Invoke user callback to get fresh output data UNLESS we are
\r
1559 // draining stream or duplex mode AND the input/output devices are
\r
1560 // different AND this function is called for the input device.
\r
1561 if ( handle->drainCounter == 0 && ( stream_.mode != DUPLEX || deviceId == outputDevice ) ) {
\r
1562 RtAudioCallback callback = (RtAudioCallback) info->callback;
\r
1563 double streamTime = getStreamTime();
\r
1564 RtAudioStreamStatus status = 0;
\r
1565 if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
\r
1566 status |= RTAUDIO_OUTPUT_UNDERFLOW;
\r
1567 handle->xrun[0] = false;
\r
1569 if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
\r
1570 status |= RTAUDIO_INPUT_OVERFLOW;
\r
1571 handle->xrun[1] = false;
\r
1574 int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
\r
1575 stream_.bufferSize, streamTime, status, info->userData );
\r
1576 if ( cbReturnValue == 2 ) {
\r
1577 stream_.state = STREAM_STOPPING;
\r
1578 handle->drainCounter = 2;
\r
1582 else if ( cbReturnValue == 1 ) {
\r
1583 handle->drainCounter = 1;
\r
1584 handle->internalDrain = true;
\r
1588 if ( stream_.mode == OUTPUT || ( stream_.mode == DUPLEX && deviceId == outputDevice ) ) {
\r
1590 if ( handle->drainCounter > 1 ) { // write zeros to the output stream
\r
1592 if ( handle->nStreams[0] == 1 ) {
\r
1593 memset( outBufferList->mBuffers[handle->iStream[0]].mData,
\r
1595 outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );
\r
1597 else { // fill multiple streams with zeros
\r
1598 for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) {
\r
1599 memset( outBufferList->mBuffers[handle->iStream[0]+i].mData,
\r
1601 outBufferList->mBuffers[handle->iStream[0]+i].mDataByteSize );
\r
1605 else if ( handle->nStreams[0] == 1 ) {
\r
1606 if ( stream_.doConvertBuffer[0] ) { // convert directly to CoreAudio stream buffer
\r
1607 convertBuffer( (char *) outBufferList->mBuffers[handle->iStream[0]].mData,
\r
1608 stream_.userBuffer[0], stream_.convertInfo[0] );
\r
1610 else { // copy from user buffer
\r
1611 memcpy( outBufferList->mBuffers[handle->iStream[0]].mData,
\r
1612 stream_.userBuffer[0],
\r
1613 outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );
\r
1616 else { // fill multiple streams
\r
1617 Float32 *inBuffer = (Float32 *) stream_.userBuffer[0];
\r
1618 if ( stream_.doConvertBuffer[0] ) {
\r
1619 convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
\r
1620 inBuffer = (Float32 *) stream_.deviceBuffer;
\r
1623 if ( stream_.deviceInterleaved[0] == false ) { // mono mode
\r
1624 UInt32 bufferBytes = outBufferList->mBuffers[handle->iStream[0]].mDataByteSize;
\r
1625 for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
\r
1626 memcpy( outBufferList->mBuffers[handle->iStream[0]+i].mData,
\r
1627 (void *)&inBuffer[i*stream_.bufferSize], bufferBytes );
\r
1630 else { // fill multiple multi-channel streams with interleaved data
\r
1631 UInt32 streamChannels, channelsLeft, inJump, outJump, inOffset;
\r
1632 Float32 *out, *in;
\r
1634 bool inInterleaved = ( stream_.userInterleaved ) ? true : false;
\r
1635 UInt32 inChannels = stream_.nUserChannels[0];
\r
1636 if ( stream_.doConvertBuffer[0] ) {
\r
1637 inInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode
\r
1638 inChannels = stream_.nDeviceChannels[0];
\r
1641 if ( inInterleaved ) inOffset = 1;
\r
1642 else inOffset = stream_.bufferSize;
\r
1644 channelsLeft = inChannels;
\r
1645 for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) {
\r
1647 out = (Float32 *) outBufferList->mBuffers[handle->iStream[0]+i].mData;
\r
1648 streamChannels = outBufferList->mBuffers[handle->iStream[0]+i].mNumberChannels;
\r
1651 // Account for possible channel offset in first stream
\r
1652 if ( i == 0 && stream_.channelOffset[0] > 0 ) {
\r
1653 streamChannels -= stream_.channelOffset[0];
\r
1654 outJump = stream_.channelOffset[0];
\r
1658 // Account for possible unfilled channels at end of the last stream
\r
1659 if ( streamChannels > channelsLeft ) {
\r
1660 outJump = streamChannels - channelsLeft;
\r
1661 streamChannels = channelsLeft;
\r
1664 // Determine input buffer offsets and skips
\r
1665 if ( inInterleaved ) {
\r
1666 inJump = inChannels;
\r
1667 in += inChannels - channelsLeft;
\r
1671 in += (inChannels - channelsLeft) * inOffset;
\r
1674 for ( unsigned int i=0; i<stream_.bufferSize; i++ ) {
\r
1675 for ( unsigned int j=0; j<streamChannels; j++ ) {
\r
1676 *out++ = in[j*inOffset];
\r
1681 channelsLeft -= streamChannels;
\r
1686 if ( handle->drainCounter ) {
\r
1687 handle->drainCounter++;
\r
1692 AudioDeviceID inputDevice;
\r
1693 inputDevice = handle->id[1];
\r
1694 if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && deviceId == inputDevice ) ) {
\r
1696 if ( handle->nStreams[1] == 1 ) {
\r
1697 if ( stream_.doConvertBuffer[1] ) { // convert directly from CoreAudio stream buffer
\r
1698 convertBuffer( stream_.userBuffer[1],
\r
1699 (char *) inBufferList->mBuffers[handle->iStream[1]].mData,
\r
1700 stream_.convertInfo[1] );
\r
1702 else { // copy to user buffer
\r
1703 memcpy( stream_.userBuffer[1],
\r
1704 inBufferList->mBuffers[handle->iStream[1]].mData,
\r
1705 inBufferList->mBuffers[handle->iStream[1]].mDataByteSize );
\r
1708 else { // read from multiple streams
\r
1709 Float32 *outBuffer = (Float32 *) stream_.userBuffer[1];
\r
1710 if ( stream_.doConvertBuffer[1] ) outBuffer = (Float32 *) stream_.deviceBuffer;
\r
1712 if ( stream_.deviceInterleaved[1] == false ) { // mono mode
\r
1713 UInt32 bufferBytes = inBufferList->mBuffers[handle->iStream[1]].mDataByteSize;
\r
1714 for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
\r
1715 memcpy( (void *)&outBuffer[i*stream_.bufferSize],
\r
1716 inBufferList->mBuffers[handle->iStream[1]+i].mData, bufferBytes );
\r
1719 else { // read from multiple multi-channel streams
\r
1720 UInt32 streamChannels, channelsLeft, inJump, outJump, outOffset;
\r
1721 Float32 *out, *in;
\r
1723 bool outInterleaved = ( stream_.userInterleaved ) ? true : false;
\r
1724 UInt32 outChannels = stream_.nUserChannels[1];
\r
1725 if ( stream_.doConvertBuffer[1] ) {
\r
1726 outInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode
\r
1727 outChannels = stream_.nDeviceChannels[1];
\r
1730 if ( outInterleaved ) outOffset = 1;
\r
1731 else outOffset = stream_.bufferSize;
\r
1733 channelsLeft = outChannels;
\r
1734 for ( unsigned int i=0; i<handle->nStreams[1]; i++ ) {
\r
1736 in = (Float32 *) inBufferList->mBuffers[handle->iStream[1]+i].mData;
\r
1737 streamChannels = inBufferList->mBuffers[handle->iStream[1]+i].mNumberChannels;
\r
1740 // Account for possible channel offset in first stream
\r
1741 if ( i == 0 && stream_.channelOffset[1] > 0 ) {
\r
1742 streamChannels -= stream_.channelOffset[1];
\r
1743 inJump = stream_.channelOffset[1];
\r
1747 // Account for possible unread channels at end of the last stream
\r
1748 if ( streamChannels > channelsLeft ) {
\r
1749 inJump = streamChannels - channelsLeft;
\r
1750 streamChannels = channelsLeft;
\r
1753 // Determine output buffer offsets and skips
\r
1754 if ( outInterleaved ) {
\r
1755 outJump = outChannels;
\r
1756 out += outChannels - channelsLeft;
\r
1760 out += (outChannels - channelsLeft) * outOffset;
\r
1763 for ( unsigned int i=0; i<stream_.bufferSize; i++ ) {
\r
1764 for ( unsigned int j=0; j<streamChannels; j++ ) {
\r
1765 out[j*outOffset] = *in++;
\r
1770 channelsLeft -= streamChannels;
\r
1774 if ( stream_.doConvertBuffer[1] ) { // convert from our internal "device" buffer
\r
1775 convertBuffer( stream_.userBuffer[1],
\r
1776 stream_.deviceBuffer,
\r
1777 stream_.convertInfo[1] );
\r
1783 //MUTEX_UNLOCK( &stream_.mutex );
\r
1785 RtApi::tickStreamTime();
\r
1789 const char* RtApiCore :: getErrorCode( OSStatus code )
\r
1793 case kAudioHardwareNotRunningError:
\r
1794 return "kAudioHardwareNotRunningError";
\r
1796 case kAudioHardwareUnspecifiedError:
\r
1797 return "kAudioHardwareUnspecifiedError";
\r
1799 case kAudioHardwareUnknownPropertyError:
\r
1800 return "kAudioHardwareUnknownPropertyError";
\r
1802 case kAudioHardwareBadPropertySizeError:
\r
1803 return "kAudioHardwareBadPropertySizeError";
\r
1805 case kAudioHardwareIllegalOperationError:
\r
1806 return "kAudioHardwareIllegalOperationError";
\r
1808 case kAudioHardwareBadObjectError:
\r
1809 return "kAudioHardwareBadObjectError";
\r
1811 case kAudioHardwareBadDeviceError:
\r
1812 return "kAudioHardwareBadDeviceError";
\r
1814 case kAudioHardwareBadStreamError:
\r
1815 return "kAudioHardwareBadStreamError";
\r
1817 case kAudioHardwareUnsupportedOperationError:
\r
1818 return "kAudioHardwareUnsupportedOperationError";
\r
1820 case kAudioDeviceUnsupportedFormatError:
\r
1821 return "kAudioDeviceUnsupportedFormatError";
\r
1823 case kAudioDevicePermissionsError:
\r
1824 return "kAudioDevicePermissionsError";
\r
1827 return "CoreAudio unknown error";
\r
1831 //******************** End of __MACOSX_CORE__ *********************//
\r
1834 #if defined(__UNIX_JACK__)
\r
1836 // JACK is a low-latency audio server, originally written for the
\r
1837 // GNU/Linux operating system and now also ported to OS-X. It can
\r
1838 // connect a number of different applications to an audio device, as
\r
1839 // well as allowing them to share audio between themselves.
\r
1841 // When using JACK with RtAudio, "devices" refer to JACK clients that
\r
1842 // have ports connected to the server. The JACK server is typically
\r
1843 // started in a terminal as follows:
\r
1845 // .jackd -d alsa -d hw:0
\r
1847 // or through an interface program such as qjackctl. Many of the
\r
1848 // parameters normally set for a stream are fixed by the JACK server
\r
1849 // and can be specified when the JACK server is started. In
\r
1852 // .jackd -d alsa -d hw:0 -r 44100 -p 512 -n 4
\r
1854 // specifies a sample rate of 44100 Hz, a buffer size of 512 sample
\r
1855 // frames, and number of buffers = 4. Once the server is running, it
\r
1856 // is not possible to override these values. If the values are not
\r
1857 // specified in the command-line, the JACK server uses default values.
\r
1859 // The JACK server does not have to be running when an instance of
\r
1860 // RtApiJack is created, though the function getDeviceCount() will
\r
1861 // report 0 devices found until JACK has been started. When no
\r
1862 // devices are available (i.e., the JACK server is not running), a
\r
1863 // stream cannot be opened.
\r
1865 #include <jack/jack.h>
\r
1866 #include <unistd.h>
\r
1869 // A structure to hold various information related to the Jack API
\r
1870 // implementation.
\r
1871 struct JackHandle {
\r
1872 jack_client_t *client;
\r
1873 jack_port_t **ports[2];
\r
1874 std::string deviceName[2];
\r
1876 pthread_cond_t condition;
\r
1877 int drainCounter; // Tracks callback counts when draining
\r
1878 bool internalDrain; // Indicates if stop is initiated from callback or not.
\r
1881 :client(0), drainCounter(0), internalDrain(false) { ports[0] = 0; ports[1] = 0; xrun[0] = false; xrun[1] = false; }
\r
1884 static void jackSilentError( const char * ) {};
\r
1886 RtApiJack :: RtApiJack()
\r
1888 // Nothing to do here.
\r
1889 #if !defined(__RTAUDIO_DEBUG__)
\r
1890 // Turn off Jack's internal error reporting.
\r
1891 jack_set_error_function( &jackSilentError );
\r
1895 RtApiJack :: ~RtApiJack()
\r
1897 if ( stream_.state != STREAM_CLOSED ) closeStream();
\r
1900 unsigned int RtApiJack :: getDeviceCount( void )
\r
1902 // See if we can become a jack client.
\r
1903 jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption;
\r
1904 jack_status_t *status = NULL;
\r
1905 jack_client_t *client = jack_client_open( "RtApiJackCount", options, status );
\r
1906 if ( client == 0 ) return 0;
\r
1908 const char **ports;
\r
1909 std::string port, previousPort;
\r
1910 unsigned int nChannels = 0, nDevices = 0;
\r
1911 ports = jack_get_ports( client, NULL, NULL, 0 );
\r
1913 // Parse the port names up to the first colon (:).
\r
1914 size_t iColon = 0;
\r
1916 port = (char *) ports[ nChannels ];
\r
1917 iColon = port.find(":");
\r
1918 if ( iColon != std::string::npos ) {
\r
1919 port = port.substr( 0, iColon + 1 );
\r
1920 if ( port != previousPort ) {
\r
1922 previousPort = port;
\r
1925 } while ( ports[++nChannels] );
\r
1929 jack_client_close( client );
\r
1933 RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device )
\r
1935 RtAudio::DeviceInfo info;
\r
1936 info.probed = false;
\r
1938 jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption
\r
1939 jack_status_t *status = NULL;
\r
1940 jack_client_t *client = jack_client_open( "RtApiJackInfo", options, status );
\r
1941 if ( client == 0 ) {
\r
1942 errorText_ = "RtApiJack::getDeviceInfo: Jack server not found or connection error!";
\r
1943 error( RtAudioError::WARNING );
\r
1947 const char **ports;
\r
1948 std::string port, previousPort;
\r
1949 unsigned int nPorts = 0, nDevices = 0;
\r
1950 ports = jack_get_ports( client, NULL, NULL, 0 );
\r
1952 // Parse the port names up to the first colon (:).
\r
1953 size_t iColon = 0;
\r
1955 port = (char *) ports[ nPorts ];
\r
1956 iColon = port.find(":");
\r
1957 if ( iColon != std::string::npos ) {
\r
1958 port = port.substr( 0, iColon );
\r
1959 if ( port != previousPort ) {
\r
1960 if ( nDevices == device ) info.name = port;
\r
1962 previousPort = port;
\r
1965 } while ( ports[++nPorts] );
\r
1969 if ( device >= nDevices ) {
\r
1970 jack_client_close( client );
\r
1971 errorText_ = "RtApiJack::getDeviceInfo: device ID is invalid!";
\r
1972 error( RtAudioError::INVALID_USE );
\r
1976 // Get the current jack server sample rate.
\r
1977 info.sampleRates.clear();
\r
1978 info.sampleRates.push_back( jack_get_sample_rate( client ) );
\r
1980 // Count the available ports containing the client name as device
\r
1981 // channels. Jack "input ports" equal RtAudio output channels.
\r
1982 unsigned int nChannels = 0;
\r
1983 ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsInput );
\r
1985 while ( ports[ nChannels ] ) nChannels++;
\r
1987 info.outputChannels = nChannels;
\r
1990 // Jack "output ports" equal RtAudio input channels.
\r
1992 ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsOutput );
\r
1994 while ( ports[ nChannels ] ) nChannels++;
\r
1996 info.inputChannels = nChannels;
\r
1999 if ( info.outputChannels == 0 && info.inputChannels == 0 ) {
\r
2000 jack_client_close(client);
\r
2001 errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!";
\r
2002 error( RtAudioError::WARNING );
\r
2006 // If device opens for both playback and capture, we determine the channels.
\r
2007 if ( info.outputChannels > 0 && info.inputChannels > 0 )
\r
2008 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
\r
2010 // Jack always uses 32-bit floats.
\r
2011 info.nativeFormats = RTAUDIO_FLOAT32;
\r
2013 // Jack doesn't provide default devices so we'll use the first available one.
\r
2014 if ( device == 0 && info.outputChannels > 0 )
\r
2015 info.isDefaultOutput = true;
\r
2016 if ( device == 0 && info.inputChannels > 0 )
\r
2017 info.isDefaultInput = true;
\r
2019 jack_client_close(client);
\r
2020 info.probed = true;
\r
2024 static int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer )
\r
2026 CallbackInfo *info = (CallbackInfo *) infoPointer;
\r
2028 RtApiJack *object = (RtApiJack *) info->object;
\r
2029 if ( object->callbackEvent( (unsigned long) nframes ) == false ) return 1;
\r
2034 // This function will be called by a spawned thread when the Jack
\r
2035 // server signals that it is shutting down. It is necessary to handle
\r
2036 // it this way because the jackShutdown() function must return before
\r
2037 // the jack_deactivate() function (in closeStream()) will return.
\r
2038 static void *jackCloseStream( void *ptr )
\r
2040 CallbackInfo *info = (CallbackInfo *) ptr;
\r
2041 RtApiJack *object = (RtApiJack *) info->object;
\r
2043 object->closeStream();
\r
2045 pthread_exit( NULL );
\r
2047 static void jackShutdown( void *infoPointer )
\r
2049 CallbackInfo *info = (CallbackInfo *) infoPointer;
\r
2050 RtApiJack *object = (RtApiJack *) info->object;
\r
2052 // Check current stream state. If stopped, then we'll assume this
\r
2053 // was called as a result of a call to RtApiJack::stopStream (the
\r
2054 // deactivation of a client handle causes this function to be called).
\r
2055 // If not, we'll assume the Jack server is shutting down or some
\r
2056 // other problem occurred and we should close the stream.
\r
2057 if ( object->isStreamRunning() == false ) return;
\r
2059 ThreadHandle threadId;
\r
2060 pthread_create( &threadId, NULL, jackCloseStream, info );
\r
2061 std::cerr << "\nRtApiJack: the Jack server is shutting down this client ... stream stopped and closed!!\n" << std::endl;
\r
2064 static int jackXrun( void *infoPointer )
\r
2066 JackHandle *handle = (JackHandle *) infoPointer;
\r
2068 if ( handle->ports[0] ) handle->xrun[0] = true;
\r
2069 if ( handle->ports[1] ) handle->xrun[1] = true;
\r
2074 bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
\r
2075 unsigned int firstChannel, unsigned int sampleRate,
\r
2076 RtAudioFormat format, unsigned int *bufferSize,
\r
2077 RtAudio::StreamOptions *options )
\r
2079 JackHandle *handle = (JackHandle *) stream_.apiHandle;
\r
2081 // Look for jack server and try to become a client (only do once per stream).
\r
2082 jack_client_t *client = 0;
\r
2083 if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) {
\r
2084 jack_options_t jackoptions = (jack_options_t) ( JackNoStartServer ); //JackNullOption;
\r
2085 jack_status_t *status = NULL;
\r
2086 if ( options && !options->streamName.empty() )
\r
2087 client = jack_client_open( options->streamName.c_str(), jackoptions, status );
\r
2089 client = jack_client_open( "RtApiJack", jackoptions, status );
\r
2090 if ( client == 0 ) {
\r
2091 errorText_ = "RtApiJack::probeDeviceOpen: Jack server not found or connection error!";
\r
2092 error( RtAudioError::WARNING );
\r
2097 // The handle must have been created on an earlier pass.
\r
2098 client = handle->client;
\r
2101 const char **ports;
\r
2102 std::string port, previousPort, deviceName;
\r
2103 unsigned int nPorts = 0, nDevices = 0;
\r
2104 ports = jack_get_ports( client, NULL, NULL, 0 );
\r
2106 // Parse the port names up to the first colon (:).
\r
2107 size_t iColon = 0;
\r
2109 port = (char *) ports[ nPorts ];
\r
2110 iColon = port.find(":");
\r
2111 if ( iColon != std::string::npos ) {
\r
2112 port = port.substr( 0, iColon );
\r
2113 if ( port != previousPort ) {
\r
2114 if ( nDevices == device ) deviceName = port;
\r
2116 previousPort = port;
\r
2119 } while ( ports[++nPorts] );
\r
2123 if ( device >= nDevices ) {
\r
2124 errorText_ = "RtApiJack::probeDeviceOpen: device ID is invalid!";
\r
2128 // Count the available ports containing the client name as device
\r
2129 // channels. Jack "input ports" equal RtAudio output channels.
\r
2130 unsigned int nChannels = 0;
\r
2131 unsigned long flag = JackPortIsInput;
\r
2132 if ( mode == INPUT ) flag = JackPortIsOutput;
\r
2133 ports = jack_get_ports( client, deviceName.c_str(), NULL, flag );
\r
2135 while ( ports[ nChannels ] ) nChannels++;
\r
2139 // Compare the jack ports for specified client to the requested number of channels.
\r
2140 if ( nChannels < (channels + firstChannel) ) {
\r
2141 errorStream_ << "RtApiJack::probeDeviceOpen: requested number of channels (" << channels << ") + offset (" << firstChannel << ") not found for specified device (" << device << ":" << deviceName << ").";
\r
2142 errorText_ = errorStream_.str();
\r
2146 // Check the jack server sample rate.
\r
2147 unsigned int jackRate = jack_get_sample_rate( client );
\r
2148 if ( sampleRate != jackRate ) {
\r
2149 jack_client_close( client );
\r
2150 errorStream_ << "RtApiJack::probeDeviceOpen: the requested sample rate (" << sampleRate << ") is different than the JACK server rate (" << jackRate << ").";
\r
2151 errorText_ = errorStream_.str();
\r
2154 stream_.sampleRate = jackRate;
\r
2156 // Get the latency of the JACK port.
\r
2157 ports = jack_get_ports( client, deviceName.c_str(), NULL, flag );
\r
2158 if ( ports[ firstChannel ] ) {
\r
2159 // Added by Ge Wang
\r
2160 jack_latency_callback_mode_t cbmode = (mode == INPUT ? JackCaptureLatency : JackPlaybackLatency);
\r
2161 // the range (usually the min and max are equal)
\r
2162 jack_latency_range_t latrange; latrange.min = latrange.max = 0;
\r
2163 // get the latency range
\r
2164 jack_port_get_latency_range( jack_port_by_name( client, ports[firstChannel] ), cbmode, &latrange );
\r
2165 // be optimistic, use the min!
\r
2166 stream_.latency[mode] = latrange.min;
\r
2167 //stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) );
\r
2171 // The jack server always uses 32-bit floating-point data.
\r
2172 stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
\r
2173 stream_.userFormat = format;
\r
2175 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
\r
2176 else stream_.userInterleaved = true;
\r
2178 // Jack always uses non-interleaved buffers.
\r
2179 stream_.deviceInterleaved[mode] = false;
\r
2181 // Jack always provides host byte-ordered data.
\r
2182 stream_.doByteSwap[mode] = false;
\r
2184 // Get the buffer size. The buffer size and number of buffers
\r
2185 // (periods) is set when the jack server is started.
\r
2186 stream_.bufferSize = (int) jack_get_buffer_size( client );
\r
2187 *bufferSize = stream_.bufferSize;
\r
2189 stream_.nDeviceChannels[mode] = channels;
\r
2190 stream_.nUserChannels[mode] = channels;
\r
2192 // Set flags for buffer conversion.
\r
2193 stream_.doConvertBuffer[mode] = false;
\r
2194 if ( stream_.userFormat != stream_.deviceFormat[mode] )
\r
2195 stream_.doConvertBuffer[mode] = true;
\r
2196 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
\r
2197 stream_.nUserChannels[mode] > 1 )
\r
2198 stream_.doConvertBuffer[mode] = true;
\r
2200 // Allocate our JackHandle structure for the stream.
\r
2201 if ( handle == 0 ) {
\r
2203 handle = new JackHandle;
\r
2205 catch ( std::bad_alloc& ) {
\r
2206 errorText_ = "RtApiJack::probeDeviceOpen: error allocating JackHandle memory.";
\r
2210 if ( pthread_cond_init(&handle->condition, NULL) ) {
\r
2211 errorText_ = "RtApiJack::probeDeviceOpen: error initializing pthread condition variable.";
\r
2214 stream_.apiHandle = (void *) handle;
\r
2215 handle->client = client;
\r
2217 handle->deviceName[mode] = deviceName;
\r
2219 // Allocate necessary internal buffers.
\r
2220 unsigned long bufferBytes;
\r
2221 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
\r
2222 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
\r
2223 if ( stream_.userBuffer[mode] == NULL ) {
\r
2224 errorText_ = "RtApiJack::probeDeviceOpen: error allocating user buffer memory.";
\r
2228 if ( stream_.doConvertBuffer[mode] ) {
\r
2230 bool makeBuffer = true;
\r
2231 if ( mode == OUTPUT )
\r
2232 bufferBytes = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
\r
2233 else { // mode == INPUT
\r
2234 bufferBytes = stream_.nDeviceChannels[1] * formatBytes( stream_.deviceFormat[1] );
\r
2235 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
\r
2236 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]);
\r
2237 if ( bufferBytes < bytesOut ) makeBuffer = false;
\r
2241 if ( makeBuffer ) {
\r
2242 bufferBytes *= *bufferSize;
\r
2243 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
\r
2244 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
\r
2245 if ( stream_.deviceBuffer == NULL ) {
\r
2246 errorText_ = "RtApiJack::probeDeviceOpen: error allocating device buffer memory.";
\r
2252 // Allocate memory for the Jack ports (channels) identifiers.
\r
2253 handle->ports[mode] = (jack_port_t **) malloc ( sizeof (jack_port_t *) * channels );
\r
2254 if ( handle->ports[mode] == NULL ) {
\r
2255 errorText_ = "RtApiJack::probeDeviceOpen: error allocating port memory.";
\r
2259 stream_.device[mode] = device;
\r
2260 stream_.channelOffset[mode] = firstChannel;
\r
2261 stream_.state = STREAM_STOPPED;
\r
2262 stream_.callbackInfo.object = (void *) this;
\r
2264 if ( stream_.mode == OUTPUT && mode == INPUT )
\r
2265 // We had already set up the stream for output.
\r
2266 stream_.mode = DUPLEX;
\r
2268 stream_.mode = mode;
\r
2269 jack_set_process_callback( handle->client, jackCallbackHandler, (void *) &stream_.callbackInfo );
\r
2270 jack_set_xrun_callback( handle->client, jackXrun, (void *) &handle );
\r
2271 jack_on_shutdown( handle->client, jackShutdown, (void *) &stream_.callbackInfo );
\r
2274 // Register our ports.
\r
2276 if ( mode == OUTPUT ) {
\r
2277 for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
\r
2278 snprintf( label, 64, "outport %d", i );
\r
2279 handle->ports[0][i] = jack_port_register( handle->client, (const char *)label,
\r
2280 JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0 );
\r
2284 for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
\r
2285 snprintf( label, 64, "inport %d", i );
\r
2286 handle->ports[1][i] = jack_port_register( handle->client, (const char *)label,
\r
2287 JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0 );
\r
2291 // Setup the buffer conversion information structure. We don't use
\r
2292 // buffers to do channel offsets, so we override that parameter
\r
2294 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );
\r
2300 pthread_cond_destroy( &handle->condition );
\r
2301 jack_client_close( handle->client );
\r
2303 if ( handle->ports[0] ) free( handle->ports[0] );
\r
2304 if ( handle->ports[1] ) free( handle->ports[1] );
\r
2307 stream_.apiHandle = 0;
\r
2310 for ( int i=0; i<2; i++ ) {
\r
2311 if ( stream_.userBuffer[i] ) {
\r
2312 free( stream_.userBuffer[i] );
\r
2313 stream_.userBuffer[i] = 0;
\r
2317 if ( stream_.deviceBuffer ) {
\r
2318 free( stream_.deviceBuffer );
\r
2319 stream_.deviceBuffer = 0;
\r
2325 void RtApiJack :: closeStream( void )
\r
2327 if ( stream_.state == STREAM_CLOSED ) {
\r
2328 errorText_ = "RtApiJack::closeStream(): no open stream to close!";
\r
2329 error( RtAudioError::WARNING );
\r
2333 JackHandle *handle = (JackHandle *) stream_.apiHandle;
\r
2336 if ( stream_.state == STREAM_RUNNING )
\r
2337 jack_deactivate( handle->client );
\r
2339 jack_client_close( handle->client );
\r
2343 if ( handle->ports[0] ) free( handle->ports[0] );
\r
2344 if ( handle->ports[1] ) free( handle->ports[1] );
\r
2345 pthread_cond_destroy( &handle->condition );
\r
2347 stream_.apiHandle = 0;
\r
2350 for ( int i=0; i<2; i++ ) {
\r
2351 if ( stream_.userBuffer[i] ) {
\r
2352 free( stream_.userBuffer[i] );
\r
2353 stream_.userBuffer[i] = 0;
\r
2357 if ( stream_.deviceBuffer ) {
\r
2358 free( stream_.deviceBuffer );
\r
2359 stream_.deviceBuffer = 0;
\r
2362 stream_.mode = UNINITIALIZED;
\r
2363 stream_.state = STREAM_CLOSED;
\r
2366 void RtApiJack :: startStream( void )
\r
2369 if ( stream_.state == STREAM_RUNNING ) {
\r
2370 errorText_ = "RtApiJack::startStream(): the stream is already running!";
\r
2371 error( RtAudioError::WARNING );
\r
2375 JackHandle *handle = (JackHandle *) stream_.apiHandle;
\r
2376 int result = jack_activate( handle->client );
\r
2378 errorText_ = "RtApiJack::startStream(): unable to activate JACK client!";
\r
2382 const char **ports;
\r
2384 // Get the list of available ports.
\r
2385 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
\r
2387 ports = jack_get_ports( handle->client, handle->deviceName[0].c_str(), NULL, JackPortIsInput);
\r
2388 if ( ports == NULL) {
\r
2389 errorText_ = "RtApiJack::startStream(): error determining available JACK input ports!";
\r
2393 // Now make the port connections. Since RtAudio wasn't designed to
\r
2394 // allow the user to select particular channels of a device, we'll
\r
2395 // just open the first "nChannels" ports with offset.
\r
2396 for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
\r
2398 if ( ports[ stream_.channelOffset[0] + i ] )
\r
2399 result = jack_connect( handle->client, jack_port_name( handle->ports[0][i] ), ports[ stream_.channelOffset[0] + i ] );
\r
2402 errorText_ = "RtApiJack::startStream(): error connecting output ports!";
\r
2409 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
\r
2411 ports = jack_get_ports( handle->client, handle->deviceName[1].c_str(), NULL, JackPortIsOutput );
\r
2412 if ( ports == NULL) {
\r
2413 errorText_ = "RtApiJack::startStream(): error determining available JACK output ports!";
\r
2417 // Now make the port connections. See note above.
\r
2418 for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
\r
2420 if ( ports[ stream_.channelOffset[1] + i ] )
\r
2421 result = jack_connect( handle->client, ports[ stream_.channelOffset[1] + i ], jack_port_name( handle->ports[1][i] ) );
\r
2424 errorText_ = "RtApiJack::startStream(): error connecting input ports!";
\r
2431 handle->drainCounter = 0;
\r
2432 handle->internalDrain = false;
\r
2433 stream_.state = STREAM_RUNNING;
\r
2436 if ( result == 0 ) return;
\r
2437 error( RtAudioError::SYSTEM_ERROR );
\r
2440 void RtApiJack :: stopStream( void )
\r
2443 if ( stream_.state == STREAM_STOPPED ) {
\r
2444 errorText_ = "RtApiJack::stopStream(): the stream is already stopped!";
\r
2445 error( RtAudioError::WARNING );
\r
2449 JackHandle *handle = (JackHandle *) stream_.apiHandle;
\r
2450 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
\r
2452 if ( handle->drainCounter == 0 ) {
\r
2453 handle->drainCounter = 2;
\r
2454 pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
\r
2458 jack_deactivate( handle->client );
\r
2459 stream_.state = STREAM_STOPPED;
\r
2462 void RtApiJack :: abortStream( void )
\r
2465 if ( stream_.state == STREAM_STOPPED ) {
\r
2466 errorText_ = "RtApiJack::abortStream(): the stream is already stopped!";
\r
2467 error( RtAudioError::WARNING );
\r
2471 JackHandle *handle = (JackHandle *) stream_.apiHandle;
\r
2472 handle->drainCounter = 2;
\r
2477 // This function will be called by a spawned thread when the user
\r
2478 // callback function signals that the stream should be stopped or
\r
2479 // aborted. It is necessary to handle it this way because the
\r
2480 // callbackEvent() function must return before the jack_deactivate()
\r
2481 // function will return.
\r
2482 static void *jackStopStream( void *ptr )
\r
2484 CallbackInfo *info = (CallbackInfo *) ptr;
\r
2485 RtApiJack *object = (RtApiJack *) info->object;
\r
2487 object->stopStream();
\r
2488 pthread_exit( NULL );
\r
2491 bool RtApiJack :: callbackEvent( unsigned long nframes )
\r
2493 if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;
\r
2494 if ( stream_.state == STREAM_CLOSED ) {
\r
2495 errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";
\r
2496 error( RtAudioError::WARNING );
\r
2499 if ( stream_.bufferSize != nframes ) {
\r
2500 errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!";
\r
2501 error( RtAudioError::WARNING );
\r
2505 CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
\r
2506 JackHandle *handle = (JackHandle *) stream_.apiHandle;
\r
2508 // Check if we were draining the stream and signal is finished.
\r
2509 if ( handle->drainCounter > 3 ) {
\r
2510 ThreadHandle threadId;
\r
2512 stream_.state = STREAM_STOPPING;
\r
2513 if ( handle->internalDrain == true )
\r
2514 pthread_create( &threadId, NULL, jackStopStream, info );
\r
2516 pthread_cond_signal( &handle->condition );
\r
2520 // Invoke user callback first, to get fresh output data.
\r
2521 if ( handle->drainCounter == 0 ) {
\r
2522 RtAudioCallback callback = (RtAudioCallback) info->callback;
\r
2523 double streamTime = getStreamTime();
\r
2524 RtAudioStreamStatus status = 0;
\r
2525 if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
\r
2526 status |= RTAUDIO_OUTPUT_UNDERFLOW;
\r
2527 handle->xrun[0] = false;
\r
2529 if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
\r
2530 status |= RTAUDIO_INPUT_OVERFLOW;
\r
2531 handle->xrun[1] = false;
\r
2533 int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
\r
2534 stream_.bufferSize, streamTime, status, info->userData );
\r
2535 if ( cbReturnValue == 2 ) {
\r
2536 stream_.state = STREAM_STOPPING;
\r
2537 handle->drainCounter = 2;
\r
2539 pthread_create( &id, NULL, jackStopStream, info );
\r
2542 else if ( cbReturnValue == 1 ) {
\r
2543 handle->drainCounter = 1;
\r
2544 handle->internalDrain = true;
\r
2548 jack_default_audio_sample_t *jackbuffer;
\r
2549 unsigned long bufferBytes = nframes * sizeof( jack_default_audio_sample_t );
\r
2550 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
\r
2552 if ( handle->drainCounter > 1 ) { // write zeros to the output stream
\r
2554 for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
\r
2555 jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
\r
2556 memset( jackbuffer, 0, bufferBytes );
\r
2560 else if ( stream_.doConvertBuffer[0] ) {
\r
2562 convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
\r
2564 for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
\r
2565 jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
\r
2566 memcpy( jackbuffer, &stream_.deviceBuffer[i*bufferBytes], bufferBytes );
\r
2569 else { // no buffer conversion
\r
2570 for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
\r
2571 jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
\r
2572 memcpy( jackbuffer, &stream_.userBuffer[0][i*bufferBytes], bufferBytes );
\r
2576 if ( handle->drainCounter ) {
\r
2577 handle->drainCounter++;
\r
2582 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
\r
2584 if ( stream_.doConvertBuffer[1] ) {
\r
2585 for ( unsigned int i=0; i<stream_.nDeviceChannels[1]; i++ ) {
\r
2586 jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );
\r
2587 memcpy( &stream_.deviceBuffer[i*bufferBytes], jackbuffer, bufferBytes );
\r
2589 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
\r
2591 else { // no buffer conversion
\r
2592 for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
\r
2593 jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );
\r
2594 memcpy( &stream_.userBuffer[1][i*bufferBytes], jackbuffer, bufferBytes );
\r
2600 RtApi::tickStreamTime();
\r
2603 //******************** End of __UNIX_JACK__ *********************//
\r
2606 #if defined(__WINDOWS_ASIO__) // ASIO API on Windows
\r
2608 // The ASIO API is designed around a callback scheme, so this
\r
2609 // implementation is similar to that used for OS-X CoreAudio and Linux
\r
2610 // Jack. The primary constraint with ASIO is that it only allows
\r
2611 // access to a single driver at a time. Thus, it is not possible to
\r
2612 // have more than one simultaneous RtAudio stream.
\r
2614 // This implementation also requires a number of external ASIO files
\r
2615 // and a few global variables. The ASIO callback scheme does not
\r
2616 // allow for the passing of user data, so we must create a global
\r
2617 // pointer to our callbackInfo structure.
\r
2619 // On unix systems, we make use of a pthread condition variable.
\r
2620 // Since there is no equivalent in Windows, I hacked something based
\r
2621 // on information found in
\r
2622 // http://www.cs.wustl.edu/~schmidt/win32-cv-1.html.
\r
2624 #include "asiosys.h"
\r
2626 #include "iasiothiscallresolver.h"
\r
2627 #include "asiodrivers.h"
\r
2630 static AsioDrivers drivers;
\r
2631 static ASIOCallbacks asioCallbacks;
\r
2632 static ASIODriverInfo driverInfo;
\r
2633 static CallbackInfo *asioCallbackInfo;
\r
2634 static bool asioXRun;
\r
2636 struct AsioHandle {
\r
2637 int drainCounter; // Tracks callback counts when draining
\r
2638 bool internalDrain; // Indicates if stop is initiated from callback or not.
\r
2639 ASIOBufferInfo *bufferInfos;
\r
2643 :drainCounter(0), internalDrain(false), bufferInfos(0) {}
\r
2646 // Function declarations (definitions at end of section)
\r
2647 static const char* getAsioErrorString( ASIOError result );
\r
2648 static void sampleRateChanged( ASIOSampleRate sRate );
\r
2649 static long asioMessages( long selector, long value, void* message, double* opt );
\r
2651 RtApiAsio :: RtApiAsio()
\r
2653 // ASIO cannot run on a multi-threaded appartment. You can call
\r
2654 // CoInitialize beforehand, but it must be for appartment threading
\r
2655 // (in which case, CoInitilialize will return S_FALSE here).
\r
2656 coInitialized_ = false;
\r
2657 HRESULT hr = CoInitialize( NULL );
\r
2658 if ( FAILED(hr) ) {
\r
2659 errorText_ = "RtApiAsio::ASIO requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)";
\r
2660 error( RtAudioError::WARNING );
\r
2662 coInitialized_ = true;
\r
2664 drivers.removeCurrentDriver();
\r
2665 driverInfo.asioVersion = 2;
\r
2667 // See note in DirectSound implementation about GetDesktopWindow().
\r
2668 driverInfo.sysRef = GetForegroundWindow();
\r
2671 RtApiAsio :: ~RtApiAsio()
\r
2673 if ( stream_.state != STREAM_CLOSED ) closeStream();
\r
2674 if ( coInitialized_ ) CoUninitialize();
\r
2677 unsigned int RtApiAsio :: getDeviceCount( void )
\r
2679 return (unsigned int) drivers.asioGetNumDev();
\r
2682 RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device )
\r
2684 RtAudio::DeviceInfo info;
\r
2685 info.probed = false;
\r
2688 unsigned int nDevices = getDeviceCount();
\r
2689 if ( nDevices == 0 ) {
\r
2690 errorText_ = "RtApiAsio::getDeviceInfo: no devices found!";
\r
2691 error( RtAudioError::INVALID_USE );
\r
2695 if ( device >= nDevices ) {
\r
2696 errorText_ = "RtApiAsio::getDeviceInfo: device ID is invalid!";
\r
2697 error( RtAudioError::INVALID_USE );
\r
2701 // If a stream is already open, we cannot probe other devices. Thus, use the saved results.
\r
2702 if ( stream_.state != STREAM_CLOSED ) {
\r
2703 if ( device >= devices_.size() ) {
\r
2704 errorText_ = "RtApiAsio::getDeviceInfo: device ID was not present before stream was opened.";
\r
2705 error( RtAudioError::WARNING );
\r
2708 return devices_[ device ];
\r
2711 char driverName[32];
\r
2712 ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 );
\r
2713 if ( result != ASE_OK ) {
\r
2714 errorStream_ << "RtApiAsio::getDeviceInfo: unable to get driver name (" << getAsioErrorString( result ) << ").";
\r
2715 errorText_ = errorStream_.str();
\r
2716 error( RtAudioError::WARNING );
\r
2720 info.name = driverName;
\r
2722 if ( !drivers.loadDriver( driverName ) ) {
\r
2723 errorStream_ << "RtApiAsio::getDeviceInfo: unable to load driver (" << driverName << ").";
\r
2724 errorText_ = errorStream_.str();
\r
2725 error( RtAudioError::WARNING );
\r
2729 result = ASIOInit( &driverInfo );
\r
2730 if ( result != ASE_OK ) {
\r
2731 errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
\r
2732 errorText_ = errorStream_.str();
\r
2733 error( RtAudioError::WARNING );
\r
2737 // Determine the device channel information.
\r
2738 long inputChannels, outputChannels;
\r
2739 result = ASIOGetChannels( &inputChannels, &outputChannels );
\r
2740 if ( result != ASE_OK ) {
\r
2741 drivers.removeCurrentDriver();
\r
2742 errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
\r
2743 errorText_ = errorStream_.str();
\r
2744 error( RtAudioError::WARNING );
\r
2748 info.outputChannels = outputChannels;
\r
2749 info.inputChannels = inputChannels;
\r
2750 if ( info.outputChannels > 0 && info.inputChannels > 0 )
\r
2751 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
\r
2753 // Determine the supported sample rates.
\r
2754 info.sampleRates.clear();
\r
2755 for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {
\r
2756 result = ASIOCanSampleRate( (ASIOSampleRate) SAMPLE_RATES[i] );
\r
2757 if ( result == ASE_OK )
\r
2758 info.sampleRates.push_back( SAMPLE_RATES[i] );
\r
2761 // Determine supported data types ... just check first channel and assume rest are the same.
\r
2762 ASIOChannelInfo channelInfo;
\r
2763 channelInfo.channel = 0;
\r
2764 channelInfo.isInput = true;
\r
2765 if ( info.inputChannels <= 0 ) channelInfo.isInput = false;
\r
2766 result = ASIOGetChannelInfo( &channelInfo );
\r
2767 if ( result != ASE_OK ) {
\r
2768 drivers.removeCurrentDriver();
\r
2769 errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting driver channel info (" << driverName << ").";
\r
2770 errorText_ = errorStream_.str();
\r
2771 error( RtAudioError::WARNING );
\r
2775 info.nativeFormats = 0;
\r
2776 if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB )
\r
2777 info.nativeFormats |= RTAUDIO_SINT16;
\r
2778 else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB )
\r
2779 info.nativeFormats |= RTAUDIO_SINT32;
\r
2780 else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB )
\r
2781 info.nativeFormats |= RTAUDIO_FLOAT32;
\r
2782 else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB )
\r
2783 info.nativeFormats |= RTAUDIO_FLOAT64;
\r
2784 else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB )
\r
2785 info.nativeFormats |= RTAUDIO_SINT24;
\r
2787 if ( info.outputChannels > 0 )
\r
2788 if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true;
\r
2789 if ( info.inputChannels > 0 )
\r
2790 if ( getDefaultInputDevice() == device ) info.isDefaultInput = true;
\r
2792 info.probed = true;
\r
2793 drivers.removeCurrentDriver();
\r
2797 static void bufferSwitch( long index, ASIOBool /*processNow*/ )
\r
2799 RtApiAsio *object = (RtApiAsio *) asioCallbackInfo->object;
\r
2800 object->callbackEvent( index );
\r
2803 void RtApiAsio :: saveDeviceInfo( void )
\r
2807 unsigned int nDevices = getDeviceCount();
\r
2808 devices_.resize( nDevices );
\r
2809 for ( unsigned int i=0; i<nDevices; i++ )
\r
2810 devices_[i] = getDeviceInfo( i );
\r
2813 bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
\r
2814 unsigned int firstChannel, unsigned int sampleRate,
\r
2815 RtAudioFormat format, unsigned int *bufferSize,
\r
2816 RtAudio::StreamOptions *options )
\r
2818 // For ASIO, a duplex stream MUST use the same driver.
\r
2819 if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] != device ) {
\r
2820 errorText_ = "RtApiAsio::probeDeviceOpen: an ASIO duplex stream must use the same device for input and output!";
\r
2824 char driverName[32];
\r
2825 ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 );
\r
2826 if ( result != ASE_OK ) {
\r
2827 errorStream_ << "RtApiAsio::probeDeviceOpen: unable to get driver name (" << getAsioErrorString( result ) << ").";
\r
2828 errorText_ = errorStream_.str();
\r
2832 // Only load the driver once for duplex stream.
\r
2833 if ( mode != INPUT || stream_.mode != OUTPUT ) {
\r
2834 // The getDeviceInfo() function will not work when a stream is open
\r
2835 // because ASIO does not allow multiple devices to run at the same
\r
2836 // time. Thus, we'll probe the system before opening a stream and
\r
2837 // save the results for use by getDeviceInfo().
\r
2838 this->saveDeviceInfo();
\r
2840 if ( !drivers.loadDriver( driverName ) ) {
\r
2841 errorStream_ << "RtApiAsio::probeDeviceOpen: unable to load driver (" << driverName << ").";
\r
2842 errorText_ = errorStream_.str();
\r
2846 result = ASIOInit( &driverInfo );
\r
2847 if ( result != ASE_OK ) {
\r
2848 errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
\r
2849 errorText_ = errorStream_.str();
\r
2854 // Check the device channel count.
\r
2855 long inputChannels, outputChannels;
\r
2856 result = ASIOGetChannels( &inputChannels, &outputChannels );
\r
2857 if ( result != ASE_OK ) {
\r
2858 drivers.removeCurrentDriver();
\r
2859 errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
\r
2860 errorText_ = errorStream_.str();
\r
2864 if ( ( mode == OUTPUT && (channels+firstChannel) > (unsigned int) outputChannels) ||
\r
2865 ( mode == INPUT && (channels+firstChannel) > (unsigned int) inputChannels) ) {
\r
2866 drivers.removeCurrentDriver();
\r
2867 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested channel count (" << channels << ") + offset (" << firstChannel << ").";
\r
2868 errorText_ = errorStream_.str();
\r
2871 stream_.nDeviceChannels[mode] = channels;
\r
2872 stream_.nUserChannels[mode] = channels;
\r
2873 stream_.channelOffset[mode] = firstChannel;
\r
2875 // Verify the sample rate is supported.
\r
2876 result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate );
\r
2877 if ( result != ASE_OK ) {
\r
2878 drivers.removeCurrentDriver();
\r
2879 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested sample rate (" << sampleRate << ").";
\r
2880 errorText_ = errorStream_.str();
\r
2884 // Get the current sample rate
\r
2885 ASIOSampleRate currentRate;
\r
2886 result = ASIOGetSampleRate( ¤tRate );
\r
2887 if ( result != ASE_OK ) {
\r
2888 drivers.removeCurrentDriver();
\r
2889 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error getting sample rate.";
\r
2890 errorText_ = errorStream_.str();
\r
2894 // Set the sample rate only if necessary
\r
2895 if ( currentRate != sampleRate ) {
\r
2896 result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate );
\r
2897 if ( result != ASE_OK ) {
\r
2898 drivers.removeCurrentDriver();
\r
2899 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ").";
\r
2900 errorText_ = errorStream_.str();
\r
2905 // Determine the driver data type.
\r
2906 ASIOChannelInfo channelInfo;
\r
2907 channelInfo.channel = 0;
\r
2908 if ( mode == OUTPUT ) channelInfo.isInput = false;
\r
2909 else channelInfo.isInput = true;
\r
2910 result = ASIOGetChannelInfo( &channelInfo );
\r
2911 if ( result != ASE_OK ) {
\r
2912 drivers.removeCurrentDriver();
\r
2913 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting data format.";
\r
2914 errorText_ = errorStream_.str();
\r
2918 // Assuming WINDOWS host is always little-endian.
\r
2919 stream_.doByteSwap[mode] = false;
\r
2920 stream_.userFormat = format;
\r
2921 stream_.deviceFormat[mode] = 0;
\r
2922 if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) {
\r
2923 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
\r
2924 if ( channelInfo.type == ASIOSTInt16MSB ) stream_.doByteSwap[mode] = true;
\r
2926 else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) {
\r
2927 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
\r
2928 if ( channelInfo.type == ASIOSTInt32MSB ) stream_.doByteSwap[mode] = true;
\r
2930 else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) {
\r
2931 stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
\r
2932 if ( channelInfo.type == ASIOSTFloat32MSB ) stream_.doByteSwap[mode] = true;
\r
2934 else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) {
\r
2935 stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
\r
2936 if ( channelInfo.type == ASIOSTFloat64MSB ) stream_.doByteSwap[mode] = true;
\r
2938 else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB ) {
\r
2939 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
\r
2940 if ( channelInfo.type == ASIOSTInt24MSB ) stream_.doByteSwap[mode] = true;
\r
2943 if ( stream_.deviceFormat[mode] == 0 ) {
\r
2944 drivers.removeCurrentDriver();
\r
2945 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") data format not supported by RtAudio.";
\r
2946 errorText_ = errorStream_.str();
\r
2950 // Set the buffer size. For a duplex stream, this will end up
\r
2951 // setting the buffer size based on the input constraints, which
\r
2953 long minSize, maxSize, preferSize, granularity;
\r
2954 result = ASIOGetBufferSize( &minSize, &maxSize, &preferSize, &granularity );
\r
2955 if ( result != ASE_OK ) {
\r
2956 drivers.removeCurrentDriver();
\r
2957 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting buffer size.";
\r
2958 errorText_ = errorStream_.str();
\r
2962 if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;
\r
2963 else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;
\r
2964 else if ( granularity == -1 ) {
\r
2965 // Make sure bufferSize is a power of two.
\r
2966 int log2_of_min_size = 0;
\r
2967 int log2_of_max_size = 0;
\r
2969 for ( unsigned int i = 0; i < sizeof(long) * 8; i++ ) {
\r
2970 if ( minSize & ((long)1 << i) ) log2_of_min_size = i;
\r
2971 if ( maxSize & ((long)1 << i) ) log2_of_max_size = i;
\r
2974 long min_delta = std::abs( (long)*bufferSize - ((long)1 << log2_of_min_size) );
\r
2975 int min_delta_num = log2_of_min_size;
\r
2977 for (int i = log2_of_min_size + 1; i <= log2_of_max_size; i++) {
\r
2978 long current_delta = std::abs( (long)*bufferSize - ((long)1 << i) );
\r
2979 if (current_delta < min_delta) {
\r
2980 min_delta = current_delta;
\r
2981 min_delta_num = i;
\r
2985 *bufferSize = ( (unsigned int)1 << min_delta_num );
\r
2986 if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;
\r
2987 else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;
\r
2989 else if ( granularity != 0 ) {
\r
2990 // Set to an even multiple of granularity, rounding up.
\r
2991 *bufferSize = (*bufferSize + granularity-1) / granularity * granularity;
\r
2994 if ( mode == INPUT && stream_.mode == OUTPUT && stream_.bufferSize != *bufferSize ) {
\r
2995 drivers.removeCurrentDriver();
\r
2996 errorText_ = "RtApiAsio::probeDeviceOpen: input/output buffersize discrepancy!";
\r
3000 stream_.bufferSize = *bufferSize;
\r
3001 stream_.nBuffers = 2;
\r
3003 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
\r
3004 else stream_.userInterleaved = true;
\r
3006 // ASIO always uses non-interleaved buffers.
\r
3007 stream_.deviceInterleaved[mode] = false;
\r
3009 // Allocate, if necessary, our AsioHandle structure for the stream.
\r
3010 AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
\r
3011 if ( handle == 0 ) {
\r
3013 handle = new AsioHandle;
\r
3015 catch ( std::bad_alloc& ) {
\r
3016 //if ( handle == NULL ) {
\r
3017 drivers.removeCurrentDriver();
\r
3018 errorText_ = "RtApiAsio::probeDeviceOpen: error allocating AsioHandle memory.";
\r
3021 handle->bufferInfos = 0;
\r
3023 // Create a manual-reset event.
\r
3024 handle->condition = CreateEvent( NULL, // no security
\r
3025 TRUE, // manual-reset
\r
3026 FALSE, // non-signaled initially
\r
3027 NULL ); // unnamed
\r
3028 stream_.apiHandle = (void *) handle;
\r
3031 // Create the ASIO internal buffers. Since RtAudio sets up input
\r
3032 // and output separately, we'll have to dispose of previously
\r
3033 // created output buffers for a duplex stream.
\r
3034 long inputLatency, outputLatency;
\r
3035 if ( mode == INPUT && stream_.mode == OUTPUT ) {
\r
3036 ASIODisposeBuffers();
\r
3037 if ( handle->bufferInfos ) free( handle->bufferInfos );
\r
3040 // Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure.
\r
3041 bool buffersAllocated = false;
\r
3042 unsigned int i, nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
\r
3043 handle->bufferInfos = (ASIOBufferInfo *) malloc( nChannels * sizeof(ASIOBufferInfo) );
\r
3044 if ( handle->bufferInfos == NULL ) {
\r
3045 errorStream_ << "RtApiAsio::probeDeviceOpen: error allocating bufferInfo memory for driver (" << driverName << ").";
\r
3046 errorText_ = errorStream_.str();
\r
3050 ASIOBufferInfo *infos;
\r
3051 infos = handle->bufferInfos;
\r
3052 for ( i=0; i<stream_.nDeviceChannels[0]; i++, infos++ ) {
\r
3053 infos->isInput = ASIOFalse;
\r
3054 infos->channelNum = i + stream_.channelOffset[0];
\r
3055 infos->buffers[0] = infos->buffers[1] = 0;
\r
3057 for ( i=0; i<stream_.nDeviceChannels[1]; i++, infos++ ) {
\r
3058 infos->isInput = ASIOTrue;
\r
3059 infos->channelNum = i + stream_.channelOffset[1];
\r
3060 infos->buffers[0] = infos->buffers[1] = 0;
\r
3063 // Set up the ASIO callback structure and create the ASIO data buffers.
\r
3064 asioCallbacks.bufferSwitch = &bufferSwitch;
\r
3065 asioCallbacks.sampleRateDidChange = &sampleRateChanged;
\r
3066 asioCallbacks.asioMessage = &asioMessages;
\r
3067 asioCallbacks.bufferSwitchTimeInfo = NULL;
\r
3068 result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks );
\r
3069 if ( result != ASE_OK ) {
\r
3070 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") creating buffers.";
\r
3071 errorText_ = errorStream_.str();
\r
3074 buffersAllocated = true;
\r
3076 // Set flags for buffer conversion.
\r
3077 stream_.doConvertBuffer[mode] = false;
\r
3078 if ( stream_.userFormat != stream_.deviceFormat[mode] )
\r
3079 stream_.doConvertBuffer[mode] = true;
\r
3080 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
\r
3081 stream_.nUserChannels[mode] > 1 )
\r
3082 stream_.doConvertBuffer[mode] = true;
\r
3084 // Allocate necessary internal buffers
\r
3085 unsigned long bufferBytes;
\r
3086 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
\r
3087 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
\r
3088 if ( stream_.userBuffer[mode] == NULL ) {
\r
3089 errorText_ = "RtApiAsio::probeDeviceOpen: error allocating user buffer memory.";
\r
3093 if ( stream_.doConvertBuffer[mode] ) {
\r
3095 bool makeBuffer = true;
\r
3096 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
\r
3097 if ( mode == INPUT ) {
\r
3098 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
\r
3099 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
\r
3100 if ( bufferBytes <= bytesOut ) makeBuffer = false;
\r
3104 if ( makeBuffer ) {
\r
3105 bufferBytes *= *bufferSize;
\r
3106 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
\r
3107 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
\r
3108 if ( stream_.deviceBuffer == NULL ) {
\r
3109 errorText_ = "RtApiAsio::probeDeviceOpen: error allocating device buffer memory.";
\r
3115 stream_.sampleRate = sampleRate;
\r
3116 stream_.device[mode] = device;
\r
3117 stream_.state = STREAM_STOPPED;
\r
3118 asioCallbackInfo = &stream_.callbackInfo;
\r
3119 stream_.callbackInfo.object = (void *) this;
\r
3120 if ( stream_.mode == OUTPUT && mode == INPUT )
\r
3121 // We had already set up an output stream.
\r
3122 stream_.mode = DUPLEX;
\r
3124 stream_.mode = mode;
\r
3126 // Determine device latencies
\r
3127 result = ASIOGetLatencies( &inputLatency, &outputLatency );
\r
3128 if ( result != ASE_OK ) {
\r
3129 errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting latency.";
\r
3130 errorText_ = errorStream_.str();
\r
3131 error( RtAudioError::WARNING); // warn but don't fail
\r
3134 stream_.latency[0] = outputLatency;
\r
3135 stream_.latency[1] = inputLatency;
\r
3138 // Setup the buffer conversion information structure. We don't use
\r
3139 // buffers to do channel offsets, so we override that parameter
\r
3141 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );
\r
3146 if ( buffersAllocated )
\r
3147 ASIODisposeBuffers();
\r
3148 drivers.removeCurrentDriver();
\r
3151 CloseHandle( handle->condition );
\r
3152 if ( handle->bufferInfos )
\r
3153 free( handle->bufferInfos );
\r
3155 stream_.apiHandle = 0;
\r
3158 for ( int i=0; i<2; i++ ) {
\r
3159 if ( stream_.userBuffer[i] ) {
\r
3160 free( stream_.userBuffer[i] );
\r
3161 stream_.userBuffer[i] = 0;
\r
3165 if ( stream_.deviceBuffer ) {
\r
3166 free( stream_.deviceBuffer );
\r
3167 stream_.deviceBuffer = 0;
\r
3173 void RtApiAsio :: closeStream()
\r
3175 if ( stream_.state == STREAM_CLOSED ) {
\r
3176 errorText_ = "RtApiAsio::closeStream(): no open stream to close!";
\r
3177 error( RtAudioError::WARNING );
\r
3181 if ( stream_.state == STREAM_RUNNING ) {
\r
3182 stream_.state = STREAM_STOPPED;
\r
3185 ASIODisposeBuffers();
\r
3186 drivers.removeCurrentDriver();
\r
3188 AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
\r
3190 CloseHandle( handle->condition );
\r
3191 if ( handle->bufferInfos )
\r
3192 free( handle->bufferInfos );
\r
3194 stream_.apiHandle = 0;
\r
3197 for ( int i=0; i<2; i++ ) {
\r
3198 if ( stream_.userBuffer[i] ) {
\r
3199 free( stream_.userBuffer[i] );
\r
3200 stream_.userBuffer[i] = 0;
\r
3204 if ( stream_.deviceBuffer ) {
\r
3205 free( stream_.deviceBuffer );
\r
3206 stream_.deviceBuffer = 0;
\r
3209 stream_.mode = UNINITIALIZED;
\r
3210 stream_.state = STREAM_CLOSED;
\r
3213 bool stopThreadCalled = false;
\r
3215 void RtApiAsio :: startStream()
\r
3218 if ( stream_.state == STREAM_RUNNING ) {
\r
3219 errorText_ = "RtApiAsio::startStream(): the stream is already running!";
\r
3220 error( RtAudioError::WARNING );
\r
3224 AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
\r
3225 ASIOError result = ASIOStart();
\r
3226 if ( result != ASE_OK ) {
\r
3227 errorStream_ << "RtApiAsio::startStream: error (" << getAsioErrorString( result ) << ") starting device.";
\r
3228 errorText_ = errorStream_.str();
\r
3232 handle->drainCounter = 0;
\r
3233 handle->internalDrain = false;
\r
3234 ResetEvent( handle->condition );
\r
3235 stream_.state = STREAM_RUNNING;
\r
3239 stopThreadCalled = false;
\r
3241 if ( result == ASE_OK ) return;
\r
3242 error( RtAudioError::SYSTEM_ERROR );
\r
3245 void RtApiAsio :: stopStream()
\r
3248 if ( stream_.state == STREAM_STOPPED ) {
\r
3249 errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!";
\r
3250 error( RtAudioError::WARNING );
\r
3254 AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
\r
3255 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
\r
3256 if ( handle->drainCounter == 0 ) {
\r
3257 handle->drainCounter = 2;
\r
3258 WaitForSingleObject( handle->condition, INFINITE ); // block until signaled
\r
3262 stream_.state = STREAM_STOPPED;
\r
3264 ASIOError result = ASIOStop();
\r
3265 if ( result != ASE_OK ) {
\r
3266 errorStream_ << "RtApiAsio::stopStream: error (" << getAsioErrorString( result ) << ") stopping device.";
\r
3267 errorText_ = errorStream_.str();
\r
3270 if ( result == ASE_OK ) return;
\r
3271 error( RtAudioError::SYSTEM_ERROR );
\r
3274 void RtApiAsio :: abortStream()
\r
3277 if ( stream_.state == STREAM_STOPPED ) {
\r
3278 errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!";
\r
3279 error( RtAudioError::WARNING );
\r
3283 // The following lines were commented-out because some behavior was
\r
3284 // noted where the device buffers need to be zeroed to avoid
\r
3285 // continuing sound, even when the device buffers are completely
\r
3286 // disposed. So now, calling abort is the same as calling stop.
\r
3287 // AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
\r
3288 // handle->drainCounter = 2;
\r
3292 // This function will be called by a spawned thread when the user
\r
3293 // callback function signals that the stream should be stopped or
\r
3294 // aborted. It is necessary to handle it this way because the
\r
3295 // callbackEvent() function must return before the ASIOStop()
\r
3296 // function will return.
\r
3297 static unsigned __stdcall asioStopStream( void *ptr )
\r
3299 CallbackInfo *info = (CallbackInfo *) ptr;
\r
3300 RtApiAsio *object = (RtApiAsio *) info->object;
\r
3302 object->stopStream();
\r
3303 _endthreadex( 0 );
\r
3307 bool RtApiAsio :: callbackEvent( long bufferIndex )
\r
3309 if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;
\r
3310 if ( stream_.state == STREAM_CLOSED ) {
\r
3311 errorText_ = "RtApiAsio::callbackEvent(): the stream is closed ... this shouldn't happen!";
\r
3312 error( RtAudioError::WARNING );
\r
3316 CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
\r
3317 AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
\r
3319 // Check if we were draining the stream and signal if finished.
\r
3320 if ( handle->drainCounter > 3 ) {
\r
3322 stream_.state = STREAM_STOPPING;
\r
3323 if ( handle->internalDrain == false )
\r
3324 SetEvent( handle->condition );
\r
3325 else { // spawn a thread to stop the stream
\r
3326 unsigned threadId;
\r
3327 stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream,
\r
3328 &stream_.callbackInfo, 0, &threadId );
\r
3333 // Invoke user callback to get fresh output data UNLESS we are
\r
3334 // draining stream.
\r
3335 if ( handle->drainCounter == 0 ) {
\r
3336 RtAudioCallback callback = (RtAudioCallback) info->callback;
\r
3337 double streamTime = getStreamTime();
\r
3338 RtAudioStreamStatus status = 0;
\r
3339 if ( stream_.mode != INPUT && asioXRun == true ) {
\r
3340 status |= RTAUDIO_OUTPUT_UNDERFLOW;
\r
3343 if ( stream_.mode != OUTPUT && asioXRun == true ) {
\r
3344 status |= RTAUDIO_INPUT_OVERFLOW;
\r
3347 int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
\r
3348 stream_.bufferSize, streamTime, status, info->userData );
\r
3349 if ( cbReturnValue == 2 ) {
\r
3350 stream_.state = STREAM_STOPPING;
\r
3351 handle->drainCounter = 2;
\r
3352 unsigned threadId;
\r
3353 stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream,
\r
3354 &stream_.callbackInfo, 0, &threadId );
\r
3357 else if ( cbReturnValue == 1 ) {
\r
3358 handle->drainCounter = 1;
\r
3359 handle->internalDrain = true;
\r
3363 unsigned int nChannels, bufferBytes, i, j;
\r
3364 nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
\r
3365 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
\r
3367 bufferBytes = stream_.bufferSize * formatBytes( stream_.deviceFormat[0] );
\r
3369 if ( handle->drainCounter > 1 ) { // write zeros to the output stream
\r
3371 for ( i=0, j=0; i<nChannels; i++ ) {
\r
3372 if ( handle->bufferInfos[i].isInput != ASIOTrue )
\r
3373 memset( handle->bufferInfos[i].buffers[bufferIndex], 0, bufferBytes );
\r
3377 else if ( stream_.doConvertBuffer[0] ) {
\r
3379 convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
\r
3380 if ( stream_.doByteSwap[0] )
\r
3381 byteSwapBuffer( stream_.deviceBuffer,
\r
3382 stream_.bufferSize * stream_.nDeviceChannels[0],
\r
3383 stream_.deviceFormat[0] );
\r
3385 for ( i=0, j=0; i<nChannels; i++ ) {
\r
3386 if ( handle->bufferInfos[i].isInput != ASIOTrue )
\r
3387 memcpy( handle->bufferInfos[i].buffers[bufferIndex],
\r
3388 &stream_.deviceBuffer[j++*bufferBytes], bufferBytes );
\r
3394 if ( stream_.doByteSwap[0] )
\r
3395 byteSwapBuffer( stream_.userBuffer[0],
\r
3396 stream_.bufferSize * stream_.nUserChannels[0],
\r
3397 stream_.userFormat );
\r
3399 for ( i=0, j=0; i<nChannels; i++ ) {
\r
3400 if ( handle->bufferInfos[i].isInput != ASIOTrue )
\r
3401 memcpy( handle->bufferInfos[i].buffers[bufferIndex],
\r
3402 &stream_.userBuffer[0][bufferBytes*j++], bufferBytes );
\r
3407 if ( handle->drainCounter ) {
\r
3408 handle->drainCounter++;
\r
3413 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
\r
3415 bufferBytes = stream_.bufferSize * formatBytes(stream_.deviceFormat[1]);
\r
3417 if (stream_.doConvertBuffer[1]) {
\r
3419 // Always interleave ASIO input data.
\r
3420 for ( i=0, j=0; i<nChannels; i++ ) {
\r
3421 if ( handle->bufferInfos[i].isInput == ASIOTrue )
\r
3422 memcpy( &stream_.deviceBuffer[j++*bufferBytes],
\r
3423 handle->bufferInfos[i].buffers[bufferIndex],
\r
3427 if ( stream_.doByteSwap[1] )
\r
3428 byteSwapBuffer( stream_.deviceBuffer,
\r
3429 stream_.bufferSize * stream_.nDeviceChannels[1],
\r
3430 stream_.deviceFormat[1] );
\r
3431 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
\r
3435 for ( i=0, j=0; i<nChannels; i++ ) {
\r
3436 if ( handle->bufferInfos[i].isInput == ASIOTrue ) {
\r
3437 memcpy( &stream_.userBuffer[1][bufferBytes*j++],
\r
3438 handle->bufferInfos[i].buffers[bufferIndex],
\r
3443 if ( stream_.doByteSwap[1] )
\r
3444 byteSwapBuffer( stream_.userBuffer[1],
\r
3445 stream_.bufferSize * stream_.nUserChannels[1],
\r
3446 stream_.userFormat );
\r
3451 // The following call was suggested by Malte Clasen. While the API
\r
3452 // documentation indicates it should not be required, some device
\r
3453 // drivers apparently do not function correctly without it.
\r
3454 ASIOOutputReady();
\r
3456 RtApi::tickStreamTime();
\r
3460 static void sampleRateChanged( ASIOSampleRate sRate )
\r
3462 // The ASIO documentation says that this usually only happens during
\r
3463 // external sync. Audio processing is not stopped by the driver,
\r
3464 // actual sample rate might not have even changed, maybe only the
\r
3465 // sample rate status of an AES/EBU or S/PDIF digital input at the
\r
3468 RtApi *object = (RtApi *) asioCallbackInfo->object;
\r
3470 object->stopStream();
\r
3472 catch ( RtAudioError &exception ) {
\r
3473 std::cerr << "\nRtApiAsio: sampleRateChanged() error (" << exception.getMessage() << ")!\n" << std::endl;
\r
3477 std::cerr << "\nRtApiAsio: driver reports sample rate changed to " << sRate << " ... stream stopped!!!\n" << std::endl;
\r
3480 static long asioMessages( long selector, long value, void* /*message*/, double* /*opt*/ )
\r
3484 switch( selector ) {
\r
3485 case kAsioSelectorSupported:
\r
3486 if ( value == kAsioResetRequest
\r
3487 || value == kAsioEngineVersion
\r
3488 || value == kAsioResyncRequest
\r
3489 || value == kAsioLatenciesChanged
\r
3490 // The following three were added for ASIO 2.0, you don't
\r
3491 // necessarily have to support them.
\r
3492 || value == kAsioSupportsTimeInfo
\r
3493 || value == kAsioSupportsTimeCode
\r
3494 || value == kAsioSupportsInputMonitor)
\r
3497 case kAsioResetRequest:
\r
3498 // Defer the task and perform the reset of the driver during the
\r
3499 // next "safe" situation. You cannot reset the driver right now,
\r
3500 // as this code is called from the driver. Reset the driver is
\r
3501 // done by completely destruct is. I.e. ASIOStop(),
\r
3502 // ASIODisposeBuffers(), Destruction Afterwards you initialize the
\r
3504 std::cerr << "\nRtApiAsio: driver reset requested!!!" << std::endl;
\r
3507 case kAsioResyncRequest:
\r
3508 // This informs the application that the driver encountered some
\r
3509 // non-fatal data loss. It is used for synchronization purposes
\r
3510 // of different media. Added mainly to work around the Win16Mutex
\r
3511 // problems in Windows 95/98 with the Windows Multimedia system,
\r
3512 // which could lose data because the Mutex was held too long by
\r
3513 // another thread. However a driver can issue it in other
\r
3514 // situations, too.
\r
3515 // std::cerr << "\nRtApiAsio: driver resync requested!!!" << std::endl;
\r
3519 case kAsioLatenciesChanged:
\r
3520 // This will inform the host application that the drivers were
\r
3521 // latencies changed. Beware, it this does not mean that the
\r
3522 // buffer sizes have changed! You might need to update internal
\r
3524 std::cerr << "\nRtApiAsio: driver latency may have changed!!!" << std::endl;
\r
3527 case kAsioEngineVersion:
\r
3528 // Return the supported ASIO version of the host application. If
\r
3529 // a host application does not implement this selector, ASIO 1.0
\r
3530 // is assumed by the driver.
\r
3533 case kAsioSupportsTimeInfo:
\r
3534 // Informs the driver whether the
\r
3535 // asioCallbacks.bufferSwitchTimeInfo() callback is supported.
\r
3536 // For compatibility with ASIO 1.0 drivers the host application
\r
3537 // should always support the "old" bufferSwitch method, too.
\r
3540 case kAsioSupportsTimeCode:
\r
3541 // Informs the driver whether application is interested in time
\r
3542 // code info. If an application does not need to know about time
\r
3543 // code, the driver has less work to do.
\r
3550 static const char* getAsioErrorString( ASIOError result )
\r
3555 const char*message;
\r
3558 static const Messages m[] =
\r
3560 { ASE_NotPresent, "Hardware input or output is not present or available." },
\r
3561 { ASE_HWMalfunction, "Hardware is malfunctioning." },
\r
3562 { ASE_InvalidParameter, "Invalid input parameter." },
\r
3563 { ASE_InvalidMode, "Invalid mode." },
\r
3564 { ASE_SPNotAdvancing, "Sample position not advancing." },
\r
3565 { ASE_NoClock, "Sample clock or rate cannot be determined or is not present." },
\r
3566 { ASE_NoMemory, "Not enough memory to complete the request." }
\r
3569 for ( unsigned int i = 0; i < sizeof(m)/sizeof(m[0]); ++i )
\r
3570 if ( m[i].value == result ) return m[i].message;
\r
3572 return "Unknown error.";
\r
3575 //******************** End of __WINDOWS_ASIO__ *********************//
\r
3579 #if defined(__WINDOWS_WASAPI__) // Windows WASAPI API
\r
3581 // Authored by Marcus Tomlinson <themarcustomlinson@gmail.com>, April 2014
\r
3582 // - Introduces support for the Windows WASAPI API
\r
3583 // - Aims to deliver bit streams to and from hardware at the lowest possible latency, via the absolute minimum buffer sizes required
\r
3584 // - Provides flexible stream configuration to an otherwise strict and inflexible WASAPI interface
\r
3585 // - Includes automatic internal conversion of sample rate, buffer size and channel count
\r
3590 #include <audioclient.h>
\r
3592 #include <mmdeviceapi.h>
\r
3593 #include <functiondiscoverykeys_devpkey.h>
\r
3595 //=============================================================================
\r
3597 #define SAFE_RELEASE( objectPtr )\
\r
3600 objectPtr->Release();\
\r
3601 objectPtr = NULL;\
\r
3604 typedef HANDLE ( __stdcall *TAvSetMmThreadCharacteristicsPtr )( LPCWSTR TaskName, LPDWORD TaskIndex );
\r
3606 //-----------------------------------------------------------------------------
\r
3608 // WASAPI dictates stream sample rate, format, channel count, and in some cases, buffer size.
\r
3609 // Therefore we must perform all necessary conversions to user buffers in order to satisfy these
\r
3610 // requirements. WasapiBuffer ring buffers are used between HwIn->UserIn and UserOut->HwOut to
\r
3611 // provide intermediate storage for read / write synchronization.
\r
3612 class WasapiBuffer
\r
3616 : buffer_( NULL ),
\r
3625 // sets the length of the internal ring buffer
\r
3626 void setBufferSize( unsigned int bufferSize, unsigned int formatBytes ) {
\r
3629 buffer_ = ( char* ) calloc( bufferSize, formatBytes );
\r
3631 bufferSize_ = bufferSize;
\r
3636 // attempt to push a buffer into the ring buffer at the current "in" index
\r
3637 bool pushBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format )
\r
3639 if ( !buffer || // incoming buffer is NULL
\r
3640 bufferSize == 0 || // incoming buffer has no data
\r
3641 bufferSize > bufferSize_ ) // incoming buffer too large
\r
3646 unsigned int relOutIndex = outIndex_;
\r
3647 unsigned int inIndexEnd = inIndex_ + bufferSize;
\r
3648 if ( relOutIndex < inIndex_ && inIndexEnd >= bufferSize_ ) {
\r
3649 relOutIndex += bufferSize_;
\r
3652 // "in" index can end on the "out" index but cannot begin at it
\r
3653 if ( inIndex_ <= relOutIndex && inIndexEnd > relOutIndex ) {
\r
3654 return false; // not enough space between "in" index and "out" index
\r
3657 // copy buffer from external to internal
\r
3658 int fromZeroSize = inIndex_ + bufferSize - bufferSize_;
\r
3659 fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize;
\r
3660 int fromInSize = bufferSize - fromZeroSize;
\r
3664 case RTAUDIO_SINT8:
\r
3665 memcpy( &( ( char* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( char ) );
\r
3666 memcpy( buffer_, &( ( char* ) buffer )[fromInSize], fromZeroSize * sizeof( char ) );
\r
3668 case RTAUDIO_SINT16:
\r
3669 memcpy( &( ( short* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( short ) );
\r
3670 memcpy( buffer_, &( ( short* ) buffer )[fromInSize], fromZeroSize * sizeof( short ) );
\r
3672 case RTAUDIO_SINT24:
\r
3673 memcpy( &( ( S24* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( S24 ) );
\r
3674 memcpy( buffer_, &( ( S24* ) buffer )[fromInSize], fromZeroSize * sizeof( S24 ) );
\r
3676 case RTAUDIO_SINT32:
\r
3677 memcpy( &( ( int* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( int ) );
\r
3678 memcpy( buffer_, &( ( int* ) buffer )[fromInSize], fromZeroSize * sizeof( int ) );
\r
3680 case RTAUDIO_FLOAT32:
\r
3681 memcpy( &( ( float* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( float ) );
\r
3682 memcpy( buffer_, &( ( float* ) buffer )[fromInSize], fromZeroSize * sizeof( float ) );
\r
3684 case RTAUDIO_FLOAT64:
\r
3685 memcpy( &( ( double* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( double ) );
\r
3686 memcpy( buffer_, &( ( double* ) buffer )[fromInSize], fromZeroSize * sizeof( double ) );
\r
3690 // update "in" index
\r
3691 inIndex_ += bufferSize;
\r
3692 inIndex_ %= bufferSize_;
\r
3697 // attempt to pull a buffer from the ring buffer from the current "out" index
\r
3698 bool pullBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format )
\r
3700 if ( !buffer || // incoming buffer is NULL
\r
3701 bufferSize == 0 || // incoming buffer has no data
\r
3702 bufferSize > bufferSize_ ) // incoming buffer too large
\r
3707 unsigned int relInIndex = inIndex_;
\r
3708 unsigned int outIndexEnd = outIndex_ + bufferSize;
\r
3709 if ( relInIndex < outIndex_ && outIndexEnd >= bufferSize_ ) {
\r
3710 relInIndex += bufferSize_;
\r
3713 // "out" index can begin at and end on the "in" index
\r
3714 if ( outIndex_ < relInIndex && outIndexEnd > relInIndex ) {
\r
3715 return false; // not enough space between "out" index and "in" index
\r
3718 // copy buffer from internal to external
\r
3719 int fromZeroSize = outIndex_ + bufferSize - bufferSize_;
\r
3720 fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize;
\r
3721 int fromOutSize = bufferSize - fromZeroSize;
\r
3725 case RTAUDIO_SINT8:
\r
3726 memcpy( buffer, &( ( char* ) buffer_ )[outIndex_], fromOutSize * sizeof( char ) );
\r
3727 memcpy( &( ( char* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( char ) );
\r
3729 case RTAUDIO_SINT16:
\r
3730 memcpy( buffer, &( ( short* ) buffer_ )[outIndex_], fromOutSize * sizeof( short ) );
\r
3731 memcpy( &( ( short* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( short ) );
\r
3733 case RTAUDIO_SINT24:
\r
3734 memcpy( buffer, &( ( S24* ) buffer_ )[outIndex_], fromOutSize * sizeof( S24 ) );
\r
3735 memcpy( &( ( S24* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( S24 ) );
\r
3737 case RTAUDIO_SINT32:
\r
3738 memcpy( buffer, &( ( int* ) buffer_ )[outIndex_], fromOutSize * sizeof( int ) );
\r
3739 memcpy( &( ( int* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( int ) );
\r
3741 case RTAUDIO_FLOAT32:
\r
3742 memcpy( buffer, &( ( float* ) buffer_ )[outIndex_], fromOutSize * sizeof( float ) );
\r
3743 memcpy( &( ( float* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( float ) );
\r
3745 case RTAUDIO_FLOAT64:
\r
3746 memcpy( buffer, &( ( double* ) buffer_ )[outIndex_], fromOutSize * sizeof( double ) );
\r
3747 memcpy( &( ( double* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( double ) );
\r
3751 // update "out" index
\r
3752 outIndex_ += bufferSize;
\r
3753 outIndex_ %= bufferSize_;
\r
3760 unsigned int bufferSize_;
\r
3761 unsigned int inIndex_;
\r
3762 unsigned int outIndex_;
\r
3765 //-----------------------------------------------------------------------------
\r
3767 // In order to satisfy WASAPI's buffer requirements, we need a means of converting sample rate and
\r
3768 // channel counts between HW and the user. The convertBufferWasapi function is used to perform
\r
3769 // these conversions between HwIn->UserIn and UserOut->HwOut during the stream callback loop.
\r
3770 // This sample rate converter favors speed over quality, and works best with conversions between
\r
3771 // one rate and its multiple. RtApiWasapi will not populate a device's sample rate list with rates
\r
3772 // that may cause artifacts via this conversion.
\r
3773 void convertBufferWasapi( char* outBuffer,
\r
3774 const char* inBuffer,
\r
3775 const unsigned int& inChannelCount,
\r
3776 const unsigned int& outChannelCount,
\r
3777 const unsigned int& inSampleRate,
\r
3778 const unsigned int& outSampleRate,
\r
3779 const unsigned int& inSampleCount,
\r
3780 unsigned int& outSampleCount,
\r
3781 const RtAudioFormat& format )
\r
3783 // calculate the new outSampleCount and relative sampleStep
\r
3784 float sampleRatio = ( float ) outSampleRate / inSampleRate;
\r
3785 float sampleStep = 1.0f / sampleRatio;
\r
3786 float inSampleFraction = 0.0f;
\r
3787 unsigned int commonChannelCount = std::min( inChannelCount, outChannelCount );
\r
3789 outSampleCount = ( unsigned int ) ( inSampleCount * sampleRatio );
\r
3791 // frame-by-frame, copy each relative input sample into it's corresponding output sample
\r
3792 for ( unsigned int outSample = 0; outSample < outSampleCount; outSample++ )
\r
3794 unsigned int inSample = ( unsigned int ) inSampleFraction;
\r
3798 case RTAUDIO_SINT8:
\r
3799 memcpy( &( ( char* ) outBuffer )[ outSample * outChannelCount ], &( ( char* ) inBuffer )[ inSample * inChannelCount ], commonChannelCount * sizeof( char ) );
\r
3801 case RTAUDIO_SINT16:
\r
3802 memcpy( &( ( short* ) outBuffer )[ outSample * outChannelCount ], &( ( short* ) inBuffer )[ inSample * inChannelCount ], commonChannelCount * sizeof( short ) );
\r
3804 case RTAUDIO_SINT24:
\r
3805 memcpy( &( ( S24* ) outBuffer )[ outSample * outChannelCount ], &( ( S24* ) inBuffer )[ inSample * inChannelCount ], commonChannelCount * sizeof( S24 ) );
\r
3807 case RTAUDIO_SINT32:
\r
3808 memcpy( &( ( int* ) outBuffer )[ outSample * outChannelCount ], &( ( int* ) inBuffer )[ inSample * inChannelCount ], commonChannelCount * sizeof( int ) );
\r
3810 case RTAUDIO_FLOAT32:
\r
3811 memcpy( &( ( float* ) outBuffer )[ outSample * outChannelCount ], &( ( float* ) inBuffer )[ inSample * inChannelCount ], commonChannelCount * sizeof( float ) );
\r
3813 case RTAUDIO_FLOAT64:
\r
3814 memcpy( &( ( double* ) outBuffer )[ outSample * outChannelCount ], &( ( double* ) inBuffer )[ inSample * inChannelCount ], commonChannelCount * sizeof( double ) );
\r
3818 // jump to next in sample
\r
3819 inSampleFraction += sampleStep;
\r
3823 //-----------------------------------------------------------------------------
\r
3825 // A structure to hold various information related to the WASAPI implementation.
\r
3826 struct WasapiHandle
\r
3828 IAudioClient* captureAudioClient;
\r
3829 IAudioClient* renderAudioClient;
\r
3830 IAudioCaptureClient* captureClient;
\r
3831 IAudioRenderClient* renderClient;
\r
3832 HANDLE captureEvent;
\r
3833 HANDLE renderEvent;
\r
3836 : captureAudioClient( NULL ),
\r
3837 renderAudioClient( NULL ),
\r
3838 captureClient( NULL ),
\r
3839 renderClient( NULL ),
\r
3840 captureEvent( NULL ),
\r
3841 renderEvent( NULL ) {}
\r
3844 //=============================================================================
\r
3846 RtApiWasapi::RtApiWasapi()
\r
3847 : coInitialized_( false ), deviceEnumerator_( NULL )
\r
3849 // WASAPI can run either apartment or multi-threaded
\r
3850 HRESULT hr = CoInitialize( NULL );
\r
3852 if ( !FAILED( hr ) )
\r
3853 coInitialized_ = true;
\r
3855 // Instantiate device enumerator
\r
3856 hr = CoCreateInstance( __uuidof( MMDeviceEnumerator ), NULL,
\r
3857 CLSCTX_ALL, __uuidof( IMMDeviceEnumerator ),
\r
3858 ( void** ) &deviceEnumerator_ );
\r
3860 if ( FAILED( hr ) ) {
\r
3861 errorText_ = "RtApiWasapi::RtApiWasapi: Unable to instantiate device enumerator";
\r
3862 error( RtAudioError::DRIVER_ERROR );
\r
3866 //-----------------------------------------------------------------------------
\r
3868 RtApiWasapi::~RtApiWasapi()
\r
3870 // if this object previously called CoInitialize()
\r
3871 if ( coInitialized_ ) {
\r
3875 if ( stream_.state != STREAM_CLOSED ) {
\r
3879 SAFE_RELEASE( deviceEnumerator_ );
\r
3882 //=============================================================================
\r
3884 unsigned int RtApiWasapi::getDeviceCount( void )
\r
3886 unsigned int captureDeviceCount = 0;
\r
3887 unsigned int renderDeviceCount = 0;
\r
3889 IMMDeviceCollection* captureDevices = NULL;
\r
3890 IMMDeviceCollection* renderDevices = NULL;
\r
3892 // Count capture devices
\r
3893 errorText_.clear();
\r
3894 HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
\r
3895 if ( FAILED( hr ) ) {
\r
3896 errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device collection.";
\r
3900 hr = captureDevices->GetCount( &captureDeviceCount );
\r
3901 if ( FAILED( hr ) ) {
\r
3902 errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device count.";
\r
3906 // Count render devices
\r
3907 hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
\r
3908 if ( FAILED( hr ) ) {
\r
3909 errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device collection.";
\r
3913 hr = renderDevices->GetCount( &renderDeviceCount );
\r
3914 if ( FAILED( hr ) ) {
\r
3915 errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device count.";
\r
3920 // release all references
\r
3921 SAFE_RELEASE( captureDevices );
\r
3922 SAFE_RELEASE( renderDevices );
\r
3924 if ( errorText_.empty() )
\r
3925 return captureDeviceCount + renderDeviceCount;
\r
3927 error( RtAudioError::DRIVER_ERROR );
\r
3931 //-----------------------------------------------------------------------------
\r
3933 RtAudio::DeviceInfo RtApiWasapi::getDeviceInfo( unsigned int device )
\r
3935 RtAudio::DeviceInfo info;
\r
3936 unsigned int captureDeviceCount = 0;
\r
3937 unsigned int renderDeviceCount = 0;
\r
3938 std::wstring deviceName;
\r
3939 std::string defaultDeviceName;
\r
3940 bool isCaptureDevice = false;
\r
3942 PROPVARIANT deviceNameProp;
\r
3943 PROPVARIANT defaultDeviceNameProp;
\r
3945 IMMDeviceCollection* captureDevices = NULL;
\r
3946 IMMDeviceCollection* renderDevices = NULL;
\r
3947 IMMDevice* devicePtr = NULL;
\r
3948 IMMDevice* defaultDevicePtr = NULL;
\r
3949 IAudioClient* audioClient = NULL;
\r
3950 IPropertyStore* devicePropStore = NULL;
\r
3951 IPropertyStore* defaultDevicePropStore = NULL;
\r
3953 WAVEFORMATEX* deviceFormat = NULL;
\r
3954 WAVEFORMATEX* closestMatchFormat = NULL;
\r
3957 info.probed = false;
\r
3959 // Count capture devices
\r
3960 errorText_.clear();
\r
3961 RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
\r
3962 HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
\r
3963 if ( FAILED( hr ) ) {
\r
3964 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device collection.";
\r
3968 hr = captureDevices->GetCount( &captureDeviceCount );
\r
3969 if ( FAILED( hr ) ) {
\r
3970 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device count.";
\r
3974 // Count render devices
\r
3975 hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
\r
3976 if ( FAILED( hr ) ) {
\r
3977 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device collection.";
\r
3981 hr = renderDevices->GetCount( &renderDeviceCount );
\r
3982 if ( FAILED( hr ) ) {
\r
3983 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device count.";
\r
3987 // validate device index
\r
3988 if ( device >= captureDeviceCount + renderDeviceCount ) {
\r
3989 errorText_ = "RtApiWasapi::getDeviceInfo: Invalid device index.";
\r
3990 errorType = RtAudioError::INVALID_USE;
\r
3994 // determine whether index falls within capture or render devices
\r
3995 if ( device >= renderDeviceCount ) {
\r
3996 hr = captureDevices->Item( device - renderDeviceCount, &devicePtr );
\r
3997 if ( FAILED( hr ) ) {
\r
3998 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device handle.";
\r
4001 isCaptureDevice = true;
\r
4004 hr = renderDevices->Item( device, &devicePtr );
\r
4005 if ( FAILED( hr ) ) {
\r
4006 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device handle.";
\r
4009 isCaptureDevice = false;
\r
4012 // get default device name
\r
4013 if ( isCaptureDevice ) {
\r
4014 hr = deviceEnumerator_->GetDefaultAudioEndpoint( eCapture, eConsole, &defaultDevicePtr );
\r
4015 if ( FAILED( hr ) ) {
\r
4016 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default capture device handle.";
\r
4021 hr = deviceEnumerator_->GetDefaultAudioEndpoint( eRender, eConsole, &defaultDevicePtr );
\r
4022 if ( FAILED( hr ) ) {
\r
4023 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default render device handle.";
\r
4028 hr = defaultDevicePtr->OpenPropertyStore( STGM_READ, &defaultDevicePropStore );
\r
4029 if ( FAILED( hr ) ) {
\r
4030 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open default device property store.";
\r
4033 PropVariantInit( &defaultDeviceNameProp );
\r
4035 hr = defaultDevicePropStore->GetValue( PKEY_Device_FriendlyName, &defaultDeviceNameProp );
\r
4036 if ( FAILED( hr ) ) {
\r
4037 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default device property: PKEY_Device_FriendlyName.";
\r
4041 deviceName = defaultDeviceNameProp.pwszVal;
\r
4042 defaultDeviceName = std::string( deviceName.begin(), deviceName.end() );
\r
4045 hr = devicePtr->OpenPropertyStore( STGM_READ, &devicePropStore );
\r
4046 if ( FAILED( hr ) ) {
\r
4047 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open device property store.";
\r
4051 PropVariantInit( &deviceNameProp );
\r
4053 hr = devicePropStore->GetValue( PKEY_Device_FriendlyName, &deviceNameProp );
\r
4054 if ( FAILED( hr ) ) {
\r
4055 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device property: PKEY_Device_FriendlyName.";
\r
4059 deviceName = deviceNameProp.pwszVal;
\r
4060 info.name = std::string( deviceName.begin(), deviceName.end() );
\r
4063 if ( isCaptureDevice ) {
\r
4064 info.isDefaultInput = info.name == defaultDeviceName;
\r
4065 info.isDefaultOutput = false;
\r
4068 info.isDefaultInput = false;
\r
4069 info.isDefaultOutput = info.name == defaultDeviceName;
\r
4073 hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, NULL, ( void** ) &audioClient );
\r
4074 if ( FAILED( hr ) ) {
\r
4075 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device audio client.";
\r
4079 hr = audioClient->GetMixFormat( &deviceFormat );
\r
4080 if ( FAILED( hr ) ) {
\r
4081 errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device mix format.";
\r
4085 if ( isCaptureDevice ) {
\r
4086 info.inputChannels = deviceFormat->nChannels;
\r
4087 info.outputChannels = 0;
\r
4088 info.duplexChannels = 0;
\r
4091 info.inputChannels = 0;
\r
4092 info.outputChannels = deviceFormat->nChannels;
\r
4093 info.duplexChannels = 0;
\r
4097 info.sampleRates.clear();
\r
4099 // allow support for sample rates that are multiples of the base rate
\r
4100 for ( unsigned int i = 0; i < MAX_SAMPLE_RATES; i++ ) {
\r
4101 if ( SAMPLE_RATES[i] < deviceFormat->nSamplesPerSec ) {
\r
4102 if ( deviceFormat->nSamplesPerSec % SAMPLE_RATES[i] == 0 ) {
\r
4103 info.sampleRates.push_back( SAMPLE_RATES[i] );
\r
4107 if ( SAMPLE_RATES[i] % deviceFormat->nSamplesPerSec == 0 ) {
\r
4108 info.sampleRates.push_back( SAMPLE_RATES[i] );
\r
4114 info.nativeFormats = 0;
\r
4116 if ( deviceFormat->wFormatTag == WAVE_FORMAT_IEEE_FLOAT ||
\r
4117 ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
\r
4118 ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_IEEE_FLOAT ) )
\r
4120 if ( deviceFormat->wBitsPerSample == 32 ) {
\r
4121 info.nativeFormats |= RTAUDIO_FLOAT32;
\r
4123 else if ( deviceFormat->wBitsPerSample == 64 ) {
\r
4124 info.nativeFormats |= RTAUDIO_FLOAT64;
\r
4127 else if ( deviceFormat->wFormatTag == WAVE_FORMAT_PCM ||
\r
4128 ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
\r
4129 ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_PCM ) )
\r
4131 if ( deviceFormat->wBitsPerSample == 8 ) {
\r
4132 info.nativeFormats |= RTAUDIO_SINT8;
\r
4134 else if ( deviceFormat->wBitsPerSample == 16 ) {
\r
4135 info.nativeFormats |= RTAUDIO_SINT16;
\r
4137 else if ( deviceFormat->wBitsPerSample == 24 ) {
\r
4138 info.nativeFormats |= RTAUDIO_SINT24;
\r
4140 else if ( deviceFormat->wBitsPerSample == 32 ) {
\r
4141 info.nativeFormats |= RTAUDIO_SINT32;
\r
4146 info.probed = true;
\r
4149 // release all references
\r
4150 PropVariantClear( &deviceNameProp );
\r
4151 PropVariantClear( &defaultDeviceNameProp );
\r
4153 SAFE_RELEASE( captureDevices );
\r
4154 SAFE_RELEASE( renderDevices );
\r
4155 SAFE_RELEASE( devicePtr );
\r
4156 SAFE_RELEASE( defaultDevicePtr );
\r
4157 SAFE_RELEASE( audioClient );
\r
4158 SAFE_RELEASE( devicePropStore );
\r
4159 SAFE_RELEASE( defaultDevicePropStore );
\r
4161 CoTaskMemFree( deviceFormat );
\r
4162 CoTaskMemFree( closestMatchFormat );
\r
4164 if ( !errorText_.empty() )
\r
4165 error( errorType );
\r
4169 //-----------------------------------------------------------------------------
\r
4171 unsigned int RtApiWasapi::getDefaultOutputDevice( void )
\r
4173 for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {
\r
4174 if ( getDeviceInfo( i ).isDefaultOutput ) {
\r
4182 //-----------------------------------------------------------------------------
\r
4184 unsigned int RtApiWasapi::getDefaultInputDevice( void )
\r
4186 for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {
\r
4187 if ( getDeviceInfo( i ).isDefaultInput ) {
\r
4195 //-----------------------------------------------------------------------------
\r
4197 void RtApiWasapi::closeStream( void )
\r
4199 if ( stream_.state == STREAM_CLOSED ) {
\r
4200 errorText_ = "RtApiWasapi::closeStream: No open stream to close.";
\r
4201 error( RtAudioError::WARNING );
\r
4205 if ( stream_.state != STREAM_STOPPED )
\r
4208 // clean up stream memory
\r
4209 SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient )
\r
4210 SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient )
\r
4212 SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureClient )
\r
4213 SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderClient )
\r
4215 if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent )
\r
4216 CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent );
\r
4218 if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent )
\r
4219 CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent );
\r
4221 delete ( WasapiHandle* ) stream_.apiHandle;
\r
4222 stream_.apiHandle = NULL;
\r
4224 for ( int i = 0; i < 2; i++ ) {
\r
4225 if ( stream_.userBuffer[i] ) {
\r
4226 free( stream_.userBuffer[i] );
\r
4227 stream_.userBuffer[i] = 0;
\r
4231 if ( stream_.deviceBuffer ) {
\r
4232 free( stream_.deviceBuffer );
\r
4233 stream_.deviceBuffer = 0;
\r
4236 // update stream state
\r
4237 stream_.state = STREAM_CLOSED;
\r
4240 //-----------------------------------------------------------------------------
\r
4242 void RtApiWasapi::startStream( void )
\r
4246 if ( stream_.state == STREAM_RUNNING ) {
\r
4247 errorText_ = "RtApiWasapi::startStream: The stream is already running.";
\r
4248 error( RtAudioError::WARNING );
\r
4252 // update stream state
\r
4253 stream_.state = STREAM_RUNNING;
\r
4255 // create WASAPI stream thread
\r
4256 stream_.callbackInfo.thread = ( ThreadHandle ) CreateThread( NULL, 0, runWasapiThread, this, CREATE_SUSPENDED, NULL );
\r
4258 if ( !stream_.callbackInfo.thread ) {
\r
4259 errorText_ = "RtApiWasapi::startStream: Unable to instantiate callback thread.";
\r
4260 error( RtAudioError::THREAD_ERROR );
\r
4263 SetThreadPriority( ( void* ) stream_.callbackInfo.thread, stream_.callbackInfo.priority );
\r
4264 ResumeThread( ( void* ) stream_.callbackInfo.thread );
\r
4268 //-----------------------------------------------------------------------------
\r
4270 void RtApiWasapi::stopStream( void )
\r
4274 if ( stream_.state == STREAM_STOPPED ) {
\r
4275 errorText_ = "RtApiWasapi::stopStream: The stream is already stopped.";
\r
4276 error( RtAudioError::WARNING );
\r
4280 // inform stream thread by setting stream state to STREAM_STOPPING
\r
4281 stream_.state = STREAM_STOPPING;
\r
4283 // wait until stream thread is stopped
\r
4284 while( stream_.state != STREAM_STOPPED ) {
\r
4288 // Wait for the last buffer to play before stopping.
\r
4289 Sleep( 1000 * stream_.bufferSize / stream_.sampleRate );
\r
4291 // stop capture client if applicable
\r
4292 if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) {
\r
4293 HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop();
\r
4294 if ( FAILED( hr ) ) {
\r
4295 errorText_ = "RtApiWasapi::stopStream: Unable to stop capture stream.";
\r
4296 error( RtAudioError::DRIVER_ERROR );
\r
4301 // stop render client if applicable
\r
4302 if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) {
\r
4303 HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop();
\r
4304 if ( FAILED( hr ) ) {
\r
4305 errorText_ = "RtApiWasapi::stopStream: Unable to stop render stream.";
\r
4306 error( RtAudioError::DRIVER_ERROR );
\r
4311 // close thread handle
\r
4312 if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
\r
4313 errorText_ = "RtApiWasapi::stopStream: Unable to close callback thread.";
\r
4314 error( RtAudioError::THREAD_ERROR );
\r
4318 stream_.callbackInfo.thread = (ThreadHandle) NULL;
\r
4321 //-----------------------------------------------------------------------------
\r
4323 void RtApiWasapi::abortStream( void )
\r
4327 if ( stream_.state == STREAM_STOPPED ) {
\r
4328 errorText_ = "RtApiWasapi::abortStream: The stream is already stopped.";
\r
4329 error( RtAudioError::WARNING );
\r
4333 // inform stream thread by setting stream state to STREAM_STOPPING
\r
4334 stream_.state = STREAM_STOPPING;
\r
4336 // wait until stream thread is stopped
\r
4337 while ( stream_.state != STREAM_STOPPED ) {
\r
4341 // stop capture client if applicable
\r
4342 if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) {
\r
4343 HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop();
\r
4344 if ( FAILED( hr ) ) {
\r
4345 errorText_ = "RtApiWasapi::abortStream: Unable to stop capture stream.";
\r
4346 error( RtAudioError::DRIVER_ERROR );
\r
4351 // stop render client if applicable
\r
4352 if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) {
\r
4353 HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop();
\r
4354 if ( FAILED( hr ) ) {
\r
4355 errorText_ = "RtApiWasapi::abortStream: Unable to stop render stream.";
\r
4356 error( RtAudioError::DRIVER_ERROR );
\r
4361 // close thread handle
\r
4362 if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
\r
4363 errorText_ = "RtApiWasapi::abortStream: Unable to close callback thread.";
\r
4364 error( RtAudioError::THREAD_ERROR );
\r
4368 stream_.callbackInfo.thread = (ThreadHandle) NULL;
\r
4371 //-----------------------------------------------------------------------------
\r
4373 bool RtApiWasapi::probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
\r
4374 unsigned int firstChannel, unsigned int sampleRate,
\r
4375 RtAudioFormat format, unsigned int* bufferSize,
\r
4376 RtAudio::StreamOptions* options )
\r
4378 bool methodResult = FAILURE;
\r
4379 unsigned int captureDeviceCount = 0;
\r
4380 unsigned int renderDeviceCount = 0;
\r
4382 IMMDeviceCollection* captureDevices = NULL;
\r
4383 IMMDeviceCollection* renderDevices = NULL;
\r
4384 IMMDevice* devicePtr = NULL;
\r
4385 WAVEFORMATEX* deviceFormat = NULL;
\r
4386 unsigned int bufferBytes;
\r
4387 stream_.state = STREAM_STOPPED;
\r
4389 // create API Handle if not already created
\r
4390 if ( !stream_.apiHandle )
\r
4391 stream_.apiHandle = ( void* ) new WasapiHandle();
\r
4393 // Count capture devices
\r
4394 errorText_.clear();
\r
4395 RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
\r
4396 HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
\r
4397 if ( FAILED( hr ) ) {
\r
4398 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device collection.";
\r
4402 hr = captureDevices->GetCount( &captureDeviceCount );
\r
4403 if ( FAILED( hr ) ) {
\r
4404 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device count.";
\r
4408 // Count render devices
\r
4409 hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
\r
4410 if ( FAILED( hr ) ) {
\r
4411 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device collection.";
\r
4415 hr = renderDevices->GetCount( &renderDeviceCount );
\r
4416 if ( FAILED( hr ) ) {
\r
4417 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device count.";
\r
4421 // validate device index
\r
4422 if ( device >= captureDeviceCount + renderDeviceCount ) {
\r
4423 errorType = RtAudioError::INVALID_USE;
\r
4424 errorText_ = "RtApiWasapi::probeDeviceOpen: Invalid device index.";
\r
4428 // determine whether index falls within capture or render devices
\r
4429 if ( device >= renderDeviceCount ) {
\r
4430 if ( mode != INPUT ) {
\r
4431 errorType = RtAudioError::INVALID_USE;
\r
4432 errorText_ = "RtApiWasapi::probeDeviceOpen: Capture device selected as output device.";
\r
4436 // retrieve captureAudioClient from devicePtr
\r
4437 IAudioClient*& captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
\r
4439 hr = captureDevices->Item( device - renderDeviceCount, &devicePtr );
\r
4440 if ( FAILED( hr ) ) {
\r
4441 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device handle.";
\r
4445 hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
\r
4446 NULL, ( void** ) &captureAudioClient );
\r
4447 if ( FAILED( hr ) ) {
\r
4448 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client.";
\r
4452 hr = captureAudioClient->GetMixFormat( &deviceFormat );
\r
4453 if ( FAILED( hr ) ) {
\r
4454 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format.";
\r
4458 stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
\r
4459 captureAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
\r
4462 if ( mode != OUTPUT ) {
\r
4463 errorType = RtAudioError::INVALID_USE;
\r
4464 errorText_ = "RtApiWasapi::probeDeviceOpen: Render device selected as input device.";
\r
4468 // retrieve renderAudioClient from devicePtr
\r
4469 IAudioClient*& renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
\r
4471 hr = renderDevices->Item( device, &devicePtr );
\r
4472 if ( FAILED( hr ) ) {
\r
4473 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device handle.";
\r
4477 hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
\r
4478 NULL, ( void** ) &renderAudioClient );
\r
4479 if ( FAILED( hr ) ) {
\r
4480 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client.";
\r
4484 hr = renderAudioClient->GetMixFormat( &deviceFormat );
\r
4485 if ( FAILED( hr ) ) {
\r
4486 errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format.";
\r
4490 stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
\r
4491 renderAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
\r
4494 // fill stream data
\r
4495 if ( ( stream_.mode == OUTPUT && mode == INPUT ) ||
\r
4496 ( stream_.mode == INPUT && mode == OUTPUT ) ) {
\r
4497 stream_.mode = DUPLEX;
\r
4500 stream_.mode = mode;
\r
4503 stream_.device[mode] = device;
\r
4504 stream_.doByteSwap[mode] = false;
\r
4505 stream_.sampleRate = sampleRate;
\r
4506 stream_.bufferSize = *bufferSize;
\r
4507 stream_.nBuffers = 1;
\r
4508 stream_.nUserChannels[mode] = channels;
\r
4509 stream_.channelOffset[mode] = firstChannel;
\r
4510 stream_.userFormat = format;
\r
4511 stream_.deviceFormat[mode] = getDeviceInfo( device ).nativeFormats;
\r
4513 if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
\r
4514 stream_.userInterleaved = false;
\r
4516 stream_.userInterleaved = true;
\r
4517 stream_.deviceInterleaved[mode] = true;
\r
4519 // Set flags for buffer conversion.
\r
4520 stream_.doConvertBuffer[mode] = false;
\r
4521 if ( stream_.userFormat != stream_.deviceFormat[mode] )
\r
4522 stream_.doConvertBuffer[mode] = true;
\r
4523 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
\r
4524 stream_.nUserChannels[mode] > 1 )
\r
4525 stream_.doConvertBuffer[mode] = true;
\r
4527 if ( stream_.doConvertBuffer[mode] )
\r
4528 setConvertInfo( mode, 0 );
\r
4530 // Allocate necessary internal buffers
\r
4531 bufferBytes = stream_.nUserChannels[mode] * stream_.bufferSize * formatBytes( stream_.userFormat );
\r
4533 stream_.userBuffer[mode] = ( char* ) calloc( bufferBytes, 1 );
\r
4534 if ( !stream_.userBuffer[mode] ) {
\r
4535 errorType = RtAudioError::MEMORY_ERROR;
\r
4536 errorText_ = "RtApiWasapi::probeDeviceOpen: Error allocating user buffer memory.";
\r
4540 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME )
\r
4541 stream_.callbackInfo.priority = 15;
\r
4543 stream_.callbackInfo.priority = 0;
\r
4545 ///! TODO: RTAUDIO_MINIMIZE_LATENCY // Provide stream buffers directly to callback
\r
4546 ///! TODO: RTAUDIO_HOG_DEVICE // Exclusive mode
\r
4548 methodResult = SUCCESS;
\r
4552 SAFE_RELEASE( captureDevices );
\r
4553 SAFE_RELEASE( renderDevices );
\r
4554 SAFE_RELEASE( devicePtr );
\r
4555 CoTaskMemFree( deviceFormat );
\r
4557 // if method failed, close the stream
\r
4558 if ( methodResult == FAILURE )
\r
4561 if ( !errorText_.empty() )
\r
4562 error( errorType );
\r
4563 return methodResult;
\r
4566 //=============================================================================
\r
4568 DWORD WINAPI RtApiWasapi::runWasapiThread( void* wasapiPtr )
\r
4571 ( ( RtApiWasapi* ) wasapiPtr )->wasapiThread();
\r
4576 DWORD WINAPI RtApiWasapi::stopWasapiThread( void* wasapiPtr )
\r
4579 ( ( RtApiWasapi* ) wasapiPtr )->stopStream();
\r
4584 DWORD WINAPI RtApiWasapi::abortWasapiThread( void* wasapiPtr )
\r
4587 ( ( RtApiWasapi* ) wasapiPtr )->abortStream();
\r
4592 //-----------------------------------------------------------------------------
\r
4594 void RtApiWasapi::wasapiThread()
\r
4596 // as this is a new thread, we must CoInitialize it
\r
4597 CoInitialize( NULL );
\r
4601 IAudioClient* captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
\r
4602 IAudioClient* renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
\r
4603 IAudioCaptureClient* captureClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureClient;
\r
4604 IAudioRenderClient* renderClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderClient;
\r
4605 HANDLE captureEvent = ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent;
\r
4606 HANDLE renderEvent = ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent;
\r
4608 WAVEFORMATEX* captureFormat = NULL;
\r
4609 WAVEFORMATEX* renderFormat = NULL;
\r
4610 float captureSrRatio = 0.0f;
\r
4611 float renderSrRatio = 0.0f;
\r
4612 WasapiBuffer captureBuffer;
\r
4613 WasapiBuffer renderBuffer;
\r
4615 // declare local stream variables
\r
4616 RtAudioCallback callback = ( RtAudioCallback ) stream_.callbackInfo.callback;
\r
4617 BYTE* streamBuffer = NULL;
\r
4618 unsigned long captureFlags = 0;
\r
4619 unsigned int bufferFrameCount = 0;
\r
4620 unsigned int numFramesPadding = 0;
\r
4621 unsigned int convBufferSize = 0;
\r
4622 bool callbackPushed = false;
\r
4623 bool callbackPulled = false;
\r
4624 bool callbackStopped = false;
\r
4625 int callbackResult = 0;
\r
4627 // convBuffer is used to store converted buffers between WASAPI and the user
\r
4628 char* convBuffer = NULL;
\r
4629 unsigned int deviceBufferSize = 0;
\r
4631 errorText_.clear();
\r
4632 RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
\r
4634 // Attempt to assign "Pro Audio" characteristic to thread
\r
4635 HMODULE AvrtDll = LoadLibrary( "AVRT.dll" );
\r
4637 DWORD taskIndex = 0;
\r
4638 TAvSetMmThreadCharacteristicsPtr AvSetMmThreadCharacteristicsPtr = ( TAvSetMmThreadCharacteristicsPtr ) GetProcAddress( AvrtDll, "AvSetMmThreadCharacteristicsW" );
\r
4639 AvSetMmThreadCharacteristicsPtr( L"Pro Audio", &taskIndex );
\r
4640 FreeLibrary( AvrtDll );
\r
4643 // start capture stream if applicable
\r
4644 if ( captureAudioClient ) {
\r
4645 hr = captureAudioClient->GetMixFormat( &captureFormat );
\r
4646 if ( FAILED( hr ) ) {
\r
4647 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
\r
4651 captureSrRatio = ( ( float ) captureFormat->nSamplesPerSec / stream_.sampleRate );
\r
4653 // initialize capture stream according to desire buffer size
\r
4654 float desiredBufferSize = stream_.bufferSize * captureSrRatio;
\r
4655 REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / captureFormat->nSamplesPerSec );
\r
4657 if ( !captureClient ) {
\r
4658 hr = captureAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
\r
4659 AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
\r
4660 desiredBufferPeriod,
\r
4661 desiredBufferPeriod,
\r
4664 if ( FAILED( hr ) ) {
\r
4665 errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize capture audio client.";
\r
4669 hr = captureAudioClient->GetService( __uuidof( IAudioCaptureClient ),
\r
4670 ( void** ) &captureClient );
\r
4671 if ( FAILED( hr ) ) {
\r
4672 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture client handle.";
\r
4676 // configure captureEvent to trigger on every available capture buffer
\r
4677 captureEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
\r
4678 if ( !captureEvent ) {
\r
4679 errorType = RtAudioError::SYSTEM_ERROR;
\r
4680 errorText_ = "RtApiWasapi::wasapiThread: Unable to create capture event.";
\r
4684 hr = captureAudioClient->SetEventHandle( captureEvent );
\r
4685 if ( FAILED( hr ) ) {
\r
4686 errorText_ = "RtApiWasapi::wasapiThread: Unable to set capture event handle.";
\r
4690 ( ( WasapiHandle* ) stream_.apiHandle )->captureClient = captureClient;
\r
4691 ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent = captureEvent;
\r
4694 unsigned int inBufferSize = 0;
\r
4695 hr = captureAudioClient->GetBufferSize( &inBufferSize );
\r
4696 if ( FAILED( hr ) ) {
\r
4697 errorText_ = "RtApiWasapi::wasapiThread: Unable to get capture buffer size.";
\r
4701 // scale outBufferSize according to stream->user sample rate ratio
\r
4702 unsigned int outBufferSize = ( unsigned int ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT];
\r
4703 inBufferSize *= stream_.nDeviceChannels[INPUT];
\r
4705 // set captureBuffer size
\r
4706 captureBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[INPUT] ) );
\r
4708 // reset the capture stream
\r
4709 hr = captureAudioClient->Reset();
\r
4710 if ( FAILED( hr ) ) {
\r
4711 errorText_ = "RtApiWasapi::wasapiThread: Unable to reset capture stream.";
\r
4715 // start the capture stream
\r
4716 hr = captureAudioClient->Start();
\r
4717 if ( FAILED( hr ) ) {
\r
4718 errorText_ = "RtApiWasapi::wasapiThread: Unable to start capture stream.";
\r
4723 // start render stream if applicable
\r
4724 if ( renderAudioClient ) {
\r
4725 hr = renderAudioClient->GetMixFormat( &renderFormat );
\r
4726 if ( FAILED( hr ) ) {
\r
4727 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
\r
4731 renderSrRatio = ( ( float ) renderFormat->nSamplesPerSec / stream_.sampleRate );
\r
4733 // initialize render stream according to desire buffer size
\r
4734 float desiredBufferSize = stream_.bufferSize * renderSrRatio;
\r
4735 REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / renderFormat->nSamplesPerSec );
\r
4737 if ( !renderClient ) {
\r
4738 hr = renderAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
\r
4739 AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
\r
4740 desiredBufferPeriod,
\r
4741 desiredBufferPeriod,
\r
4744 if ( FAILED( hr ) ) {
\r
4745 errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize render audio client.";
\r
4749 hr = renderAudioClient->GetService( __uuidof( IAudioRenderClient ),
\r
4750 ( void** ) &renderClient );
\r
4751 if ( FAILED( hr ) ) {
\r
4752 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render client handle.";
\r
4756 // configure renderEvent to trigger on every available render buffer
\r
4757 renderEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
\r
4758 if ( !renderEvent ) {
\r
4759 errorType = RtAudioError::SYSTEM_ERROR;
\r
4760 errorText_ = "RtApiWasapi::wasapiThread: Unable to create render event.";
\r
4764 hr = renderAudioClient->SetEventHandle( renderEvent );
\r
4765 if ( FAILED( hr ) ) {
\r
4766 errorText_ = "RtApiWasapi::wasapiThread: Unable to set render event handle.";
\r
4770 ( ( WasapiHandle* ) stream_.apiHandle )->renderClient = renderClient;
\r
4771 ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent = renderEvent;
\r
4774 unsigned int outBufferSize = 0;
\r
4775 hr = renderAudioClient->GetBufferSize( &outBufferSize );
\r
4776 if ( FAILED( hr ) ) {
\r
4777 errorText_ = "RtApiWasapi::wasapiThread: Unable to get render buffer size.";
\r
4781 // scale inBufferSize according to user->stream sample rate ratio
\r
4782 unsigned int inBufferSize = ( unsigned int ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT];
\r
4783 outBufferSize *= stream_.nDeviceChannels[OUTPUT];
\r
4785 // set renderBuffer size
\r
4786 renderBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[OUTPUT] ) );
\r
4788 // reset the render stream
\r
4789 hr = renderAudioClient->Reset();
\r
4790 if ( FAILED( hr ) ) {
\r
4791 errorText_ = "RtApiWasapi::wasapiThread: Unable to reset render stream.";
\r
4795 // start the render stream
\r
4796 hr = renderAudioClient->Start();
\r
4797 if ( FAILED( hr ) ) {
\r
4798 errorText_ = "RtApiWasapi::wasapiThread: Unable to start render stream.";
\r
4803 if ( stream_.mode == INPUT ) {
\r
4804 deviceBufferSize = ( size_t ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
\r
4806 else if ( stream_.mode == OUTPUT ) {
\r
4807 deviceBufferSize = ( size_t ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
\r
4809 else if ( stream_.mode == DUPLEX ) {
\r
4810 deviceBufferSize = std::max( ( size_t ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
\r
4811 ( size_t ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
\r
4814 convBuffer = ( char* ) malloc( deviceBufferSize );
\r
4815 stream_.deviceBuffer = ( char* ) malloc( deviceBufferSize );
\r
4816 if ( !convBuffer || !stream_.deviceBuffer ) {
\r
4817 errorType = RtAudioError::MEMORY_ERROR;
\r
4818 errorText_ = "RtApiWasapi::wasapiThread: Error allocating device buffer memory.";
\r
4822 // stream process loop
\r
4823 while ( stream_.state != STREAM_STOPPING ) {
\r
4824 if ( !callbackPulled ) {
\r
4827 // 1. Pull callback buffer from inputBuffer
\r
4828 // 2. If 1. was successful: Convert callback buffer to user sample rate and channel count
\r
4829 // Convert callback buffer to user format
\r
4831 if ( captureAudioClient ) {
\r
4832 // Pull callback buffer from inputBuffer
\r
4833 callbackPulled = captureBuffer.pullBuffer( convBuffer,
\r
4834 ( unsigned int ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT],
\r
4835 stream_.deviceFormat[INPUT] );
\r
4837 if ( callbackPulled ) {
\r
4838 // Convert callback buffer to user sample rate and channel count
\r
4839 convertBufferWasapi( stream_.deviceBuffer,
\r
4841 stream_.nDeviceChannels[INPUT],
\r
4842 stream_.nUserChannels[INPUT],
\r
4843 captureFormat->nSamplesPerSec,
\r
4844 stream_.sampleRate,
\r
4845 ( unsigned int ) ( stream_.bufferSize * captureSrRatio ),
\r
4847 stream_.deviceFormat[INPUT] );
\r
4849 if ( stream_.doConvertBuffer[INPUT] ) {
\r
4850 // Convert callback buffer to user format
\r
4851 convertBuffer( stream_.userBuffer[INPUT],
\r
4852 stream_.deviceBuffer,
\r
4853 stream_.convertInfo[INPUT] );
\r
4856 // no conversion, simple copy deviceBuffer to userBuffer
\r
4857 memcpy( stream_.userBuffer[INPUT],
\r
4858 stream_.deviceBuffer,
\r
4859 stream_.bufferSize * stream_.nUserChannels[INPUT] * formatBytes( stream_.userFormat ) );
\r
4864 // if there is no capture stream, set callbackPulled flag
\r
4865 callbackPulled = true;
\r
4868 // Execute Callback
\r
4869 // ================
\r
4870 // 1. Execute user callback method
\r
4871 // 2. Handle return value from callback
\r
4873 // if callback has not requested the stream to stop
\r
4874 if ( callbackPulled && !callbackStopped ) {
\r
4875 // Execute user callback method
\r
4876 callbackResult = callback( stream_.userBuffer[OUTPUT],
\r
4877 stream_.userBuffer[INPUT],
\r
4878 stream_.bufferSize,
\r
4880 captureFlags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY ? RTAUDIO_INPUT_OVERFLOW : 0,
\r
4881 stream_.callbackInfo.userData );
\r
4883 // Handle return value from callback
\r
4884 if ( callbackResult == 1 ) {
\r
4885 // instantiate a thread to stop this thread
\r
4886 HANDLE threadHandle = CreateThread( NULL, 0, stopWasapiThread, this, 0, NULL );
\r
4887 if ( !threadHandle ) {
\r
4888 errorType = RtAudioError::THREAD_ERROR;
\r
4889 errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream stop thread.";
\r
4892 else if ( !CloseHandle( threadHandle ) ) {
\r
4893 errorType = RtAudioError::THREAD_ERROR;
\r
4894 errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream stop thread handle.";
\r
4898 callbackStopped = true;
\r
4900 else if ( callbackResult == 2 ) {
\r
4901 // instantiate a thread to stop this thread
\r
4902 HANDLE threadHandle = CreateThread( NULL, 0, abortWasapiThread, this, 0, NULL );
\r
4903 if ( !threadHandle ) {
\r
4904 errorType = RtAudioError::THREAD_ERROR;
\r
4905 errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream abort thread.";
\r
4908 else if ( !CloseHandle( threadHandle ) ) {
\r
4909 errorType = RtAudioError::THREAD_ERROR;
\r
4910 errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream abort thread handle.";
\r
4914 callbackStopped = true;
\r
4919 // Callback Output
\r
4920 // ===============
\r
4921 // 1. Convert callback buffer to stream format
\r
4922 // 2. Convert callback buffer to stream sample rate and channel count
\r
4923 // 3. Push callback buffer into outputBuffer
\r
4925 if ( renderAudioClient && callbackPulled ) {
\r
4926 if ( stream_.doConvertBuffer[OUTPUT] ) {
\r
4927 // Convert callback buffer to stream format
\r
4928 convertBuffer( stream_.deviceBuffer,
\r
4929 stream_.userBuffer[OUTPUT],
\r
4930 stream_.convertInfo[OUTPUT] );
\r
4932 // Convert callback buffer to stream sample rate and channel count
\r
4933 convertBufferWasapi( convBuffer,
\r
4934 stream_.deviceBuffer,
\r
4935 stream_.nUserChannels[OUTPUT],
\r
4936 stream_.nDeviceChannels[OUTPUT],
\r
4937 stream_.sampleRate,
\r
4938 renderFormat->nSamplesPerSec,
\r
4939 stream_.bufferSize,
\r
4941 stream_.deviceFormat[OUTPUT] );
\r
4944 // Convert callback buffer to stream sample rate and channel count
\r
4945 convertBufferWasapi( convBuffer,
\r
4946 stream_.userBuffer[OUTPUT],
\r
4947 stream_.nUserChannels[OUTPUT],
\r
4948 stream_.nDeviceChannels[OUTPUT],
\r
4949 stream_.sampleRate,
\r
4950 renderFormat->nSamplesPerSec,
\r
4951 stream_.bufferSize,
\r
4953 stream_.deviceFormat[OUTPUT] );
\r
4956 // Push callback buffer into outputBuffer
\r
4957 callbackPushed = renderBuffer.pushBuffer( convBuffer,
\r
4958 convBufferSize * stream_.nDeviceChannels[OUTPUT],
\r
4959 stream_.deviceFormat[OUTPUT] );
\r
4964 // 1. Get capture buffer from stream
\r
4965 // 2. Push capture buffer into inputBuffer
\r
4966 // 3. If 2. was successful: Release capture buffer
\r
4968 if ( captureAudioClient ) {
\r
4969 // if the callback input buffer was not pulled from captureBuffer, wait for next capture event
\r
4970 if ( !callbackPulled ) {
\r
4971 WaitForSingleObject( captureEvent, INFINITE );
\r
4974 // Get capture buffer from stream
\r
4975 hr = captureClient->GetBuffer( &streamBuffer,
\r
4976 &bufferFrameCount,
\r
4977 &captureFlags, NULL, NULL );
\r
4978 if ( FAILED( hr ) ) {
\r
4979 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture buffer.";
\r
4983 if ( bufferFrameCount != 0 ) {
\r
4984 // Push capture buffer into inputBuffer
\r
4985 if ( captureBuffer.pushBuffer( ( char* ) streamBuffer,
\r
4986 bufferFrameCount * stream_.nDeviceChannels[INPUT],
\r
4987 stream_.deviceFormat[INPUT] ) )
\r
4989 // Release capture buffer
\r
4990 hr = captureClient->ReleaseBuffer( bufferFrameCount );
\r
4991 if ( FAILED( hr ) ) {
\r
4992 errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
\r
4998 // Inform WASAPI that capture was unsuccessful
\r
4999 hr = captureClient->ReleaseBuffer( 0 );
\r
5000 if ( FAILED( hr ) ) {
\r
5001 errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
\r
5008 // Inform WASAPI that capture was unsuccessful
\r
5009 hr = captureClient->ReleaseBuffer( 0 );
\r
5010 if ( FAILED( hr ) ) {
\r
5011 errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
\r
5019 // 1. Get render buffer from stream
\r
5020 // 2. Pull next buffer from outputBuffer
\r
5021 // 3. If 2. was successful: Fill render buffer with next buffer
\r
5022 // Release render buffer
\r
5024 if ( renderAudioClient ) {
\r
5025 // if the callback output buffer was not pushed to renderBuffer, wait for next render event
\r
5026 if ( callbackPulled && !callbackPushed ) {
\r
5027 WaitForSingleObject( renderEvent, INFINITE );
\r
5030 // Get render buffer from stream
\r
5031 hr = renderAudioClient->GetBufferSize( &bufferFrameCount );
\r
5032 if ( FAILED( hr ) ) {
\r
5033 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer size.";
\r
5037 hr = renderAudioClient->GetCurrentPadding( &numFramesPadding );
\r
5038 if ( FAILED( hr ) ) {
\r
5039 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer padding.";
\r
5043 bufferFrameCount -= numFramesPadding;
\r
5045 if ( bufferFrameCount != 0 ) {
\r
5046 hr = renderClient->GetBuffer( bufferFrameCount, &streamBuffer );
\r
5047 if ( FAILED( hr ) ) {
\r
5048 errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer.";
\r
5052 // Pull next buffer from outputBuffer
\r
5053 // Fill render buffer with next buffer
\r
5054 if ( renderBuffer.pullBuffer( ( char* ) streamBuffer,
\r
5055 bufferFrameCount * stream_.nDeviceChannels[OUTPUT],
\r
5056 stream_.deviceFormat[OUTPUT] ) )
\r
5058 // Release render buffer
\r
5059 hr = renderClient->ReleaseBuffer( bufferFrameCount, 0 );
\r
5060 if ( FAILED( hr ) ) {
\r
5061 errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
\r
5067 // Inform WASAPI that render was unsuccessful
\r
5068 hr = renderClient->ReleaseBuffer( 0, 0 );
\r
5069 if ( FAILED( hr ) ) {
\r
5070 errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
\r
5077 // Inform WASAPI that render was unsuccessful
\r
5078 hr = renderClient->ReleaseBuffer( 0, 0 );
\r
5079 if ( FAILED( hr ) ) {
\r
5080 errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
\r
5086 // if the callback buffer was pushed renderBuffer reset callbackPulled flag
\r
5087 if ( callbackPushed ) {
\r
5088 callbackPulled = false;
\r
5091 // tick stream time
\r
5092 RtApi::tickStreamTime();
\r
5097 CoTaskMemFree( captureFormat );
\r
5098 CoTaskMemFree( renderFormat );
\r
5100 //delete convBuffer;
\r
5101 free ( convBuffer );
\r
5105 // update stream state
\r
5106 stream_.state = STREAM_STOPPED;
\r
5108 if ( errorText_.empty() )
\r
5111 error( errorType );
\r
5114 //******************** End of __WINDOWS_WASAPI__ *********************//
\r
5118 #if defined(__WINDOWS_DS__) // Windows DirectSound API
\r
5120 // Modified by Robin Davies, October 2005
\r
5121 // - Improvements to DirectX pointer chasing.
\r
5122 // - Bug fix for non-power-of-two Asio granularity used by Edirol PCR-A30.
\r
5123 // - Auto-call CoInitialize for DSOUND and ASIO platforms.
\r
5124 // Various revisions for RtAudio 4.0 by Gary Scavone, April 2007
\r
5125 // Changed device query structure for RtAudio 4.0.7, January 2010
\r
5127 #include <dsound.h>
\r
5128 #include <assert.h>
\r
5129 #include <algorithm>
\r
5131 #if defined(__MINGW32__)
\r
5132 // missing from latest mingw winapi
\r
5133 #define WAVE_FORMAT_96M08 0x00010000 /* 96 kHz, Mono, 8-bit */
\r
5134 #define WAVE_FORMAT_96S08 0x00020000 /* 96 kHz, Stereo, 8-bit */
\r
5135 #define WAVE_FORMAT_96M16 0x00040000 /* 96 kHz, Mono, 16-bit */
\r
5136 #define WAVE_FORMAT_96S16 0x00080000 /* 96 kHz, Stereo, 16-bit */
\r
5139 #define MINIMUM_DEVICE_BUFFER_SIZE 32768
\r
5141 #ifdef _MSC_VER // if Microsoft Visual C++
\r
5142 #pragma comment( lib, "winmm.lib" ) // then, auto-link winmm.lib. Otherwise, it has to be added manually.
\r
5145 static inline DWORD dsPointerBetween( DWORD pointer, DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize )
\r
5147 if ( pointer > bufferSize ) pointer -= bufferSize;
\r
5148 if ( laterPointer < earlierPointer ) laterPointer += bufferSize;
\r
5149 if ( pointer < earlierPointer ) pointer += bufferSize;
\r
5150 return pointer >= earlierPointer && pointer < laterPointer;
\r
5153 // A structure to hold various information related to the DirectSound
\r
5154 // API implementation.
\r
5156 unsigned int drainCounter; // Tracks callback counts when draining
\r
5157 bool internalDrain; // Indicates if stop is initiated from callback or not.
\r
5161 UINT bufferPointer[2];
\r
5162 DWORD dsBufferSize[2];
\r
5163 DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by.
\r
5167 :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; }
\r
5170 // Declarations for utility functions, callbacks, and structures
\r
5171 // specific to the DirectSound implementation.
\r
5172 static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
\r
5173 LPCTSTR description,
\r
5175 LPVOID lpContext );
\r
5177 static const char* getErrorString( int code );
\r
5179 static unsigned __stdcall callbackHandler( void *ptr );
\r
5188 : found(false) { validId[0] = false; validId[1] = false; }
\r
5191 struct DsProbeData {
\r
5193 std::vector<struct DsDevice>* dsDevices;
\r
5196 RtApiDs :: RtApiDs()
\r
5198 // Dsound will run both-threaded. If CoInitialize fails, then just
\r
5199 // accept whatever the mainline chose for a threading model.
\r
5200 coInitialized_ = false;
\r
5201 HRESULT hr = CoInitialize( NULL );
\r
5202 if ( !FAILED( hr ) ) coInitialized_ = true;
\r
5205 RtApiDs :: ~RtApiDs()
\r
5207 if ( coInitialized_ ) CoUninitialize(); // balanced call.
\r
5208 if ( stream_.state != STREAM_CLOSED ) closeStream();
\r
5211 // The DirectSound default output is always the first device.
\r
5212 unsigned int RtApiDs :: getDefaultOutputDevice( void )
\r
5217 // The DirectSound default input is always the first input device,
\r
5218 // which is the first capture device enumerated.
\r
5219 unsigned int RtApiDs :: getDefaultInputDevice( void )
\r
5224 unsigned int RtApiDs :: getDeviceCount( void )
\r
5226 // Set query flag for previously found devices to false, so that we
\r
5227 // can check for any devices that have disappeared.
\r
5228 for ( unsigned int i=0; i<dsDevices.size(); i++ )
\r
5229 dsDevices[i].found = false;
\r
5231 // Query DirectSound devices.
\r
5232 struct DsProbeData probeInfo;
\r
5233 probeInfo.isInput = false;
\r
5234 probeInfo.dsDevices = &dsDevices;
\r
5235 HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );
\r
5236 if ( FAILED( result ) ) {
\r
5237 errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!";
\r
5238 errorText_ = errorStream_.str();
\r
5239 error( RtAudioError::WARNING );
\r
5242 // Query DirectSoundCapture devices.
\r
5243 probeInfo.isInput = true;
\r
5244 result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );
\r
5245 if ( FAILED( result ) ) {
\r
5246 errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!";
\r
5247 errorText_ = errorStream_.str();
\r
5248 error( RtAudioError::WARNING );
\r
5251 // Clean out any devices that may have disappeared.
\r
5252 std::vector< int > indices;
\r
5253 for ( unsigned int i=0; i<dsDevices.size(); i++ )
\r
5254 if ( dsDevices[i].found == false ) indices.push_back( i );
\r
5255 //unsigned int nErased = 0;
\r
5256 for ( unsigned int i=0; i<indices.size(); i++ )
\r
5257 dsDevices.erase( dsDevices.begin()+indices[i] );
\r
5258 //dsDevices.erase( dsDevices.begin()-nErased++ );
\r
5260 return static_cast<unsigned int>(dsDevices.size());
\r
5263 RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device )
\r
5265 RtAudio::DeviceInfo info;
\r
5266 info.probed = false;
\r
5268 if ( dsDevices.size() == 0 ) {
\r
5269 // Force a query of all devices
\r
5271 if ( dsDevices.size() == 0 ) {
\r
5272 errorText_ = "RtApiDs::getDeviceInfo: no devices found!";
\r
5273 error( RtAudioError::INVALID_USE );
\r
5278 if ( device >= dsDevices.size() ) {
\r
5279 errorText_ = "RtApiDs::getDeviceInfo: device ID is invalid!";
\r
5280 error( RtAudioError::INVALID_USE );
\r
5285 if ( dsDevices[ device ].validId[0] == false ) goto probeInput;
\r
5287 LPDIRECTSOUND output;
\r
5289 result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );
\r
5290 if ( FAILED( result ) ) {
\r
5291 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!";
\r
5292 errorText_ = errorStream_.str();
\r
5293 error( RtAudioError::WARNING );
\r
5297 outCaps.dwSize = sizeof( outCaps );
\r
5298 result = output->GetCaps( &outCaps );
\r
5299 if ( FAILED( result ) ) {
\r
5300 output->Release();
\r
5301 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting capabilities!";
\r
5302 errorText_ = errorStream_.str();
\r
5303 error( RtAudioError::WARNING );
\r
5307 // Get output channel information.
\r
5308 info.outputChannels = ( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1;
\r
5310 // Get sample rate information.
\r
5311 info.sampleRates.clear();
\r
5312 for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
\r
5313 if ( SAMPLE_RATES[k] >= (unsigned int) outCaps.dwMinSecondarySampleRate &&
\r
5314 SAMPLE_RATES[k] <= (unsigned int) outCaps.dwMaxSecondarySampleRate )
\r
5315 info.sampleRates.push_back( SAMPLE_RATES[k] );
\r
5318 // Get format information.
\r
5319 if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT ) info.nativeFormats |= RTAUDIO_SINT16;
\r
5320 if ( outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) info.nativeFormats |= RTAUDIO_SINT8;
\r
5322 output->Release();
\r
5324 if ( getDefaultOutputDevice() == device )
\r
5325 info.isDefaultOutput = true;
\r
5327 if ( dsDevices[ device ].validId[1] == false ) {
\r
5328 info.name = dsDevices[ device ].name;
\r
5329 info.probed = true;
\r
5335 LPDIRECTSOUNDCAPTURE input;
\r
5336 result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );
\r
5337 if ( FAILED( result ) ) {
\r
5338 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";
\r
5339 errorText_ = errorStream_.str();
\r
5340 error( RtAudioError::WARNING );
\r
5345 inCaps.dwSize = sizeof( inCaps );
\r
5346 result = input->GetCaps( &inCaps );
\r
5347 if ( FAILED( result ) ) {
\r
5349 errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsDevices[ device ].name << ")!";
\r
5350 errorText_ = errorStream_.str();
\r
5351 error( RtAudioError::WARNING );
\r
5355 // Get input channel information.
\r
5356 info.inputChannels = inCaps.dwChannels;
\r
5358 // Get sample rate and format information.
\r
5359 std::vector<unsigned int> rates;
\r
5360 if ( inCaps.dwChannels >= 2 ) {
\r
5361 if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.nativeFormats |= RTAUDIO_SINT16;
\r
5362 if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.nativeFormats |= RTAUDIO_SINT16;
\r
5363 if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.nativeFormats |= RTAUDIO_SINT16;
\r
5364 if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.nativeFormats |= RTAUDIO_SINT16;
\r
5365 if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.nativeFormats |= RTAUDIO_SINT8;
\r
5366 if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.nativeFormats |= RTAUDIO_SINT8;
\r
5367 if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.nativeFormats |= RTAUDIO_SINT8;
\r
5368 if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.nativeFormats |= RTAUDIO_SINT8;
\r
5370 if ( info.nativeFormats & RTAUDIO_SINT16 ) {
\r
5371 if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) rates.push_back( 11025 );
\r
5372 if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) rates.push_back( 22050 );
\r
5373 if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) rates.push_back( 44100 );
\r
5374 if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) rates.push_back( 96000 );
\r
5376 else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
\r
5377 if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) rates.push_back( 11025 );
\r
5378 if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) rates.push_back( 22050 );
\r
5379 if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) rates.push_back( 44100 );
\r
5380 if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) rates.push_back( 96000 );
\r
5383 else if ( inCaps.dwChannels == 1 ) {
\r
5384 if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.nativeFormats |= RTAUDIO_SINT16;
\r
5385 if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.nativeFormats |= RTAUDIO_SINT16;
\r
5386 if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.nativeFormats |= RTAUDIO_SINT16;
\r
5387 if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.nativeFormats |= RTAUDIO_SINT16;
\r
5388 if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.nativeFormats |= RTAUDIO_SINT8;
\r
5389 if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.nativeFormats |= RTAUDIO_SINT8;
\r
5390 if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.nativeFormats |= RTAUDIO_SINT8;
\r
5391 if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.nativeFormats |= RTAUDIO_SINT8;
\r
5393 if ( info.nativeFormats & RTAUDIO_SINT16 ) {
\r
5394 if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) rates.push_back( 11025 );
\r
5395 if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) rates.push_back( 22050 );
\r
5396 if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) rates.push_back( 44100 );
\r
5397 if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) rates.push_back( 96000 );
\r
5399 else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
\r
5400 if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) rates.push_back( 11025 );
\r
5401 if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) rates.push_back( 22050 );
\r
5402 if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) rates.push_back( 44100 );
\r
5403 if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) rates.push_back( 96000 );
\r
5406 else info.inputChannels = 0; // technically, this would be an error
\r
5410 if ( info.inputChannels == 0 ) return info;
\r
5412 // Copy the supported rates to the info structure but avoid duplication.
\r
5414 for ( unsigned int i=0; i<rates.size(); i++ ) {
\r
5416 for ( unsigned int j=0; j<info.sampleRates.size(); j++ ) {
\r
5417 if ( rates[i] == info.sampleRates[j] ) {
\r
5422 if ( found == false ) info.sampleRates.push_back( rates[i] );
\r
5424 std::sort( info.sampleRates.begin(), info.sampleRates.end() );
\r
5426 // If device opens for both playback and capture, we determine the channels.
\r
5427 if ( info.outputChannels > 0 && info.inputChannels > 0 )
\r
5428 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
\r
5430 if ( device == 0 ) info.isDefaultInput = true;
\r
5432 // Copy name and return.
\r
5433 info.name = dsDevices[ device ].name;
\r
5434 info.probed = true;
\r
5438 bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
\r
5439 unsigned int firstChannel, unsigned int sampleRate,
\r
5440 RtAudioFormat format, unsigned int *bufferSize,
\r
5441 RtAudio::StreamOptions *options )
\r
5443 if ( channels + firstChannel > 2 ) {
\r
5444 errorText_ = "RtApiDs::probeDeviceOpen: DirectSound does not support more than 2 channels per device.";
\r
5448 size_t nDevices = dsDevices.size();
\r
5449 if ( nDevices == 0 ) {
\r
5450 // This should not happen because a check is made before this function is called.
\r
5451 errorText_ = "RtApiDs::probeDeviceOpen: no devices found!";
\r
5455 if ( device >= nDevices ) {
\r
5456 // This should not happen because a check is made before this function is called.
\r
5457 errorText_ = "RtApiDs::probeDeviceOpen: device ID is invalid!";
\r
5461 if ( mode == OUTPUT ) {
\r
5462 if ( dsDevices[ device ].validId[0] == false ) {
\r
5463 errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support output!";
\r
5464 errorText_ = errorStream_.str();
\r
5468 else { // mode == INPUT
\r
5469 if ( dsDevices[ device ].validId[1] == false ) {
\r
5470 errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support input!";
\r
5471 errorText_ = errorStream_.str();
\r
5476 // According to a note in PortAudio, using GetDesktopWindow()
\r
5477 // instead of GetForegroundWindow() is supposed to avoid problems
\r
5478 // that occur when the application's window is not the foreground
\r
5479 // window. Also, if the application window closes before the
\r
5480 // DirectSound buffer, DirectSound can crash. In the past, I had
\r
5481 // problems when using GetDesktopWindow() but it seems fine now
\r
5482 // (January 2010). I'll leave it commented here.
\r
5483 // HWND hWnd = GetForegroundWindow();
\r
5484 HWND hWnd = GetDesktopWindow();
\r
5486 // Check the numberOfBuffers parameter and limit the lowest value to
\r
5487 // two. This is a judgement call and a value of two is probably too
\r
5488 // low for capture, but it should work for playback.
\r
5490 if ( options ) nBuffers = options->numberOfBuffers;
\r
5491 if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) nBuffers = 2;
\r
5492 if ( nBuffers < 2 ) nBuffers = 3;
\r
5494 // Check the lower range of the user-specified buffer size and set
\r
5495 // (arbitrarily) to a lower bound of 32.
\r
5496 if ( *bufferSize < 32 ) *bufferSize = 32;
\r
5498 // Create the wave format structure. The data format setting will
\r
5499 // be determined later.
\r
5500 WAVEFORMATEX waveFormat;
\r
5501 ZeroMemory( &waveFormat, sizeof(WAVEFORMATEX) );
\r
5502 waveFormat.wFormatTag = WAVE_FORMAT_PCM;
\r
5503 waveFormat.nChannels = channels + firstChannel;
\r
5504 waveFormat.nSamplesPerSec = (unsigned long) sampleRate;
\r
5506 // Determine the device buffer size. By default, we'll use the value
\r
5507 // defined above (32K), but we will grow it to make allowances for
\r
5508 // very large software buffer sizes.
\r
5509 DWORD dsBufferSize = MINIMUM_DEVICE_BUFFER_SIZE;
\r
5510 DWORD dsPointerLeadTime = 0;
\r
5512 void *ohandle = 0, *bhandle = 0;
\r
5514 if ( mode == OUTPUT ) {
\r
5516 LPDIRECTSOUND output;
\r
5517 result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );
\r
5518 if ( FAILED( result ) ) {
\r
5519 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!";
\r
5520 errorText_ = errorStream_.str();
\r
5525 outCaps.dwSize = sizeof( outCaps );
\r
5526 result = output->GetCaps( &outCaps );
\r
5527 if ( FAILED( result ) ) {
\r
5528 output->Release();
\r
5529 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting capabilities (" << dsDevices[ device ].name << ")!";
\r
5530 errorText_ = errorStream_.str();
\r
5534 // Check channel information.
\r
5535 if ( channels + firstChannel == 2 && !( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ) {
\r
5536 errorStream_ << "RtApiDs::getDeviceInfo: the output device (" << dsDevices[ device ].name << ") does not support stereo playback.";
\r
5537 errorText_ = errorStream_.str();
\r
5541 // Check format information. Use 16-bit format unless not
\r
5542 // supported or user requests 8-bit.
\r
5543 if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT &&
\r
5544 !( format == RTAUDIO_SINT8 && outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) ) {
\r
5545 waveFormat.wBitsPerSample = 16;
\r
5546 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
\r
5549 waveFormat.wBitsPerSample = 8;
\r
5550 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
\r
5552 stream_.userFormat = format;
\r
5554 // Update wave format structure and buffer information.
\r
5555 waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
\r
5556 waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
\r
5557 dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
\r
5559 // If the user wants an even bigger buffer, increase the device buffer size accordingly.
\r
5560 while ( dsPointerLeadTime * 2U > dsBufferSize )
\r
5561 dsBufferSize *= 2;
\r
5563 // Set cooperative level to DSSCL_EXCLUSIVE ... sound stops when window focus changes.
\r
5564 // result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE );
\r
5565 // Set cooperative level to DSSCL_PRIORITY ... sound remains when window focus changes.
\r
5566 result = output->SetCooperativeLevel( hWnd, DSSCL_PRIORITY );
\r
5567 if ( FAILED( result ) ) {
\r
5568 output->Release();
\r
5569 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting cooperative level (" << dsDevices[ device ].name << ")!";
\r
5570 errorText_ = errorStream_.str();
\r
5574 // Even though we will write to the secondary buffer, we need to
\r
5575 // access the primary buffer to set the correct output format
\r
5576 // (since the default is 8-bit, 22 kHz!). Setup the DS primary
\r
5577 // buffer description.
\r
5578 DSBUFFERDESC bufferDescription;
\r
5579 ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
\r
5580 bufferDescription.dwSize = sizeof( DSBUFFERDESC );
\r
5581 bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER;
\r
5583 // Obtain the primary buffer
\r
5584 LPDIRECTSOUNDBUFFER buffer;
\r
5585 result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
\r
5586 if ( FAILED( result ) ) {
\r
5587 output->Release();
\r
5588 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") accessing primary buffer (" << dsDevices[ device ].name << ")!";
\r
5589 errorText_ = errorStream_.str();
\r
5593 // Set the primary DS buffer sound format.
\r
5594 result = buffer->SetFormat( &waveFormat );
\r
5595 if ( FAILED( result ) ) {
\r
5596 output->Release();
\r
5597 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting primary buffer format (" << dsDevices[ device ].name << ")!";
\r
5598 errorText_ = errorStream_.str();
\r
5602 // Setup the secondary DS buffer description.
\r
5603 ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
\r
5604 bufferDescription.dwSize = sizeof( DSBUFFERDESC );
\r
5605 bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
\r
5606 DSBCAPS_GLOBALFOCUS |
\r
5607 DSBCAPS_GETCURRENTPOSITION2 |
\r
5608 DSBCAPS_LOCHARDWARE ); // Force hardware mixing
\r
5609 bufferDescription.dwBufferBytes = dsBufferSize;
\r
5610 bufferDescription.lpwfxFormat = &waveFormat;
\r
5612 // Try to create the secondary DS buffer. If that doesn't work,
\r
5613 // try to use software mixing. Otherwise, there's a problem.
\r
5614 result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
\r
5615 if ( FAILED( result ) ) {
\r
5616 bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
\r
5617 DSBCAPS_GLOBALFOCUS |
\r
5618 DSBCAPS_GETCURRENTPOSITION2 |
\r
5619 DSBCAPS_LOCSOFTWARE ); // Force software mixing
\r
5620 result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
\r
5621 if ( FAILED( result ) ) {
\r
5622 output->Release();
\r
5623 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating secondary buffer (" << dsDevices[ device ].name << ")!";
\r
5624 errorText_ = errorStream_.str();
\r
5629 // Get the buffer size ... might be different from what we specified.
\r
5631 dsbcaps.dwSize = sizeof( DSBCAPS );
\r
5632 result = buffer->GetCaps( &dsbcaps );
\r
5633 if ( FAILED( result ) ) {
\r
5634 output->Release();
\r
5635 buffer->Release();
\r
5636 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";
\r
5637 errorText_ = errorStream_.str();
\r
5641 dsBufferSize = dsbcaps.dwBufferBytes;
\r
5643 // Lock the DS buffer
\r
5646 result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );
\r
5647 if ( FAILED( result ) ) {
\r
5648 output->Release();
\r
5649 buffer->Release();
\r
5650 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking buffer (" << dsDevices[ device ].name << ")!";
\r
5651 errorText_ = errorStream_.str();
\r
5655 // Zero the DS buffer
\r
5656 ZeroMemory( audioPtr, dataLen );
\r
5658 // Unlock the DS buffer
\r
5659 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
\r
5660 if ( FAILED( result ) ) {
\r
5661 output->Release();
\r
5662 buffer->Release();
\r
5663 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking buffer (" << dsDevices[ device ].name << ")!";
\r
5664 errorText_ = errorStream_.str();
\r
5668 ohandle = (void *) output;
\r
5669 bhandle = (void *) buffer;
\r
5672 if ( mode == INPUT ) {
\r
5674 LPDIRECTSOUNDCAPTURE input;
\r
5675 result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );
\r
5676 if ( FAILED( result ) ) {
\r
5677 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";
\r
5678 errorText_ = errorStream_.str();
\r
5683 inCaps.dwSize = sizeof( inCaps );
\r
5684 result = input->GetCaps( &inCaps );
\r
5685 if ( FAILED( result ) ) {
\r
5687 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsDevices[ device ].name << ")!";
\r
5688 errorText_ = errorStream_.str();
\r
5692 // Check channel information.
\r
5693 if ( inCaps.dwChannels < channels + firstChannel ) {
\r
5694 errorText_ = "RtApiDs::getDeviceInfo: the input device does not support requested input channels.";
\r
5698 // Check format information. Use 16-bit format unless user
\r
5699 // requests 8-bit.
\r
5700 DWORD deviceFormats;
\r
5701 if ( channels + firstChannel == 2 ) {
\r
5702 deviceFormats = WAVE_FORMAT_1S08 | WAVE_FORMAT_2S08 | WAVE_FORMAT_4S08 | WAVE_FORMAT_96S08;
\r
5703 if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
\r
5704 waveFormat.wBitsPerSample = 8;
\r
5705 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
\r
5707 else { // assume 16-bit is supported
\r
5708 waveFormat.wBitsPerSample = 16;
\r
5709 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
\r
5712 else { // channel == 1
\r
5713 deviceFormats = WAVE_FORMAT_1M08 | WAVE_FORMAT_2M08 | WAVE_FORMAT_4M08 | WAVE_FORMAT_96M08;
\r
5714 if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
\r
5715 waveFormat.wBitsPerSample = 8;
\r
5716 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
\r
5718 else { // assume 16-bit is supported
\r
5719 waveFormat.wBitsPerSample = 16;
\r
5720 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
\r
5723 stream_.userFormat = format;
\r
5725 // Update wave format structure and buffer information.
\r
5726 waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
\r
5727 waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
\r
5728 dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
\r
5730 // If the user wants an even bigger buffer, increase the device buffer size accordingly.
\r
5731 while ( dsPointerLeadTime * 2U > dsBufferSize )
\r
5732 dsBufferSize *= 2;
\r
5734 // Setup the secondary DS buffer description.
\r
5735 DSCBUFFERDESC bufferDescription;
\r
5736 ZeroMemory( &bufferDescription, sizeof( DSCBUFFERDESC ) );
\r
5737 bufferDescription.dwSize = sizeof( DSCBUFFERDESC );
\r
5738 bufferDescription.dwFlags = 0;
\r
5739 bufferDescription.dwReserved = 0;
\r
5740 bufferDescription.dwBufferBytes = dsBufferSize;
\r
5741 bufferDescription.lpwfxFormat = &waveFormat;
\r
5743 // Create the capture buffer.
\r
5744 LPDIRECTSOUNDCAPTUREBUFFER buffer;
\r
5745 result = input->CreateCaptureBuffer( &bufferDescription, &buffer, NULL );
\r
5746 if ( FAILED( result ) ) {
\r
5748 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating input buffer (" << dsDevices[ device ].name << ")!";
\r
5749 errorText_ = errorStream_.str();
\r
5753 // Get the buffer size ... might be different from what we specified.
\r
5754 DSCBCAPS dscbcaps;
\r
5755 dscbcaps.dwSize = sizeof( DSCBCAPS );
\r
5756 result = buffer->GetCaps( &dscbcaps );
\r
5757 if ( FAILED( result ) ) {
\r
5759 buffer->Release();
\r
5760 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";
\r
5761 errorText_ = errorStream_.str();
\r
5765 dsBufferSize = dscbcaps.dwBufferBytes;
\r
5767 // NOTE: We could have a problem here if this is a duplex stream
\r
5768 // and the play and capture hardware buffer sizes are different
\r
5769 // (I'm actually not sure if that is a problem or not).
\r
5770 // Currently, we are not verifying that.
\r
5772 // Lock the capture buffer
\r
5775 result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );
\r
5776 if ( FAILED( result ) ) {
\r
5778 buffer->Release();
\r
5779 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking input buffer (" << dsDevices[ device ].name << ")!";
\r
5780 errorText_ = errorStream_.str();
\r
5784 // Zero the buffer
\r
5785 ZeroMemory( audioPtr, dataLen );
\r
5787 // Unlock the buffer
\r
5788 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
\r
5789 if ( FAILED( result ) ) {
\r
5791 buffer->Release();
\r
5792 errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking input buffer (" << dsDevices[ device ].name << ")!";
\r
5793 errorText_ = errorStream_.str();
\r
5797 ohandle = (void *) input;
\r
5798 bhandle = (void *) buffer;
\r
5801 // Set various stream parameters
\r
5802 DsHandle *handle = 0;
\r
5803 stream_.nDeviceChannels[mode] = channels + firstChannel;
\r
5804 stream_.nUserChannels[mode] = channels;
\r
5805 stream_.bufferSize = *bufferSize;
\r
5806 stream_.channelOffset[mode] = firstChannel;
\r
5807 stream_.deviceInterleaved[mode] = true;
\r
5808 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
\r
5809 else stream_.userInterleaved = true;
\r
5811 // Set flag for buffer conversion
\r
5812 stream_.doConvertBuffer[mode] = false;
\r
5813 if (stream_.nUserChannels[mode] != stream_.nDeviceChannels[mode])
\r
5814 stream_.doConvertBuffer[mode] = true;
\r
5815 if (stream_.userFormat != stream_.deviceFormat[mode])
\r
5816 stream_.doConvertBuffer[mode] = true;
\r
5817 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
\r
5818 stream_.nUserChannels[mode] > 1 )
\r
5819 stream_.doConvertBuffer[mode] = true;
\r
5821 // Allocate necessary internal buffers
\r
5822 long bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
\r
5823 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
\r
5824 if ( stream_.userBuffer[mode] == NULL ) {
\r
5825 errorText_ = "RtApiDs::probeDeviceOpen: error allocating user buffer memory.";
\r
5829 if ( stream_.doConvertBuffer[mode] ) {
\r
5831 bool makeBuffer = true;
\r
5832 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
\r
5833 if ( mode == INPUT ) {
\r
5834 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
\r
5835 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
\r
5836 if ( bufferBytes <= (long) bytesOut ) makeBuffer = false;
\r
5840 if ( makeBuffer ) {
\r
5841 bufferBytes *= *bufferSize;
\r
5842 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
\r
5843 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
\r
5844 if ( stream_.deviceBuffer == NULL ) {
\r
5845 errorText_ = "RtApiDs::probeDeviceOpen: error allocating device buffer memory.";
\r
5851 // Allocate our DsHandle structures for the stream.
\r
5852 if ( stream_.apiHandle == 0 ) {
\r
5854 handle = new DsHandle;
\r
5856 catch ( std::bad_alloc& ) {
\r
5857 errorText_ = "RtApiDs::probeDeviceOpen: error allocating AsioHandle memory.";
\r
5861 // Create a manual-reset event.
\r
5862 handle->condition = CreateEvent( NULL, // no security
\r
5863 TRUE, // manual-reset
\r
5864 FALSE, // non-signaled initially
\r
5865 NULL ); // unnamed
\r
5866 stream_.apiHandle = (void *) handle;
\r
5869 handle = (DsHandle *) stream_.apiHandle;
\r
5870 handle->id[mode] = ohandle;
\r
5871 handle->buffer[mode] = bhandle;
\r
5872 handle->dsBufferSize[mode] = dsBufferSize;
\r
5873 handle->dsPointerLeadTime[mode] = dsPointerLeadTime;
\r
5875 stream_.device[mode] = device;
\r
5876 stream_.state = STREAM_STOPPED;
\r
5877 if ( stream_.mode == OUTPUT && mode == INPUT )
\r
5878 // We had already set up an output stream.
\r
5879 stream_.mode = DUPLEX;
\r
5881 stream_.mode = mode;
\r
5882 stream_.nBuffers = nBuffers;
\r
5883 stream_.sampleRate = sampleRate;
\r
5885 // Setup the buffer conversion information structure.
\r
5886 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
\r
5888 // Setup the callback thread.
\r
5889 if ( stream_.callbackInfo.isRunning == false ) {
\r
5890 unsigned threadId;
\r
5891 stream_.callbackInfo.isRunning = true;
\r
5892 stream_.callbackInfo.object = (void *) this;
\r
5893 stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &callbackHandler,
\r
5894 &stream_.callbackInfo, 0, &threadId );
\r
5895 if ( stream_.callbackInfo.thread == 0 ) {
\r
5896 errorText_ = "RtApiDs::probeDeviceOpen: error creating callback thread!";
\r
5900 // Boost DS thread priority
\r
5901 SetThreadPriority( (HANDLE) stream_.callbackInfo.thread, THREAD_PRIORITY_HIGHEST );
\r
5907 if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
\r
5908 LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
\r
5909 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
\r
5910 if ( buffer ) buffer->Release();
\r
5911 object->Release();
\r
5913 if ( handle->buffer[1] ) {
\r
5914 LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
\r
5915 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
\r
5916 if ( buffer ) buffer->Release();
\r
5917 object->Release();
\r
5919 CloseHandle( handle->condition );
\r
5921 stream_.apiHandle = 0;
\r
5924 for ( int i=0; i<2; i++ ) {
\r
5925 if ( stream_.userBuffer[i] ) {
\r
5926 free( stream_.userBuffer[i] );
\r
5927 stream_.userBuffer[i] = 0;
\r
5931 if ( stream_.deviceBuffer ) {
\r
5932 free( stream_.deviceBuffer );
\r
5933 stream_.deviceBuffer = 0;
\r
5936 stream_.state = STREAM_CLOSED;
\r
5940 void RtApiDs :: closeStream()
\r
5942 if ( stream_.state == STREAM_CLOSED ) {
\r
5943 errorText_ = "RtApiDs::closeStream(): no open stream to close!";
\r
5944 error( RtAudioError::WARNING );
\r
5948 // Stop the callback thread.
\r
5949 stream_.callbackInfo.isRunning = false;
\r
5950 WaitForSingleObject( (HANDLE) stream_.callbackInfo.thread, INFINITE );
\r
5951 CloseHandle( (HANDLE) stream_.callbackInfo.thread );
\r
5953 DsHandle *handle = (DsHandle *) stream_.apiHandle;
\r
5955 if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
\r
5956 LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
\r
5957 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
\r
5960 buffer->Release();
\r
5962 object->Release();
\r
5964 if ( handle->buffer[1] ) {
\r
5965 LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
\r
5966 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
\r
5969 buffer->Release();
\r
5971 object->Release();
\r
5973 CloseHandle( handle->condition );
\r
5975 stream_.apiHandle = 0;
\r
5978 for ( int i=0; i<2; i++ ) {
\r
5979 if ( stream_.userBuffer[i] ) {
\r
5980 free( stream_.userBuffer[i] );
\r
5981 stream_.userBuffer[i] = 0;
\r
5985 if ( stream_.deviceBuffer ) {
\r
5986 free( stream_.deviceBuffer );
\r
5987 stream_.deviceBuffer = 0;
\r
5990 stream_.mode = UNINITIALIZED;
\r
5991 stream_.state = STREAM_CLOSED;
\r
5994 void RtApiDs :: startStream()
\r
5997 if ( stream_.state == STREAM_RUNNING ) {
\r
5998 errorText_ = "RtApiDs::startStream(): the stream is already running!";
\r
5999 error( RtAudioError::WARNING );
\r
6003 DsHandle *handle = (DsHandle *) stream_.apiHandle;
\r
6005 // Increase scheduler frequency on lesser windows (a side-effect of
\r
6006 // increasing timer accuracy). On greater windows (Win2K or later),
\r
6007 // this is already in effect.
\r
6008 timeBeginPeriod( 1 );
\r
6010 buffersRolling = false;
\r
6011 duplexPrerollBytes = 0;
\r
6013 if ( stream_.mode == DUPLEX ) {
\r
6014 // 0.5 seconds of silence in DUPLEX mode while the devices spin up and synchronize.
\r
6015 duplexPrerollBytes = (int) ( 0.5 * stream_.sampleRate * formatBytes( stream_.deviceFormat[1] ) * stream_.nDeviceChannels[1] );
\r
6018 HRESULT result = 0;
\r
6019 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
\r
6021 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
\r
6022 result = buffer->Play( 0, 0, DSBPLAY_LOOPING );
\r
6023 if ( FAILED( result ) ) {
\r
6024 errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting output buffer!";
\r
6025 errorText_ = errorStream_.str();
\r
6030 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
\r
6032 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
\r
6033 result = buffer->Start( DSCBSTART_LOOPING );
\r
6034 if ( FAILED( result ) ) {
\r
6035 errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting input buffer!";
\r
6036 errorText_ = errorStream_.str();
\r
6041 handle->drainCounter = 0;
\r
6042 handle->internalDrain = false;
\r
6043 ResetEvent( handle->condition );
\r
6044 stream_.state = STREAM_RUNNING;
\r
6047 if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );
\r
6050 void RtApiDs :: stopStream()
\r
6053 if ( stream_.state == STREAM_STOPPED ) {
\r
6054 errorText_ = "RtApiDs::stopStream(): the stream is already stopped!";
\r
6055 error( RtAudioError::WARNING );
\r
6059 HRESULT result = 0;
\r
6062 DsHandle *handle = (DsHandle *) stream_.apiHandle;
\r
6063 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
\r
6064 if ( handle->drainCounter == 0 ) {
\r
6065 handle->drainCounter = 2;
\r
6066 WaitForSingleObject( handle->condition, INFINITE ); // block until signaled
\r
6069 stream_.state = STREAM_STOPPED;
\r
6071 // Stop the buffer and clear memory
\r
6072 LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
\r
6073 result = buffer->Stop();
\r
6074 if ( FAILED( result ) ) {
\r
6075 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping output buffer!";
\r
6076 errorText_ = errorStream_.str();
\r
6080 // Lock the buffer and clear it so that if we start to play again,
\r
6081 // we won't have old data playing.
\r
6082 result = buffer->Lock( 0, handle->dsBufferSize[0], &audioPtr, &dataLen, NULL, NULL, 0 );
\r
6083 if ( FAILED( result ) ) {
\r
6084 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking output buffer!";
\r
6085 errorText_ = errorStream_.str();
\r
6089 // Zero the DS buffer
\r
6090 ZeroMemory( audioPtr, dataLen );
\r
6092 // Unlock the DS buffer
\r
6093 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
\r
6094 if ( FAILED( result ) ) {
\r
6095 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking output buffer!";
\r
6096 errorText_ = errorStream_.str();
\r
6100 // If we start playing again, we must begin at beginning of buffer.
\r
6101 handle->bufferPointer[0] = 0;
\r
6104 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
\r
6105 LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
\r
6109 stream_.state = STREAM_STOPPED;
\r
6111 result = buffer->Stop();
\r
6112 if ( FAILED( result ) ) {
\r
6113 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping input buffer!";
\r
6114 errorText_ = errorStream_.str();
\r
6118 // Lock the buffer and clear it so that if we start to play again,
\r
6119 // we won't have old data playing.
\r
6120 result = buffer->Lock( 0, handle->dsBufferSize[1], &audioPtr, &dataLen, NULL, NULL, 0 );
\r
6121 if ( FAILED( result ) ) {
\r
6122 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking input buffer!";
\r
6123 errorText_ = errorStream_.str();
\r
6127 // Zero the DS buffer
\r
6128 ZeroMemory( audioPtr, dataLen );
\r
6130 // Unlock the DS buffer
\r
6131 result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
\r
6132 if ( FAILED( result ) ) {
\r
6133 errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking input buffer!";
\r
6134 errorText_ = errorStream_.str();
\r
6138 // If we start recording again, we must begin at beginning of buffer.
\r
6139 handle->bufferPointer[1] = 0;
\r
6143 timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows.
\r
6144 if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );
\r
6147 void RtApiDs :: abortStream()
\r
6150 if ( stream_.state == STREAM_STOPPED ) {
\r
6151 errorText_ = "RtApiDs::abortStream(): the stream is already stopped!";
\r
6152 error( RtAudioError::WARNING );
\r
6156 DsHandle *handle = (DsHandle *) stream_.apiHandle;
\r
6157 handle->drainCounter = 2;
\r
6162 void RtApiDs :: callbackEvent()
\r
6164 if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) {
\r
6165 Sleep( 50 ); // sleep 50 milliseconds
\r
6169 if ( stream_.state == STREAM_CLOSED ) {
\r
6170 errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!";
\r
6171 error( RtAudioError::WARNING );
\r
6175 CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
\r
6176 DsHandle *handle = (DsHandle *) stream_.apiHandle;
\r
6178 // Check if we were draining the stream and signal is finished.
\r
6179 if ( handle->drainCounter > stream_.nBuffers + 2 ) {
\r
6181 stream_.state = STREAM_STOPPING;
\r
6182 if ( handle->internalDrain == false )
\r
6183 SetEvent( handle->condition );
\r
6189 // Invoke user callback to get fresh output data UNLESS we are
\r
6190 // draining stream.
\r
6191 if ( handle->drainCounter == 0 ) {
\r
6192 RtAudioCallback callback = (RtAudioCallback) info->callback;
\r
6193 double streamTime = getStreamTime();
\r
6194 RtAudioStreamStatus status = 0;
\r
6195 if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
\r
6196 status |= RTAUDIO_OUTPUT_UNDERFLOW;
\r
6197 handle->xrun[0] = false;
\r
6199 if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
\r
6200 status |= RTAUDIO_INPUT_OVERFLOW;
\r
6201 handle->xrun[1] = false;
\r
6203 int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
\r
6204 stream_.bufferSize, streamTime, status, info->userData );
\r
6205 if ( cbReturnValue == 2 ) {
\r
6206 stream_.state = STREAM_STOPPING;
\r
6207 handle->drainCounter = 2;
\r
6211 else if ( cbReturnValue == 1 ) {
\r
6212 handle->drainCounter = 1;
\r
6213 handle->internalDrain = true;
\r
6218 DWORD currentWritePointer, safeWritePointer;
\r
6219 DWORD currentReadPointer, safeReadPointer;
\r
6220 UINT nextWritePointer;
\r
6222 LPVOID buffer1 = NULL;
\r
6223 LPVOID buffer2 = NULL;
\r
6224 DWORD bufferSize1 = 0;
\r
6225 DWORD bufferSize2 = 0;
\r
6230 if ( buffersRolling == false ) {
\r
6231 if ( stream_.mode == DUPLEX ) {
\r
6232 //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
\r
6234 // It takes a while for the devices to get rolling. As a result,
\r
6235 // there's no guarantee that the capture and write device pointers
\r
6236 // will move in lockstep. Wait here for both devices to start
\r
6237 // rolling, and then set our buffer pointers accordingly.
\r
6238 // e.g. Crystal Drivers: the capture buffer starts up 5700 to 9600
\r
6239 // bytes later than the write buffer.
\r
6241 // Stub: a serious risk of having a pre-emptive scheduling round
\r
6242 // take place between the two GetCurrentPosition calls... but I'm
\r
6243 // really not sure how to solve the problem. Temporarily boost to
\r
6244 // Realtime priority, maybe; but I'm not sure what priority the
\r
6245 // DirectSound service threads run at. We *should* be roughly
\r
6246 // within a ms or so of correct.
\r
6248 LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
\r
6249 LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
\r
6251 DWORD startSafeWritePointer, startSafeReadPointer;
\r
6253 result = dsWriteBuffer->GetCurrentPosition( NULL, &startSafeWritePointer );
\r
6254 if ( FAILED( result ) ) {
\r
6255 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
\r
6256 errorText_ = errorStream_.str();
\r
6257 error( RtAudioError::SYSTEM_ERROR );
\r
6260 result = dsCaptureBuffer->GetCurrentPosition( NULL, &startSafeReadPointer );
\r
6261 if ( FAILED( result ) ) {
\r
6262 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
\r
6263 errorText_ = errorStream_.str();
\r
6264 error( RtAudioError::SYSTEM_ERROR );
\r
6268 result = dsWriteBuffer->GetCurrentPosition( NULL, &safeWritePointer );
\r
6269 if ( FAILED( result ) ) {
\r
6270 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
\r
6271 errorText_ = errorStream_.str();
\r
6272 error( RtAudioError::SYSTEM_ERROR );
\r
6275 result = dsCaptureBuffer->GetCurrentPosition( NULL, &safeReadPointer );
\r
6276 if ( FAILED( result ) ) {
\r
6277 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
\r
6278 errorText_ = errorStream_.str();
\r
6279 error( RtAudioError::SYSTEM_ERROR );
\r
6282 if ( safeWritePointer != startSafeWritePointer && safeReadPointer != startSafeReadPointer ) break;
\r
6286 //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
\r
6288 handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];
\r
6289 if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];
\r
6290 handle->bufferPointer[1] = safeReadPointer;
\r
6292 else if ( stream_.mode == OUTPUT ) {
\r
6294 // Set the proper nextWritePosition after initial startup.
\r
6295 LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
\r
6296 result = dsWriteBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer );
\r
6297 if ( FAILED( result ) ) {
\r
6298 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
\r
6299 errorText_ = errorStream_.str();
\r
6300 error( RtAudioError::SYSTEM_ERROR );
\r
6303 handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];
\r
6304 if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];
\r
6307 buffersRolling = true;
\r
6310 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
\r
6312 LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
\r
6314 if ( handle->drainCounter > 1 ) { // write zeros to the output stream
\r
6315 bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
\r
6316 bufferBytes *= formatBytes( stream_.userFormat );
\r
6317 memset( stream_.userBuffer[0], 0, bufferBytes );
\r
6320 // Setup parameters and do buffer conversion if necessary.
\r
6321 if ( stream_.doConvertBuffer[0] ) {
\r
6322 buffer = stream_.deviceBuffer;
\r
6323 convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
\r
6324 bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[0];
\r
6325 bufferBytes *= formatBytes( stream_.deviceFormat[0] );
\r
6328 buffer = stream_.userBuffer[0];
\r
6329 bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
\r
6330 bufferBytes *= formatBytes( stream_.userFormat );
\r
6333 // No byte swapping necessary in DirectSound implementation.
\r
6335 // Ahhh ... windoze. 16-bit data is signed but 8-bit data is
\r
6336 // unsigned. So, we need to convert our signed 8-bit data here to
\r
6338 if ( stream_.deviceFormat[0] == RTAUDIO_SINT8 )
\r
6339 for ( int i=0; i<bufferBytes; i++ ) buffer[i] = (unsigned char) ( buffer[i] + 128 );
\r
6341 DWORD dsBufferSize = handle->dsBufferSize[0];
\r
6342 nextWritePointer = handle->bufferPointer[0];
\r
6344 DWORD endWrite, leadPointer;
\r
6346 // Find out where the read and "safe write" pointers are.
\r
6347 result = dsBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer );
\r
6348 if ( FAILED( result ) ) {
\r
6349 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
\r
6350 errorText_ = errorStream_.str();
\r
6351 error( RtAudioError::SYSTEM_ERROR );
\r
6355 // We will copy our output buffer into the region between
\r
6356 // safeWritePointer and leadPointer. If leadPointer is not
\r
6357 // beyond the next endWrite position, wait until it is.
\r
6358 leadPointer = safeWritePointer + handle->dsPointerLeadTime[0];
\r
6359 //std::cout << "safeWritePointer = " << safeWritePointer << ", leadPointer = " << leadPointer << ", nextWritePointer = " << nextWritePointer << std::endl;
\r
6360 if ( leadPointer > dsBufferSize ) leadPointer -= dsBufferSize;
\r
6361 if ( leadPointer < nextWritePointer ) leadPointer += dsBufferSize; // unwrap offset
\r
6362 endWrite = nextWritePointer + bufferBytes;
\r
6364 // Check whether the entire write region is behind the play pointer.
\r
6365 if ( leadPointer >= endWrite ) break;
\r
6367 // If we are here, then we must wait until the leadPointer advances
\r
6368 // beyond the end of our next write region. We use the
\r
6369 // Sleep() function to suspend operation until that happens.
\r
6370 double millis = ( endWrite - leadPointer ) * 1000.0;
\r
6371 millis /= ( formatBytes( stream_.deviceFormat[0]) * stream_.nDeviceChannels[0] * stream_.sampleRate);
\r
6372 if ( millis < 1.0 ) millis = 1.0;
\r
6373 Sleep( (DWORD) millis );
\r
6376 if ( dsPointerBetween( nextWritePointer, safeWritePointer, currentWritePointer, dsBufferSize )
\r
6377 || dsPointerBetween( endWrite, safeWritePointer, currentWritePointer, dsBufferSize ) ) {
\r
6378 // We've strayed into the forbidden zone ... resync the read pointer.
\r
6379 handle->xrun[0] = true;
\r
6380 nextWritePointer = safeWritePointer + handle->dsPointerLeadTime[0] - bufferBytes;
\r
6381 if ( nextWritePointer >= dsBufferSize ) nextWritePointer -= dsBufferSize;
\r
6382 handle->bufferPointer[0] = nextWritePointer;
\r
6383 endWrite = nextWritePointer + bufferBytes;
\r
6386 // Lock free space in the buffer
\r
6387 result = dsBuffer->Lock( nextWritePointer, bufferBytes, &buffer1,
\r
6388 &bufferSize1, &buffer2, &bufferSize2, 0 );
\r
6389 if ( FAILED( result ) ) {
\r
6390 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking buffer during playback!";
\r
6391 errorText_ = errorStream_.str();
\r
6392 error( RtAudioError::SYSTEM_ERROR );
\r
6396 // Copy our buffer into the DS buffer
\r
6397 CopyMemory( buffer1, buffer, bufferSize1 );
\r
6398 if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 );
\r
6400 // Update our buffer offset and unlock sound buffer
\r
6401 dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
\r
6402 if ( FAILED( result ) ) {
\r
6403 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!";
\r
6404 errorText_ = errorStream_.str();
\r
6405 error( RtAudioError::SYSTEM_ERROR );
\r
6408 nextWritePointer = ( nextWritePointer + bufferSize1 + bufferSize2 ) % dsBufferSize;
\r
6409 handle->bufferPointer[0] = nextWritePointer;
\r
6411 if ( handle->drainCounter ) {
\r
6412 handle->drainCounter++;
\r
6417 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
\r
6419 // Setup parameters.
\r
6420 if ( stream_.doConvertBuffer[1] ) {
\r
6421 buffer = stream_.deviceBuffer;
\r
6422 bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[1];
\r
6423 bufferBytes *= formatBytes( stream_.deviceFormat[1] );
\r
6426 buffer = stream_.userBuffer[1];
\r
6427 bufferBytes = stream_.bufferSize * stream_.nUserChannels[1];
\r
6428 bufferBytes *= formatBytes( stream_.userFormat );
\r
6431 LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
\r
6432 long nextReadPointer = handle->bufferPointer[1];
\r
6433 DWORD dsBufferSize = handle->dsBufferSize[1];
\r
6435 // Find out where the write and "safe read" pointers are.
\r
6436 result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer );
\r
6437 if ( FAILED( result ) ) {
\r
6438 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
\r
6439 errorText_ = errorStream_.str();
\r
6440 error( RtAudioError::SYSTEM_ERROR );
\r
6444 if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset
\r
6445 DWORD endRead = nextReadPointer + bufferBytes;
\r
6447 // Handling depends on whether we are INPUT or DUPLEX.
\r
6448 // If we're in INPUT mode then waiting is a good thing. If we're in DUPLEX mode,
\r
6449 // then a wait here will drag the write pointers into the forbidden zone.
\r
6451 // In DUPLEX mode, rather than wait, we will back off the read pointer until
\r
6452 // it's in a safe position. This causes dropouts, but it seems to be the only
\r
6453 // practical way to sync up the read and write pointers reliably, given the
\r
6454 // the very complex relationship between phase and increment of the read and write
\r
6457 // In order to minimize audible dropouts in DUPLEX mode, we will
\r
6458 // provide a pre-roll period of 0.5 seconds in which we return
\r
6459 // zeros from the read buffer while the pointers sync up.
\r
6461 if ( stream_.mode == DUPLEX ) {
\r
6462 if ( safeReadPointer < endRead ) {
\r
6463 if ( duplexPrerollBytes <= 0 ) {
\r
6464 // Pre-roll time over. Be more agressive.
\r
6465 int adjustment = endRead-safeReadPointer;
\r
6467 handle->xrun[1] = true;
\r
6469 // - large adjustments: we've probably run out of CPU cycles, so just resync exactly,
\r
6470 // and perform fine adjustments later.
\r
6471 // - small adjustments: back off by twice as much.
\r
6472 if ( adjustment >= 2*bufferBytes )
\r
6473 nextReadPointer = safeReadPointer-2*bufferBytes;
\r
6475 nextReadPointer = safeReadPointer-bufferBytes-adjustment;
\r
6477 if ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;
\r
6481 // In pre=roll time. Just do it.
\r
6482 nextReadPointer = safeReadPointer - bufferBytes;
\r
6483 while ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;
\r
6485 endRead = nextReadPointer + bufferBytes;
\r
6488 else { // mode == INPUT
\r
6489 while ( safeReadPointer < endRead && stream_.callbackInfo.isRunning ) {
\r
6490 // See comments for playback.
\r
6491 double millis = (endRead - safeReadPointer) * 1000.0;
\r
6492 millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate);
\r
6493 if ( millis < 1.0 ) millis = 1.0;
\r
6494 Sleep( (DWORD) millis );
\r
6496 // Wake up and find out where we are now.
\r
6497 result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer );
\r
6498 if ( FAILED( result ) ) {
\r
6499 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
\r
6500 errorText_ = errorStream_.str();
\r
6501 error( RtAudioError::SYSTEM_ERROR );
\r
6505 if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset
\r
6509 // Lock free space in the buffer
\r
6510 result = dsBuffer->Lock( nextReadPointer, bufferBytes, &buffer1,
\r
6511 &bufferSize1, &buffer2, &bufferSize2, 0 );
\r
6512 if ( FAILED( result ) ) {
\r
6513 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking capture buffer!";
\r
6514 errorText_ = errorStream_.str();
\r
6515 error( RtAudioError::SYSTEM_ERROR );
\r
6519 if ( duplexPrerollBytes <= 0 ) {
\r
6520 // Copy our buffer into the DS buffer
\r
6521 CopyMemory( buffer, buffer1, bufferSize1 );
\r
6522 if ( buffer2 != NULL ) CopyMemory( buffer+bufferSize1, buffer2, bufferSize2 );
\r
6525 memset( buffer, 0, bufferSize1 );
\r
6526 if ( buffer2 != NULL ) memset( buffer + bufferSize1, 0, bufferSize2 );
\r
6527 duplexPrerollBytes -= bufferSize1 + bufferSize2;
\r
6530 // Update our buffer offset and unlock sound buffer
\r
6531 nextReadPointer = ( nextReadPointer + bufferSize1 + bufferSize2 ) % dsBufferSize;
\r
6532 dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
\r
6533 if ( FAILED( result ) ) {
\r
6534 errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking capture buffer!";
\r
6535 errorText_ = errorStream_.str();
\r
6536 error( RtAudioError::SYSTEM_ERROR );
\r
6539 handle->bufferPointer[1] = nextReadPointer;
\r
6541 // No byte swapping necessary in DirectSound implementation.
\r
6543 // If necessary, convert 8-bit data from unsigned to signed.
\r
6544 if ( stream_.deviceFormat[1] == RTAUDIO_SINT8 )
\r
6545 for ( int j=0; j<bufferBytes; j++ ) buffer[j] = (signed char) ( buffer[j] - 128 );
\r
6547 // Do buffer conversion if necessary.
\r
6548 if ( stream_.doConvertBuffer[1] )
\r
6549 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
\r
6553 RtApi::tickStreamTime();
\r
6556 // Definitions for utility functions and callbacks
\r
6557 // specific to the DirectSound implementation.
\r
6559 static unsigned __stdcall callbackHandler( void *ptr )
\r
6561 CallbackInfo *info = (CallbackInfo *) ptr;
\r
6562 RtApiDs *object = (RtApiDs *) info->object;
\r
6563 bool* isRunning = &info->isRunning;
\r
6565 while ( *isRunning == true ) {
\r
6566 object->callbackEvent();
\r
6569 _endthreadex( 0 );
\r
6573 #include "tchar.h"
\r
6575 static std::string convertTChar( LPCTSTR name )
\r
6577 #if defined( UNICODE ) || defined( _UNICODE )
\r
6578 int length = WideCharToMultiByte(CP_UTF8, 0, name, -1, NULL, 0, NULL, NULL);
\r
6579 std::string s( length-1, '\0' );
\r
6580 WideCharToMultiByte(CP_UTF8, 0, name, -1, &s[0], length, NULL, NULL);
\r
6582 std::string s( name );
\r
6588 static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
\r
6589 LPCTSTR description,
\r
6590 LPCTSTR /*module*/,
\r
6591 LPVOID lpContext )
\r
6593 struct DsProbeData& probeInfo = *(struct DsProbeData*) lpContext;
\r
6594 std::vector<struct DsDevice>& dsDevices = *probeInfo.dsDevices;
\r
6597 bool validDevice = false;
\r
6598 if ( probeInfo.isInput == true ) {
\r
6600 LPDIRECTSOUNDCAPTURE object;
\r
6602 hr = DirectSoundCaptureCreate( lpguid, &object, NULL );
\r
6603 if ( hr != DS_OK ) return TRUE;
\r
6605 caps.dwSize = sizeof(caps);
\r
6606 hr = object->GetCaps( &caps );
\r
6607 if ( hr == DS_OK ) {
\r
6608 if ( caps.dwChannels > 0 && caps.dwFormats > 0 )
\r
6609 validDevice = true;
\r
6611 object->Release();
\r
6615 LPDIRECTSOUND object;
\r
6616 hr = DirectSoundCreate( lpguid, &object, NULL );
\r
6617 if ( hr != DS_OK ) return TRUE;
\r
6619 caps.dwSize = sizeof(caps);
\r
6620 hr = object->GetCaps( &caps );
\r
6621 if ( hr == DS_OK ) {
\r
6622 if ( caps.dwFlags & DSCAPS_PRIMARYMONO || caps.dwFlags & DSCAPS_PRIMARYSTEREO )
\r
6623 validDevice = true;
\r
6625 object->Release();
\r
6628 // If good device, then save its name and guid.
\r
6629 std::string name = convertTChar( description );
\r
6630 //if ( name == "Primary Sound Driver" || name == "Primary Sound Capture Driver" )
\r
6631 if ( lpguid == NULL )
\r
6632 name = "Default Device";
\r
6633 if ( validDevice ) {
\r
6634 for ( unsigned int i=0; i<dsDevices.size(); i++ ) {
\r
6635 if ( dsDevices[i].name == name ) {
\r
6636 dsDevices[i].found = true;
\r
6637 if ( probeInfo.isInput ) {
\r
6638 dsDevices[i].id[1] = lpguid;
\r
6639 dsDevices[i].validId[1] = true;
\r
6642 dsDevices[i].id[0] = lpguid;
\r
6643 dsDevices[i].validId[0] = true;
\r
6650 device.name = name;
\r
6651 device.found = true;
\r
6652 if ( probeInfo.isInput ) {
\r
6653 device.id[1] = lpguid;
\r
6654 device.validId[1] = true;
\r
6657 device.id[0] = lpguid;
\r
6658 device.validId[0] = true;
\r
6660 dsDevices.push_back( device );
\r
6666 static const char* getErrorString( int code )
\r
6670 case DSERR_ALLOCATED:
\r
6671 return "Already allocated";
\r
6673 case DSERR_CONTROLUNAVAIL:
\r
6674 return "Control unavailable";
\r
6676 case DSERR_INVALIDPARAM:
\r
6677 return "Invalid parameter";
\r
6679 case DSERR_INVALIDCALL:
\r
6680 return "Invalid call";
\r
6682 case DSERR_GENERIC:
\r
6683 return "Generic error";
\r
6685 case DSERR_PRIOLEVELNEEDED:
\r
6686 return "Priority level needed";
\r
6688 case DSERR_OUTOFMEMORY:
\r
6689 return "Out of memory";
\r
6691 case DSERR_BADFORMAT:
\r
6692 return "The sample rate or the channel format is not supported";
\r
6694 case DSERR_UNSUPPORTED:
\r
6695 return "Not supported";
\r
6697 case DSERR_NODRIVER:
\r
6698 return "No driver";
\r
6700 case DSERR_ALREADYINITIALIZED:
\r
6701 return "Already initialized";
\r
6703 case DSERR_NOAGGREGATION:
\r
6704 return "No aggregation";
\r
6706 case DSERR_BUFFERLOST:
\r
6707 return "Buffer lost";
\r
6709 case DSERR_OTHERAPPHASPRIO:
\r
6710 return "Another application already has priority";
\r
6712 case DSERR_UNINITIALIZED:
\r
6713 return "Uninitialized";
\r
6716 return "DirectSound unknown error";
\r
6719 //******************** End of __WINDOWS_DS__ *********************//
\r
6723 #if defined(__LINUX_ALSA__)
\r
6725 #include <alsa/asoundlib.h>
\r
6726 #include <unistd.h>
\r
6728 // A structure to hold various information related to the ALSA API
\r
6729 // implementation.
\r
6730 struct AlsaHandle {
\r
6731 snd_pcm_t *handles[2];
\r
6732 bool synchronized;
\r
6734 pthread_cond_t runnable_cv;
\r
6738 :synchronized(false), runnable(false) { xrun[0] = false; xrun[1] = false; }
\r
6741 static void *alsaCallbackHandler( void * ptr );
\r
6743 RtApiAlsa :: RtApiAlsa()
\r
6745 // Nothing to do here.
\r
6748 RtApiAlsa :: ~RtApiAlsa()
\r
6750 if ( stream_.state != STREAM_CLOSED ) closeStream();
\r
6753 unsigned int RtApiAlsa :: getDeviceCount( void )
\r
6755 unsigned nDevices = 0;
\r
6756 int result, subdevice, card;
\r
6758 snd_ctl_t *handle;
\r
6760 // Count cards and devices
\r
6762 snd_card_next( &card );
\r
6763 while ( card >= 0 ) {
\r
6764 sprintf( name, "hw:%d", card );
\r
6765 result = snd_ctl_open( &handle, name, 0 );
\r
6766 if ( result < 0 ) {
\r
6767 errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << ".";
\r
6768 errorText_ = errorStream_.str();
\r
6769 error( RtAudioError::WARNING );
\r
6774 result = snd_ctl_pcm_next_device( handle, &subdevice );
\r
6775 if ( result < 0 ) {
\r
6776 errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
\r
6777 errorText_ = errorStream_.str();
\r
6778 error( RtAudioError::WARNING );
\r
6781 if ( subdevice < 0 )
\r
6786 snd_ctl_close( handle );
\r
6787 snd_card_next( &card );
\r
6790 result = snd_ctl_open( &handle, "default", 0 );
\r
6791 if (result == 0) {
\r
6793 snd_ctl_close( handle );
\r
6799 RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device )
\r
6801 RtAudio::DeviceInfo info;
\r
6802 info.probed = false;
\r
6804 unsigned nDevices = 0;
\r
6805 int result, subdevice, card;
\r
6807 snd_ctl_t *chandle;
\r
6809 // Count cards and devices
\r
6811 snd_card_next( &card );
\r
6812 while ( card >= 0 ) {
\r
6813 sprintf( name, "hw:%d", card );
\r
6814 result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
\r
6815 if ( result < 0 ) {
\r
6816 errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << ".";
\r
6817 errorText_ = errorStream_.str();
\r
6818 error( RtAudioError::WARNING );
\r
6823 result = snd_ctl_pcm_next_device( chandle, &subdevice );
\r
6824 if ( result < 0 ) {
\r
6825 errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
\r
6826 errorText_ = errorStream_.str();
\r
6827 error( RtAudioError::WARNING );
\r
6830 if ( subdevice < 0 ) break;
\r
6831 if ( nDevices == device ) {
\r
6832 sprintf( name, "hw:%d,%d", card, subdevice );
\r
6838 snd_ctl_close( chandle );
\r
6839 snd_card_next( &card );
\r
6842 result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
\r
6843 if ( result == 0 ) {
\r
6844 if ( nDevices == device ) {
\r
6845 strcpy( name, "default" );
\r
6851 if ( nDevices == 0 ) {
\r
6852 errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!";
\r
6853 error( RtAudioError::INVALID_USE );
\r
6857 if ( device >= nDevices ) {
\r
6858 errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!";
\r
6859 error( RtAudioError::INVALID_USE );
\r
6865 // If a stream is already open, we cannot probe the stream devices.
\r
6866 // Thus, use the saved results.
\r
6867 if ( stream_.state != STREAM_CLOSED &&
\r
6868 ( stream_.device[0] == device || stream_.device[1] == device ) ) {
\r
6869 snd_ctl_close( chandle );
\r
6870 if ( device >= devices_.size() ) {
\r
6871 errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened.";
\r
6872 error( RtAudioError::WARNING );
\r
6875 return devices_[ device ];
\r
6878 int openMode = SND_PCM_ASYNC;
\r
6879 snd_pcm_stream_t stream;
\r
6880 snd_pcm_info_t *pcminfo;
\r
6881 snd_pcm_info_alloca( &pcminfo );
\r
6882 snd_pcm_t *phandle;
\r
6883 snd_pcm_hw_params_t *params;
\r
6884 snd_pcm_hw_params_alloca( ¶ms );
\r
6886 // First try for playback unless default device (which has subdev -1)
\r
6887 stream = SND_PCM_STREAM_PLAYBACK;
\r
6888 snd_pcm_info_set_stream( pcminfo, stream );
\r
6889 if ( subdevice != -1 ) {
\r
6890 snd_pcm_info_set_device( pcminfo, subdevice );
\r
6891 snd_pcm_info_set_subdevice( pcminfo, 0 );
\r
6893 result = snd_ctl_pcm_info( chandle, pcminfo );
\r
6894 if ( result < 0 ) {
\r
6895 // Device probably doesn't support playback.
\r
6896 goto captureProbe;
\r
6900 result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK );
\r
6901 if ( result < 0 ) {
\r
6902 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
\r
6903 errorText_ = errorStream_.str();
\r
6904 error( RtAudioError::WARNING );
\r
6905 goto captureProbe;
\r
6908 // The device is open ... fill the parameter structure.
\r
6909 result = snd_pcm_hw_params_any( phandle, params );
\r
6910 if ( result < 0 ) {
\r
6911 snd_pcm_close( phandle );
\r
6912 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
\r
6913 errorText_ = errorStream_.str();
\r
6914 error( RtAudioError::WARNING );
\r
6915 goto captureProbe;
\r
6918 // Get output channel information.
\r
6919 unsigned int value;
\r
6920 result = snd_pcm_hw_params_get_channels_max( params, &value );
\r
6921 if ( result < 0 ) {
\r
6922 snd_pcm_close( phandle );
\r
6923 errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << ".";
\r
6924 errorText_ = errorStream_.str();
\r
6925 error( RtAudioError::WARNING );
\r
6926 goto captureProbe;
\r
6928 info.outputChannels = value;
\r
6929 snd_pcm_close( phandle );
\r
6932 stream = SND_PCM_STREAM_CAPTURE;
\r
6933 snd_pcm_info_set_stream( pcminfo, stream );
\r
6935 // Now try for capture unless default device (with subdev = -1)
\r
6936 if ( subdevice != -1 ) {
\r
6937 result = snd_ctl_pcm_info( chandle, pcminfo );
\r
6938 snd_ctl_close( chandle );
\r
6939 if ( result < 0 ) {
\r
6940 // Device probably doesn't support capture.
\r
6941 if ( info.outputChannels == 0 ) return info;
\r
6942 goto probeParameters;
\r
6946 snd_ctl_close( chandle );
\r
6948 result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
\r
6949 if ( result < 0 ) {
\r
6950 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
\r
6951 errorText_ = errorStream_.str();
\r
6952 error( RtAudioError::WARNING );
\r
6953 if ( info.outputChannels == 0 ) return info;
\r
6954 goto probeParameters;
\r
6957 // The device is open ... fill the parameter structure.
\r
6958 result = snd_pcm_hw_params_any( phandle, params );
\r
6959 if ( result < 0 ) {
\r
6960 snd_pcm_close( phandle );
\r
6961 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
\r
6962 errorText_ = errorStream_.str();
\r
6963 error( RtAudioError::WARNING );
\r
6964 if ( info.outputChannels == 0 ) return info;
\r
6965 goto probeParameters;
\r
6968 result = snd_pcm_hw_params_get_channels_max( params, &value );
\r
6969 if ( result < 0 ) {
\r
6970 snd_pcm_close( phandle );
\r
6971 errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << ".";
\r
6972 errorText_ = errorStream_.str();
\r
6973 error( RtAudioError::WARNING );
\r
6974 if ( info.outputChannels == 0 ) return info;
\r
6975 goto probeParameters;
\r
6977 info.inputChannels = value;
\r
6978 snd_pcm_close( phandle );
\r
6980 // If device opens for both playback and capture, we determine the channels.
\r
6981 if ( info.outputChannels > 0 && info.inputChannels > 0 )
\r
6982 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
\r
6984 // ALSA doesn't provide default devices so we'll use the first available one.
\r
6985 if ( device == 0 && info.outputChannels > 0 )
\r
6986 info.isDefaultOutput = true;
\r
6987 if ( device == 0 && info.inputChannels > 0 )
\r
6988 info.isDefaultInput = true;
\r
6991 // At this point, we just need to figure out the supported data
\r
6992 // formats and sample rates. We'll proceed by opening the device in
\r
6993 // the direction with the maximum number of channels, or playback if
\r
6994 // they are equal. This might limit our sample rate options, but so
\r
6997 if ( info.outputChannels >= info.inputChannels )
\r
6998 stream = SND_PCM_STREAM_PLAYBACK;
\r
7000 stream = SND_PCM_STREAM_CAPTURE;
\r
7001 snd_pcm_info_set_stream( pcminfo, stream );
\r
7003 result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
\r
7004 if ( result < 0 ) {
\r
7005 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
\r
7006 errorText_ = errorStream_.str();
\r
7007 error( RtAudioError::WARNING );
\r
7011 // The device is open ... fill the parameter structure.
\r
7012 result = snd_pcm_hw_params_any( phandle, params );
\r
7013 if ( result < 0 ) {
\r
7014 snd_pcm_close( phandle );
\r
7015 errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
\r
7016 errorText_ = errorStream_.str();
\r
7017 error( RtAudioError::WARNING );
\r
7021 // Test our discrete set of sample rate values.
\r
7022 info.sampleRates.clear();
\r
7023 for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {
\r
7024 if ( snd_pcm_hw_params_test_rate( phandle, params, SAMPLE_RATES[i], 0 ) == 0 )
\r
7025 info.sampleRates.push_back( SAMPLE_RATES[i] );
\r
7027 if ( info.sampleRates.size() == 0 ) {
\r
7028 snd_pcm_close( phandle );
\r
7029 errorStream_ << "RtApiAlsa::getDeviceInfo: no supported sample rates found for device (" << name << ").";
\r
7030 errorText_ = errorStream_.str();
\r
7031 error( RtAudioError::WARNING );
\r
7035 // Probe the supported data formats ... we don't care about endian-ness just yet
\r
7036 snd_pcm_format_t format;
\r
7037 info.nativeFormats = 0;
\r
7038 format = SND_PCM_FORMAT_S8;
\r
7039 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
\r
7040 info.nativeFormats |= RTAUDIO_SINT8;
\r
7041 format = SND_PCM_FORMAT_S16;
\r
7042 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
\r
7043 info.nativeFormats |= RTAUDIO_SINT16;
\r
7044 format = SND_PCM_FORMAT_S24;
\r
7045 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
\r
7046 info.nativeFormats |= RTAUDIO_SINT24;
\r
7047 format = SND_PCM_FORMAT_S32;
\r
7048 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
\r
7049 info.nativeFormats |= RTAUDIO_SINT32;
\r
7050 format = SND_PCM_FORMAT_FLOAT;
\r
7051 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
\r
7052 info.nativeFormats |= RTAUDIO_FLOAT32;
\r
7053 format = SND_PCM_FORMAT_FLOAT64;
\r
7054 if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
\r
7055 info.nativeFormats |= RTAUDIO_FLOAT64;
\r
7057 // Check that we have at least one supported format
\r
7058 if ( info.nativeFormats == 0 ) {
\r
7059 snd_pcm_close( phandle );
\r
7060 errorStream_ << "RtApiAlsa::getDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio.";
\r
7061 errorText_ = errorStream_.str();
\r
7062 error( RtAudioError::WARNING );
\r
7066 // Get the device name
\r
7068 result = snd_card_get_name( card, &cardname );
\r
7069 if ( result >= 0 ) {
\r
7070 sprintf( name, "hw:%s,%d", cardname, subdevice );
\r
7075 // That's all ... close the device and return
\r
7076 snd_pcm_close( phandle );
\r
7077 info.probed = true;
\r
7081 void RtApiAlsa :: saveDeviceInfo( void )
\r
7085 unsigned int nDevices = getDeviceCount();
\r
7086 devices_.resize( nDevices );
\r
7087 for ( unsigned int i=0; i<nDevices; i++ )
\r
7088 devices_[i] = getDeviceInfo( i );
\r
7091 bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
\r
7092 unsigned int firstChannel, unsigned int sampleRate,
\r
7093 RtAudioFormat format, unsigned int *bufferSize,
\r
7094 RtAudio::StreamOptions *options )
\r
7097 #if defined(__RTAUDIO_DEBUG__)
\r
7098 snd_output_t *out;
\r
7099 snd_output_stdio_attach(&out, stderr, 0);
\r
7102 // I'm not using the "plug" interface ... too much inconsistent behavior.
\r
7104 unsigned nDevices = 0;
\r
7105 int result, subdevice, card;
\r
7107 snd_ctl_t *chandle;
\r
7109 if ( options && options->flags & RTAUDIO_ALSA_USE_DEFAULT )
\r
7110 snprintf(name, sizeof(name), "%s", "default");
\r
7112 // Count cards and devices
\r
7114 snd_card_next( &card );
\r
7115 while ( card >= 0 ) {
\r
7116 sprintf( name, "hw:%d", card );
\r
7117 result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
\r
7118 if ( result < 0 ) {
\r
7119 errorStream_ << "RtApiAlsa::probeDeviceOpen: control open, card = " << card << ", " << snd_strerror( result ) << ".";
\r
7120 errorText_ = errorStream_.str();
\r
7125 result = snd_ctl_pcm_next_device( chandle, &subdevice );
\r
7126 if ( result < 0 ) break;
\r
7127 if ( subdevice < 0 ) break;
\r
7128 if ( nDevices == device ) {
\r
7129 sprintf( name, "hw:%d,%d", card, subdevice );
\r
7130 snd_ctl_close( chandle );
\r
7135 snd_ctl_close( chandle );
\r
7136 snd_card_next( &card );
\r
7139 result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
\r
7140 if ( result == 0 ) {
\r
7141 if ( nDevices == device ) {
\r
7142 strcpy( name, "default" );
\r
7148 if ( nDevices == 0 ) {
\r
7149 // This should not happen because a check is made before this function is called.
\r
7150 errorText_ = "RtApiAlsa::probeDeviceOpen: no devices found!";
\r
7154 if ( device >= nDevices ) {
\r
7155 // This should not happen because a check is made before this function is called.
\r
7156 errorText_ = "RtApiAlsa::probeDeviceOpen: device ID is invalid!";
\r
7163 // The getDeviceInfo() function will not work for a device that is
\r
7164 // already open. Thus, we'll probe the system before opening a
\r
7165 // stream and save the results for use by getDeviceInfo().
\r
7166 if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) // only do once
\r
7167 this->saveDeviceInfo();
\r
7169 snd_pcm_stream_t stream;
\r
7170 if ( mode == OUTPUT )
\r
7171 stream = SND_PCM_STREAM_PLAYBACK;
\r
7173 stream = SND_PCM_STREAM_CAPTURE;
\r
7175 snd_pcm_t *phandle;
\r
7176 int openMode = SND_PCM_ASYNC;
\r
7177 result = snd_pcm_open( &phandle, name, stream, openMode );
\r
7178 if ( result < 0 ) {
\r
7179 if ( mode == OUTPUT )
\r
7180 errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for output.";
\r
7182 errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for input.";
\r
7183 errorText_ = errorStream_.str();
\r
7187 // Fill the parameter structure.
\r
7188 snd_pcm_hw_params_t *hw_params;
\r
7189 snd_pcm_hw_params_alloca( &hw_params );
\r
7190 result = snd_pcm_hw_params_any( phandle, hw_params );
\r
7191 if ( result < 0 ) {
\r
7192 snd_pcm_close( phandle );
\r
7193 errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") parameters, " << snd_strerror( result ) << ".";
\r
7194 errorText_ = errorStream_.str();
\r
7198 #if defined(__RTAUDIO_DEBUG__)
\r
7199 fprintf( stderr, "\nRtApiAlsa: dump hardware params just after device open:\n\n" );
\r
7200 snd_pcm_hw_params_dump( hw_params, out );
\r
7203 // Set access ... check user preference.
\r
7204 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) {
\r
7205 stream_.userInterleaved = false;
\r
7206 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
\r
7207 if ( result < 0 ) {
\r
7208 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
\r
7209 stream_.deviceInterleaved[mode] = true;
\r
7212 stream_.deviceInterleaved[mode] = false;
\r
7215 stream_.userInterleaved = true;
\r
7216 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
\r
7217 if ( result < 0 ) {
\r
7218 result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
\r
7219 stream_.deviceInterleaved[mode] = false;
\r
7222 stream_.deviceInterleaved[mode] = true;
\r
7225 if ( result < 0 ) {
\r
7226 snd_pcm_close( phandle );
\r
7227 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") access, " << snd_strerror( result ) << ".";
\r
7228 errorText_ = errorStream_.str();
\r
7232 // Determine how to set the device format.
\r
7233 stream_.userFormat = format;
\r
7234 snd_pcm_format_t deviceFormat = SND_PCM_FORMAT_UNKNOWN;
\r
7236 if ( format == RTAUDIO_SINT8 )
\r
7237 deviceFormat = SND_PCM_FORMAT_S8;
\r
7238 else if ( format == RTAUDIO_SINT16 )
\r
7239 deviceFormat = SND_PCM_FORMAT_S16;
\r
7240 else if ( format == RTAUDIO_SINT24 )
\r
7241 deviceFormat = SND_PCM_FORMAT_S24;
\r
7242 else if ( format == RTAUDIO_SINT32 )
\r
7243 deviceFormat = SND_PCM_FORMAT_S32;
\r
7244 else if ( format == RTAUDIO_FLOAT32 )
\r
7245 deviceFormat = SND_PCM_FORMAT_FLOAT;
\r
7246 else if ( format == RTAUDIO_FLOAT64 )
\r
7247 deviceFormat = SND_PCM_FORMAT_FLOAT64;
\r
7249 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat) == 0) {
\r
7250 stream_.deviceFormat[mode] = format;
\r
7254 // The user requested format is not natively supported by the device.
\r
7255 deviceFormat = SND_PCM_FORMAT_FLOAT64;
\r
7256 if ( snd_pcm_hw_params_test_format( phandle, hw_params, deviceFormat ) == 0 ) {
\r
7257 stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
\r
7261 deviceFormat = SND_PCM_FORMAT_FLOAT;
\r
7262 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
\r
7263 stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
\r
7267 deviceFormat = SND_PCM_FORMAT_S32;
\r
7268 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
\r
7269 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
\r
7273 deviceFormat = SND_PCM_FORMAT_S24;
\r
7274 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
\r
7275 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
\r
7279 deviceFormat = SND_PCM_FORMAT_S16;
\r
7280 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
\r
7281 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
\r
7285 deviceFormat = SND_PCM_FORMAT_S8;
\r
7286 if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
\r
7287 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
\r
7291 // If we get here, no supported format was found.
\r
7292 snd_pcm_close( phandle );
\r
7293 errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device " << device << " data format not supported by RtAudio.";
\r
7294 errorText_ = errorStream_.str();
\r
7298 result = snd_pcm_hw_params_set_format( phandle, hw_params, deviceFormat );
\r
7299 if ( result < 0 ) {
\r
7300 snd_pcm_close( phandle );
\r
7301 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") data format, " << snd_strerror( result ) << ".";
\r
7302 errorText_ = errorStream_.str();
\r
7306 // Determine whether byte-swaping is necessary.
\r
7307 stream_.doByteSwap[mode] = false;
\r
7308 if ( deviceFormat != SND_PCM_FORMAT_S8 ) {
\r
7309 result = snd_pcm_format_cpu_endian( deviceFormat );
\r
7310 if ( result == 0 )
\r
7311 stream_.doByteSwap[mode] = true;
\r
7312 else if (result < 0) {
\r
7313 snd_pcm_close( phandle );
\r
7314 errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") endian-ness, " << snd_strerror( result ) << ".";
\r
7315 errorText_ = errorStream_.str();
\r
7320 // Set the sample rate.
\r
7321 result = snd_pcm_hw_params_set_rate_near( phandle, hw_params, (unsigned int*) &sampleRate, 0 );
\r
7322 if ( result < 0 ) {
\r
7323 snd_pcm_close( phandle );
\r
7324 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting sample rate on device (" << name << "), " << snd_strerror( result ) << ".";
\r
7325 errorText_ = errorStream_.str();
\r
7329 // Determine the number of channels for this device. We support a possible
\r
7330 // minimum device channel number > than the value requested by the user.
\r
7331 stream_.nUserChannels[mode] = channels;
\r
7332 unsigned int value;
\r
7333 result = snd_pcm_hw_params_get_channels_max( hw_params, &value );
\r
7334 unsigned int deviceChannels = value;
\r
7335 if ( result < 0 || deviceChannels < channels + firstChannel ) {
\r
7336 snd_pcm_close( phandle );
\r
7337 errorStream_ << "RtApiAlsa::probeDeviceOpen: requested channel parameters not supported by device (" << name << "), " << snd_strerror( result ) << ".";
\r
7338 errorText_ = errorStream_.str();
\r
7342 result = snd_pcm_hw_params_get_channels_min( hw_params, &value );
\r
7343 if ( result < 0 ) {
\r
7344 snd_pcm_close( phandle );
\r
7345 errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting minimum channels for device (" << name << "), " << snd_strerror( result ) << ".";
\r
7346 errorText_ = errorStream_.str();
\r
7349 deviceChannels = value;
\r
7350 if ( deviceChannels < channels + firstChannel ) deviceChannels = channels + firstChannel;
\r
7351 stream_.nDeviceChannels[mode] = deviceChannels;
\r
7353 // Set the device channels.
\r
7354 result = snd_pcm_hw_params_set_channels( phandle, hw_params, deviceChannels );
\r
7355 if ( result < 0 ) {
\r
7356 snd_pcm_close( phandle );
\r
7357 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting channels for device (" << name << "), " << snd_strerror( result ) << ".";
\r
7358 errorText_ = errorStream_.str();
\r
7362 // Set the buffer (or period) size.
\r
7364 snd_pcm_uframes_t periodSize = *bufferSize;
\r
7365 result = snd_pcm_hw_params_set_period_size_near( phandle, hw_params, &periodSize, &dir );
\r
7366 if ( result < 0 ) {
\r
7367 snd_pcm_close( phandle );
\r
7368 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting period size for device (" << name << "), " << snd_strerror( result ) << ".";
\r
7369 errorText_ = errorStream_.str();
\r
7372 *bufferSize = periodSize;
\r
7374 // Set the buffer number, which in ALSA is referred to as the "period".
\r
7375 unsigned int periods = 0;
\r
7376 if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2;
\r
7377 if ( options && options->numberOfBuffers > 0 ) periods = options->numberOfBuffers;
\r
7378 if ( periods < 2 ) periods = 4; // a fairly safe default value
\r
7379 result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir );
\r
7380 if ( result < 0 ) {
\r
7381 snd_pcm_close( phandle );
\r
7382 errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << ".";
\r
7383 errorText_ = errorStream_.str();
\r
7387 // If attempting to setup a duplex stream, the bufferSize parameter
\r
7388 // MUST be the same in both directions!
\r
7389 if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {
\r
7390 snd_pcm_close( phandle );
\r
7391 errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ").";
\r
7392 errorText_ = errorStream_.str();
\r
7396 stream_.bufferSize = *bufferSize;
\r
7398 // Install the hardware configuration
\r
7399 result = snd_pcm_hw_params( phandle, hw_params );
\r
7400 if ( result < 0 ) {
\r
7401 snd_pcm_close( phandle );
\r
7402 errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing hardware configuration on device (" << name << "), " << snd_strerror( result ) << ".";
\r
7403 errorText_ = errorStream_.str();
\r
7407 #if defined(__RTAUDIO_DEBUG__)
\r
7408 fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n");
\r
7409 snd_pcm_hw_params_dump( hw_params, out );
\r
7412 // Set the software configuration to fill buffers with zeros and prevent device stopping on xruns.
\r
7413 snd_pcm_sw_params_t *sw_params = NULL;
\r
7414 snd_pcm_sw_params_alloca( &sw_params );
\r
7415 snd_pcm_sw_params_current( phandle, sw_params );
\r
7416 snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize );
\r
7417 snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, ULONG_MAX );
\r
7418 snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 );
\r
7420 // The following two settings were suggested by Theo Veenker
\r
7421 //snd_pcm_sw_params_set_avail_min( phandle, sw_params, *bufferSize );
\r
7422 //snd_pcm_sw_params_set_xfer_align( phandle, sw_params, 1 );
\r
7424 // here are two options for a fix
\r
7425 //snd_pcm_sw_params_set_silence_size( phandle, sw_params, ULONG_MAX );
\r
7426 snd_pcm_uframes_t val;
\r
7427 snd_pcm_sw_params_get_boundary( sw_params, &val );
\r
7428 snd_pcm_sw_params_set_silence_size( phandle, sw_params, val );
\r
7430 result = snd_pcm_sw_params( phandle, sw_params );
\r
7431 if ( result < 0 ) {
\r
7432 snd_pcm_close( phandle );
\r
7433 errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing software configuration on device (" << name << "), " << snd_strerror( result ) << ".";
\r
7434 errorText_ = errorStream_.str();
\r
7438 #if defined(__RTAUDIO_DEBUG__)
\r
7439 fprintf(stderr, "\nRtApiAlsa: dump software params after installation:\n\n");
\r
7440 snd_pcm_sw_params_dump( sw_params, out );
\r
7443 // Set flags for buffer conversion
\r
7444 stream_.doConvertBuffer[mode] = false;
\r
7445 if ( stream_.userFormat != stream_.deviceFormat[mode] )
\r
7446 stream_.doConvertBuffer[mode] = true;
\r
7447 if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
\r
7448 stream_.doConvertBuffer[mode] = true;
\r
7449 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
\r
7450 stream_.nUserChannels[mode] > 1 )
\r
7451 stream_.doConvertBuffer[mode] = true;
\r
7453 // Allocate the ApiHandle if necessary and then save.
\r
7454 AlsaHandle *apiInfo = 0;
\r
7455 if ( stream_.apiHandle == 0 ) {
\r
7457 apiInfo = (AlsaHandle *) new AlsaHandle;
\r
7459 catch ( std::bad_alloc& ) {
\r
7460 errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating AlsaHandle memory.";
\r
7464 if ( pthread_cond_init( &apiInfo->runnable_cv, NULL ) ) {
\r
7465 errorText_ = "RtApiAlsa::probeDeviceOpen: error initializing pthread condition variable.";
\r
7469 stream_.apiHandle = (void *) apiInfo;
\r
7470 apiInfo->handles[0] = 0;
\r
7471 apiInfo->handles[1] = 0;
\r
7474 apiInfo = (AlsaHandle *) stream_.apiHandle;
\r
7476 apiInfo->handles[mode] = phandle;
\r
7479 // Allocate necessary internal buffers.
\r
7480 unsigned long bufferBytes;
\r
7481 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
\r
7482 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
\r
7483 if ( stream_.userBuffer[mode] == NULL ) {
\r
7484 errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating user buffer memory.";
\r
7488 if ( stream_.doConvertBuffer[mode] ) {
\r
7490 bool makeBuffer = true;
\r
7491 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
\r
7492 if ( mode == INPUT ) {
\r
7493 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
\r
7494 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
\r
7495 if ( bufferBytes <= bytesOut ) makeBuffer = false;
\r
7499 if ( makeBuffer ) {
\r
7500 bufferBytes *= *bufferSize;
\r
7501 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
\r
7502 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
\r
7503 if ( stream_.deviceBuffer == NULL ) {
\r
7504 errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating device buffer memory.";
\r
7510 stream_.sampleRate = sampleRate;
\r
7511 stream_.nBuffers = periods;
\r
7512 stream_.device[mode] = device;
\r
7513 stream_.state = STREAM_STOPPED;
\r
7515 // Setup the buffer conversion information structure.
\r
7516 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
\r
7518 // Setup thread if necessary.
\r
7519 if ( stream_.mode == OUTPUT && mode == INPUT ) {
\r
7520 // We had already set up an output stream.
\r
7521 stream_.mode = DUPLEX;
\r
7522 // Link the streams if possible.
\r
7523 apiInfo->synchronized = false;
\r
7524 if ( snd_pcm_link( apiInfo->handles[0], apiInfo->handles[1] ) == 0 )
\r
7525 apiInfo->synchronized = true;
\r
7527 errorText_ = "RtApiAlsa::probeDeviceOpen: unable to synchronize input and output devices.";
\r
7528 error( RtAudioError::WARNING );
\r
7532 stream_.mode = mode;
\r
7534 // Setup callback thread.
\r
7535 stream_.callbackInfo.object = (void *) this;
\r
7537 // Set the thread attributes for joinable and realtime scheduling
\r
7538 // priority (optional). The higher priority will only take affect
\r
7539 // if the program is run as root or suid. Note, under Linux
\r
7540 // processes with CAP_SYS_NICE privilege, a user can change
\r
7541 // scheduling policy and priority (thus need not be root). See
\r
7542 // POSIX "capabilities".
\r
7543 pthread_attr_t attr;
\r
7544 pthread_attr_init( &attr );
\r
7545 pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
\r
7547 #ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
\r
7548 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
\r
7549 // We previously attempted to increase the audio callback priority
\r
7550 // to SCHED_RR here via the attributes. However, while no errors
\r
7551 // were reported in doing so, it did not work. So, now this is
\r
7552 // done in the alsaCallbackHandler function.
\r
7553 stream_.callbackInfo.doRealtime = true;
\r
7554 int priority = options->priority;
\r
7555 int min = sched_get_priority_min( SCHED_RR );
\r
7556 int max = sched_get_priority_max( SCHED_RR );
\r
7557 if ( priority < min ) priority = min;
\r
7558 else if ( priority > max ) priority = max;
\r
7559 stream_.callbackInfo.priority = priority;
\r
7563 stream_.callbackInfo.isRunning = true;
\r
7564 result = pthread_create( &stream_.callbackInfo.thread, &attr, alsaCallbackHandler, &stream_.callbackInfo );
\r
7565 pthread_attr_destroy( &attr );
\r
7567 stream_.callbackInfo.isRunning = false;
\r
7568 errorText_ = "RtApiAlsa::error creating callback thread!";
\r
7577 pthread_cond_destroy( &apiInfo->runnable_cv );
\r
7578 if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
\r
7579 if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
\r
7581 stream_.apiHandle = 0;
\r
7584 if ( phandle) snd_pcm_close( phandle );
\r
7586 for ( int i=0; i<2; i++ ) {
\r
7587 if ( stream_.userBuffer[i] ) {
\r
7588 free( stream_.userBuffer[i] );
\r
7589 stream_.userBuffer[i] = 0;
\r
7593 if ( stream_.deviceBuffer ) {
\r
7594 free( stream_.deviceBuffer );
\r
7595 stream_.deviceBuffer = 0;
\r
7598 stream_.state = STREAM_CLOSED;
\r
7602 void RtApiAlsa :: closeStream()
\r
7604 if ( stream_.state == STREAM_CLOSED ) {
\r
7605 errorText_ = "RtApiAlsa::closeStream(): no open stream to close!";
\r
7606 error( RtAudioError::WARNING );
\r
7610 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
\r
7611 stream_.callbackInfo.isRunning = false;
\r
7612 MUTEX_LOCK( &stream_.mutex );
\r
7613 if ( stream_.state == STREAM_STOPPED ) {
\r
7614 apiInfo->runnable = true;
\r
7615 pthread_cond_signal( &apiInfo->runnable_cv );
\r
7617 MUTEX_UNLOCK( &stream_.mutex );
\r
7618 pthread_join( stream_.callbackInfo.thread, NULL );
\r
7620 if ( stream_.state == STREAM_RUNNING ) {
\r
7621 stream_.state = STREAM_STOPPED;
\r
7622 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
\r
7623 snd_pcm_drop( apiInfo->handles[0] );
\r
7624 if ( stream_.mode == INPUT || stream_.mode == DUPLEX )
\r
7625 snd_pcm_drop( apiInfo->handles[1] );
\r
7629 pthread_cond_destroy( &apiInfo->runnable_cv );
\r
7630 if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
\r
7631 if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
\r
7633 stream_.apiHandle = 0;
\r
7636 for ( int i=0; i<2; i++ ) {
\r
7637 if ( stream_.userBuffer[i] ) {
\r
7638 free( stream_.userBuffer[i] );
\r
7639 stream_.userBuffer[i] = 0;
\r
7643 if ( stream_.deviceBuffer ) {
\r
7644 free( stream_.deviceBuffer );
\r
7645 stream_.deviceBuffer = 0;
\r
7648 stream_.mode = UNINITIALIZED;
\r
7649 stream_.state = STREAM_CLOSED;
\r
7652 void RtApiAlsa :: startStream()
\r
7654 // This method calls snd_pcm_prepare if the device isn't already in that state.
\r
7657 if ( stream_.state == STREAM_RUNNING ) {
\r
7658 errorText_ = "RtApiAlsa::startStream(): the stream is already running!";
\r
7659 error( RtAudioError::WARNING );
\r
7663 MUTEX_LOCK( &stream_.mutex );
\r
7666 snd_pcm_state_t state;
\r
7667 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
\r
7668 snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
\r
7669 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
\r
7670 state = snd_pcm_state( handle[0] );
\r
7671 if ( state != SND_PCM_STATE_PREPARED ) {
\r
7672 result = snd_pcm_prepare( handle[0] );
\r
7673 if ( result < 0 ) {
\r
7674 errorStream_ << "RtApiAlsa::startStream: error preparing output pcm device, " << snd_strerror( result ) << ".";
\r
7675 errorText_ = errorStream_.str();
\r
7681 if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
\r
7682 result = snd_pcm_drop(handle[1]); // fix to remove stale data received since device has been open
\r
7683 state = snd_pcm_state( handle[1] );
\r
7684 if ( state != SND_PCM_STATE_PREPARED ) {
\r
7685 result = snd_pcm_prepare( handle[1] );
\r
7686 if ( result < 0 ) {
\r
7687 errorStream_ << "RtApiAlsa::startStream: error preparing input pcm device, " << snd_strerror( result ) << ".";
\r
7688 errorText_ = errorStream_.str();
\r
7694 stream_.state = STREAM_RUNNING;
\r
7697 apiInfo->runnable = true;
\r
7698 pthread_cond_signal( &apiInfo->runnable_cv );
\r
7699 MUTEX_UNLOCK( &stream_.mutex );
\r
7701 if ( result >= 0 ) return;
\r
7702 error( RtAudioError::SYSTEM_ERROR );
\r
7705 void RtApiAlsa :: stopStream()
\r
7708 if ( stream_.state == STREAM_STOPPED ) {
\r
7709 errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!";
\r
7710 error( RtAudioError::WARNING );
\r
7714 stream_.state = STREAM_STOPPED;
\r
7715 MUTEX_LOCK( &stream_.mutex );
\r
7718 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
\r
7719 snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
\r
7720 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
\r
7721 if ( apiInfo->synchronized )
\r
7722 result = snd_pcm_drop( handle[0] );
\r
7724 result = snd_pcm_drain( handle[0] );
\r
7725 if ( result < 0 ) {
\r
7726 errorStream_ << "RtApiAlsa::stopStream: error draining output pcm device, " << snd_strerror( result ) << ".";
\r
7727 errorText_ = errorStream_.str();
\r
7732 if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
\r
7733 result = snd_pcm_drop( handle[1] );
\r
7734 if ( result < 0 ) {
\r
7735 errorStream_ << "RtApiAlsa::stopStream: error stopping input pcm device, " << snd_strerror( result ) << ".";
\r
7736 errorText_ = errorStream_.str();
\r
7742 apiInfo->runnable = false; // fixes high CPU usage when stopped
\r
7743 MUTEX_UNLOCK( &stream_.mutex );
\r
7745 if ( result >= 0 ) return;
\r
7746 error( RtAudioError::SYSTEM_ERROR );
\r
7749 void RtApiAlsa :: abortStream()
\r
7752 if ( stream_.state == STREAM_STOPPED ) {
\r
7753 errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!";
\r
7754 error( RtAudioError::WARNING );
\r
7758 stream_.state = STREAM_STOPPED;
\r
7759 MUTEX_LOCK( &stream_.mutex );
\r
7762 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
\r
7763 snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
\r
7764 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
\r
7765 result = snd_pcm_drop( handle[0] );
\r
7766 if ( result < 0 ) {
\r
7767 errorStream_ << "RtApiAlsa::abortStream: error aborting output pcm device, " << snd_strerror( result ) << ".";
\r
7768 errorText_ = errorStream_.str();
\r
7773 if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
\r
7774 result = snd_pcm_drop( handle[1] );
\r
7775 if ( result < 0 ) {
\r
7776 errorStream_ << "RtApiAlsa::abortStream: error aborting input pcm device, " << snd_strerror( result ) << ".";
\r
7777 errorText_ = errorStream_.str();
\r
7783 apiInfo->runnable = false; // fixes high CPU usage when stopped
\r
7784 MUTEX_UNLOCK( &stream_.mutex );
\r
7786 if ( result >= 0 ) return;
\r
7787 error( RtAudioError::SYSTEM_ERROR );
\r
7790 void RtApiAlsa :: callbackEvent()
\r
7792 AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
\r
7793 if ( stream_.state == STREAM_STOPPED ) {
\r
7794 MUTEX_LOCK( &stream_.mutex );
\r
7795 while ( !apiInfo->runnable )
\r
7796 pthread_cond_wait( &apiInfo->runnable_cv, &stream_.mutex );
\r
7798 if ( stream_.state != STREAM_RUNNING ) {
\r
7799 MUTEX_UNLOCK( &stream_.mutex );
\r
7802 MUTEX_UNLOCK( &stream_.mutex );
\r
7805 if ( stream_.state == STREAM_CLOSED ) {
\r
7806 errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!";
\r
7807 error( RtAudioError::WARNING );
\r
7811 int doStopStream = 0;
\r
7812 RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
\r
7813 double streamTime = getStreamTime();
\r
7814 RtAudioStreamStatus status = 0;
\r
7815 if ( stream_.mode != INPUT && apiInfo->xrun[0] == true ) {
\r
7816 status |= RTAUDIO_OUTPUT_UNDERFLOW;
\r
7817 apiInfo->xrun[0] = false;
\r
7819 if ( stream_.mode != OUTPUT && apiInfo->xrun[1] == true ) {
\r
7820 status |= RTAUDIO_INPUT_OVERFLOW;
\r
7821 apiInfo->xrun[1] = false;
\r
7823 doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
\r
7824 stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
\r
7826 if ( doStopStream == 2 ) {
\r
7831 MUTEX_LOCK( &stream_.mutex );
\r
7833 // The state might change while waiting on a mutex.
\r
7834 if ( stream_.state == STREAM_STOPPED ) goto unlock;
\r
7839 snd_pcm_t **handle;
\r
7840 snd_pcm_sframes_t frames;
\r
7841 RtAudioFormat format;
\r
7842 handle = (snd_pcm_t **) apiInfo->handles;
\r
7844 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
\r
7846 // Setup parameters.
\r
7847 if ( stream_.doConvertBuffer[1] ) {
\r
7848 buffer = stream_.deviceBuffer;
\r
7849 channels = stream_.nDeviceChannels[1];
\r
7850 format = stream_.deviceFormat[1];
\r
7853 buffer = stream_.userBuffer[1];
\r
7854 channels = stream_.nUserChannels[1];
\r
7855 format = stream_.userFormat;
\r
7858 // Read samples from device in interleaved/non-interleaved format.
\r
7859 if ( stream_.deviceInterleaved[1] )
\r
7860 result = snd_pcm_readi( handle[1], buffer, stream_.bufferSize );
\r
7862 void *bufs[channels];
\r
7863 size_t offset = stream_.bufferSize * formatBytes( format );
\r
7864 for ( int i=0; i<channels; i++ )
\r
7865 bufs[i] = (void *) (buffer + (i * offset));
\r
7866 result = snd_pcm_readn( handle[1], bufs, stream_.bufferSize );
\r
7869 if ( result < (int) stream_.bufferSize ) {
\r
7870 // Either an error or overrun occured.
\r
7871 if ( result == -EPIPE ) {
\r
7872 snd_pcm_state_t state = snd_pcm_state( handle[1] );
\r
7873 if ( state == SND_PCM_STATE_XRUN ) {
\r
7874 apiInfo->xrun[1] = true;
\r
7875 result = snd_pcm_prepare( handle[1] );
\r
7876 if ( result < 0 ) {
\r
7877 errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after overrun, " << snd_strerror( result ) << ".";
\r
7878 errorText_ = errorStream_.str();
\r
7882 errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
\r
7883 errorText_ = errorStream_.str();
\r
7887 errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << ".";
\r
7888 errorText_ = errorStream_.str();
\r
7890 error( RtAudioError::WARNING );
\r
7894 // Do byte swapping if necessary.
\r
7895 if ( stream_.doByteSwap[1] )
\r
7896 byteSwapBuffer( buffer, stream_.bufferSize * channels, format );
\r
7898 // Do buffer conversion if necessary.
\r
7899 if ( stream_.doConvertBuffer[1] )
\r
7900 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
\r
7902 // Check stream latency
\r
7903 result = snd_pcm_delay( handle[1], &frames );
\r
7904 if ( result == 0 && frames > 0 ) stream_.latency[1] = frames;
\r
7909 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
\r
7911 // Setup parameters and do buffer conversion if necessary.
\r
7912 if ( stream_.doConvertBuffer[0] ) {
\r
7913 buffer = stream_.deviceBuffer;
\r
7914 convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
\r
7915 channels = stream_.nDeviceChannels[0];
\r
7916 format = stream_.deviceFormat[0];
\r
7919 buffer = stream_.userBuffer[0];
\r
7920 channels = stream_.nUserChannels[0];
\r
7921 format = stream_.userFormat;
\r
7924 // Do byte swapping if necessary.
\r
7925 if ( stream_.doByteSwap[0] )
\r
7926 byteSwapBuffer(buffer, stream_.bufferSize * channels, format);
\r
7928 // Write samples to device in interleaved/non-interleaved format.
\r
7929 if ( stream_.deviceInterleaved[0] )
\r
7930 result = snd_pcm_writei( handle[0], buffer, stream_.bufferSize );
\r
7932 void *bufs[channels];
\r
7933 size_t offset = stream_.bufferSize * formatBytes( format );
\r
7934 for ( int i=0; i<channels; i++ )
\r
7935 bufs[i] = (void *) (buffer + (i * offset));
\r
7936 result = snd_pcm_writen( handle[0], bufs, stream_.bufferSize );
\r
7939 if ( result < (int) stream_.bufferSize ) {
\r
7940 // Either an error or underrun occured.
\r
7941 if ( result == -EPIPE ) {
\r
7942 snd_pcm_state_t state = snd_pcm_state( handle[0] );
\r
7943 if ( state == SND_PCM_STATE_XRUN ) {
\r
7944 apiInfo->xrun[0] = true;
\r
7945 result = snd_pcm_prepare( handle[0] );
\r
7946 if ( result < 0 ) {
\r
7947 errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after underrun, " << snd_strerror( result ) << ".";
\r
7948 errorText_ = errorStream_.str();
\r
7952 errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
\r
7953 errorText_ = errorStream_.str();
\r
7957 errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << ".";
\r
7958 errorText_ = errorStream_.str();
\r
7960 error( RtAudioError::WARNING );
\r
7964 // Check stream latency
\r
7965 result = snd_pcm_delay( handle[0], &frames );
\r
7966 if ( result == 0 && frames > 0 ) stream_.latency[0] = frames;
\r
7970 MUTEX_UNLOCK( &stream_.mutex );
\r
7972 RtApi::tickStreamTime();
\r
7973 if ( doStopStream == 1 ) this->stopStream();
\r
7976 static void *alsaCallbackHandler( void *ptr )
\r
7978 CallbackInfo *info = (CallbackInfo *) ptr;
\r
7979 RtApiAlsa *object = (RtApiAlsa *) info->object;
\r
7980 bool *isRunning = &info->isRunning;
\r
7982 #ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
\r
7983 if ( &info->doRealtime ) {
\r
7984 pthread_t tID = pthread_self(); // ID of this thread
\r
7985 sched_param prio = { info->priority }; // scheduling priority of thread
\r
7986 pthread_setschedparam( tID, SCHED_RR, &prio );
\r
7990 while ( *isRunning == true ) {
\r
7991 pthread_testcancel();
\r
7992 object->callbackEvent();
\r
7995 pthread_exit( NULL );
\r
7998 //******************** End of __LINUX_ALSA__ *********************//
\r
8001 #if defined(__LINUX_PULSE__)
\r
8003 // Code written by Peter Meerwald, pmeerw@pmeerw.net
\r
8004 // and Tristan Matthews.
\r
8006 #include <pulse/error.h>
\r
8007 #include <pulse/simple.h>
\r
8010 static const unsigned int SUPPORTED_SAMPLERATES[] = { 8000, 16000, 22050, 32000,
\r
8011 44100, 48000, 96000, 0};
\r
8013 struct rtaudio_pa_format_mapping_t {
\r
8014 RtAudioFormat rtaudio_format;
\r
8015 pa_sample_format_t pa_format;
\r
8018 static const rtaudio_pa_format_mapping_t supported_sampleformats[] = {
\r
8019 {RTAUDIO_SINT16, PA_SAMPLE_S16LE},
\r
8020 {RTAUDIO_SINT32, PA_SAMPLE_S32LE},
\r
8021 {RTAUDIO_FLOAT32, PA_SAMPLE_FLOAT32LE},
\r
8022 {0, PA_SAMPLE_INVALID}};
\r
8024 struct PulseAudioHandle {
\r
8025 pa_simple *s_play;
\r
8028 pthread_cond_t runnable_cv;
\r
8030 PulseAudioHandle() : s_play(0), s_rec(0), runnable(false) { }
\r
8033 RtApiPulse::~RtApiPulse()
\r
8035 if ( stream_.state != STREAM_CLOSED )
\r
8039 unsigned int RtApiPulse::getDeviceCount( void )
\r
8044 RtAudio::DeviceInfo RtApiPulse::getDeviceInfo( unsigned int /*device*/ )
\r
8046 RtAudio::DeviceInfo info;
\r
8047 info.probed = true;
\r
8048 info.name = "PulseAudio";
\r
8049 info.outputChannels = 2;
\r
8050 info.inputChannels = 2;
\r
8051 info.duplexChannels = 2;
\r
8052 info.isDefaultOutput = true;
\r
8053 info.isDefaultInput = true;
\r
8055 for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr )
\r
8056 info.sampleRates.push_back( *sr );
\r
8058 info.nativeFormats = RTAUDIO_SINT16 | RTAUDIO_SINT32 | RTAUDIO_FLOAT32;
\r
8063 static void *pulseaudio_callback( void * user )
\r
8065 CallbackInfo *cbi = static_cast<CallbackInfo *>( user );
\r
8066 RtApiPulse *context = static_cast<RtApiPulse *>( cbi->object );
\r
8067 volatile bool *isRunning = &cbi->isRunning;
\r
8069 while ( *isRunning ) {
\r
8070 pthread_testcancel();
\r
8071 context->callbackEvent();
\r
8074 pthread_exit( NULL );
\r
8077 void RtApiPulse::closeStream( void )
\r
8079 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
\r
8081 stream_.callbackInfo.isRunning = false;
\r
8083 MUTEX_LOCK( &stream_.mutex );
\r
8084 if ( stream_.state == STREAM_STOPPED ) {
\r
8085 pah->runnable = true;
\r
8086 pthread_cond_signal( &pah->runnable_cv );
\r
8088 MUTEX_UNLOCK( &stream_.mutex );
\r
8090 pthread_join( pah->thread, 0 );
\r
8091 if ( pah->s_play ) {
\r
8092 pa_simple_flush( pah->s_play, NULL );
\r
8093 pa_simple_free( pah->s_play );
\r
8096 pa_simple_free( pah->s_rec );
\r
8098 pthread_cond_destroy( &pah->runnable_cv );
\r
8100 stream_.apiHandle = 0;
\r
8103 if ( stream_.userBuffer[0] ) {
\r
8104 free( stream_.userBuffer[0] );
\r
8105 stream_.userBuffer[0] = 0;
\r
8107 if ( stream_.userBuffer[1] ) {
\r
8108 free( stream_.userBuffer[1] );
\r
8109 stream_.userBuffer[1] = 0;
\r
8112 stream_.state = STREAM_CLOSED;
\r
8113 stream_.mode = UNINITIALIZED;
\r
8116 void RtApiPulse::callbackEvent( void )
\r
8118 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
\r
8120 if ( stream_.state == STREAM_STOPPED ) {
\r
8121 MUTEX_LOCK( &stream_.mutex );
\r
8122 while ( !pah->runnable )
\r
8123 pthread_cond_wait( &pah->runnable_cv, &stream_.mutex );
\r
8125 if ( stream_.state != STREAM_RUNNING ) {
\r
8126 MUTEX_UNLOCK( &stream_.mutex );
\r
8129 MUTEX_UNLOCK( &stream_.mutex );
\r
8132 if ( stream_.state == STREAM_CLOSED ) {
\r
8133 errorText_ = "RtApiPulse::callbackEvent(): the stream is closed ... "
\r
8134 "this shouldn't happen!";
\r
8135 error( RtAudioError::WARNING );
\r
8139 RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
\r
8140 double streamTime = getStreamTime();
\r
8141 RtAudioStreamStatus status = 0;
\r
8142 int doStopStream = callback( stream_.userBuffer[OUTPUT], stream_.userBuffer[INPUT],
\r
8143 stream_.bufferSize, streamTime, status,
\r
8144 stream_.callbackInfo.userData );
\r
8146 if ( doStopStream == 2 ) {
\r
8151 MUTEX_LOCK( &stream_.mutex );
\r
8152 void *pulse_in = stream_.doConvertBuffer[INPUT] ? stream_.deviceBuffer : stream_.userBuffer[INPUT];
\r
8153 void *pulse_out = stream_.doConvertBuffer[OUTPUT] ? stream_.deviceBuffer : stream_.userBuffer[OUTPUT];
\r
8155 if ( stream_.state != STREAM_RUNNING )
\r
8160 if (stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
\r
8161 if ( stream_.doConvertBuffer[OUTPUT] ) {
\r
8162 convertBuffer( stream_.deviceBuffer,
\r
8163 stream_.userBuffer[OUTPUT],
\r
8164 stream_.convertInfo[OUTPUT] );
\r
8165 bytes = stream_.nDeviceChannels[OUTPUT] * stream_.bufferSize *
\r
8166 formatBytes( stream_.deviceFormat[OUTPUT] );
\r
8168 bytes = stream_.nUserChannels[OUTPUT] * stream_.bufferSize *
\r
8169 formatBytes( stream_.userFormat );
\r
8171 if ( pa_simple_write( pah->s_play, pulse_out, bytes, &pa_error ) < 0 ) {
\r
8172 errorStream_ << "RtApiPulse::callbackEvent: audio write error, " <<
\r
8173 pa_strerror( pa_error ) << ".";
\r
8174 errorText_ = errorStream_.str();
\r
8175 error( RtAudioError::WARNING );
\r
8179 if ( stream_.mode == INPUT || stream_.mode == DUPLEX) {
\r
8180 if ( stream_.doConvertBuffer[INPUT] )
\r
8181 bytes = stream_.nDeviceChannels[INPUT] * stream_.bufferSize *
\r
8182 formatBytes( stream_.deviceFormat[INPUT] );
\r
8184 bytes = stream_.nUserChannels[INPUT] * stream_.bufferSize *
\r
8185 formatBytes( stream_.userFormat );
\r
8187 if ( pa_simple_read( pah->s_rec, pulse_in, bytes, &pa_error ) < 0 ) {
\r
8188 errorStream_ << "RtApiPulse::callbackEvent: audio read error, " <<
\r
8189 pa_strerror( pa_error ) << ".";
\r
8190 errorText_ = errorStream_.str();
\r
8191 error( RtAudioError::WARNING );
\r
8193 if ( stream_.doConvertBuffer[INPUT] ) {
\r
8194 convertBuffer( stream_.userBuffer[INPUT],
\r
8195 stream_.deviceBuffer,
\r
8196 stream_.convertInfo[INPUT] );
\r
8201 MUTEX_UNLOCK( &stream_.mutex );
\r
8202 RtApi::tickStreamTime();
\r
8204 if ( doStopStream == 1 )
\r
8208 void RtApiPulse::startStream( void )
\r
8210 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
\r
8212 if ( stream_.state == STREAM_CLOSED ) {
\r
8213 errorText_ = "RtApiPulse::startStream(): the stream is not open!";
\r
8214 error( RtAudioError::INVALID_USE );
\r
8217 if ( stream_.state == STREAM_RUNNING ) {
\r
8218 errorText_ = "RtApiPulse::startStream(): the stream is already running!";
\r
8219 error( RtAudioError::WARNING );
\r
8223 MUTEX_LOCK( &stream_.mutex );
\r
8225 stream_.state = STREAM_RUNNING;
\r
8227 pah->runnable = true;
\r
8228 pthread_cond_signal( &pah->runnable_cv );
\r
8229 MUTEX_UNLOCK( &stream_.mutex );
\r
8232 void RtApiPulse::stopStream( void )
\r
8234 PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
\r
8236 if ( stream_.state == STREAM_CLOSED ) {
\r
8237 errorText_ = "RtApiPulse::stopStream(): the stream is not open!";
\r
8238 error( RtAudioError::INVALID_USE );
\r
8241 if ( stream_.state == STREAM_STOPPED ) {
\r
8242 errorText_ = "RtApiPulse::stopStream(): the stream is already stopped!";
\r
8243 error( RtAudioError::WARNING );
\r
8247 stream_.state = STREAM_STOPPED;
\r
8248 MUTEX_LOCK( &stream_.mutex );
\r
8250 if ( pah && pah->s_play ) {
\r
8252 if ( pa_simple_drain( pah->s_play, &pa_error ) < 0 ) {
\r
8253 errorStream_ << "RtApiPulse::stopStream: error draining output device, " <<
\r
8254 pa_strerror( pa_error ) << ".";
\r
8255 errorText_ = errorStream_.str();
\r
8256 MUTEX_UNLOCK( &stream_.mutex );
\r
8257 error( RtAudioError::SYSTEM_ERROR );
\r
8262 stream_.state = STREAM_STOPPED;
\r
8263 MUTEX_UNLOCK( &stream_.mutex );
\r
8266 void RtApiPulse::abortStream( void )
\r
8268 PulseAudioHandle *pah = static_cast<PulseAudioHandle*>( stream_.apiHandle );
\r
8270 if ( stream_.state == STREAM_CLOSED ) {
\r
8271 errorText_ = "RtApiPulse::abortStream(): the stream is not open!";
\r
8272 error( RtAudioError::INVALID_USE );
\r
8275 if ( stream_.state == STREAM_STOPPED ) {
\r
8276 errorText_ = "RtApiPulse::abortStream(): the stream is already stopped!";
\r
8277 error( RtAudioError::WARNING );
\r
8281 stream_.state = STREAM_STOPPED;
\r
8282 MUTEX_LOCK( &stream_.mutex );
\r
8284 if ( pah && pah->s_play ) {
\r
8286 if ( pa_simple_flush( pah->s_play, &pa_error ) < 0 ) {
\r
8287 errorStream_ << "RtApiPulse::abortStream: error flushing output device, " <<
\r
8288 pa_strerror( pa_error ) << ".";
\r
8289 errorText_ = errorStream_.str();
\r
8290 MUTEX_UNLOCK( &stream_.mutex );
\r
8291 error( RtAudioError::SYSTEM_ERROR );
\r
8296 stream_.state = STREAM_STOPPED;
\r
8297 MUTEX_UNLOCK( &stream_.mutex );
\r
8300 bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode,
\r
8301 unsigned int channels, unsigned int firstChannel,
\r
8302 unsigned int sampleRate, RtAudioFormat format,
\r
8303 unsigned int *bufferSize, RtAudio::StreamOptions *options )
\r
8305 PulseAudioHandle *pah = 0;
\r
8306 unsigned long bufferBytes = 0;
\r
8307 pa_sample_spec ss;
\r
8309 if ( device != 0 ) return false;
\r
8310 if ( mode != INPUT && mode != OUTPUT ) return false;
\r
8311 if ( channels != 1 && channels != 2 ) {
\r
8312 errorText_ = "RtApiPulse::probeDeviceOpen: unsupported number of channels.";
\r
8315 ss.channels = channels;
\r
8317 if ( firstChannel != 0 ) return false;
\r
8319 bool sr_found = false;
\r
8320 for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr ) {
\r
8321 if ( sampleRate == *sr ) {
\r
8323 stream_.sampleRate = sampleRate;
\r
8324 ss.rate = sampleRate;
\r
8328 if ( !sr_found ) {
\r
8329 errorText_ = "RtApiPulse::probeDeviceOpen: unsupported sample rate.";
\r
8333 bool sf_found = 0;
\r
8334 for ( const rtaudio_pa_format_mapping_t *sf = supported_sampleformats;
\r
8335 sf->rtaudio_format && sf->pa_format != PA_SAMPLE_INVALID; ++sf ) {
\r
8336 if ( format == sf->rtaudio_format ) {
\r
8338 stream_.userFormat = sf->rtaudio_format;
\r
8339 stream_.deviceFormat[mode] = stream_.userFormat;
\r
8340 ss.format = sf->pa_format;
\r
8344 if ( !sf_found ) { // Use internal data format conversion.
\r
8345 stream_.userFormat = format;
\r
8346 stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
\r
8347 ss.format = PA_SAMPLE_FLOAT32LE;
\r
8350 // Set other stream parameters.
\r
8351 if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
\r
8352 else stream_.userInterleaved = true;
\r
8353 stream_.deviceInterleaved[mode] = true;
\r
8354 stream_.nBuffers = 1;
\r
8355 stream_.doByteSwap[mode] = false;
\r
8356 stream_.nUserChannels[mode] = channels;
\r
8357 stream_.nDeviceChannels[mode] = channels + firstChannel;
\r
8358 stream_.channelOffset[mode] = 0;
\r
8359 std::string streamName = "RtAudio";
\r
8361 // Set flags for buffer conversion.
\r
8362 stream_.doConvertBuffer[mode] = false;
\r
8363 if ( stream_.userFormat != stream_.deviceFormat[mode] )
\r
8364 stream_.doConvertBuffer[mode] = true;
\r
8365 if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
\r
8366 stream_.doConvertBuffer[mode] = true;
\r
8368 // Allocate necessary internal buffers.
\r
8369 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
\r
8370 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
\r
8371 if ( stream_.userBuffer[mode] == NULL ) {
\r
8372 errorText_ = "RtApiPulse::probeDeviceOpen: error allocating user buffer memory.";
\r
8375 stream_.bufferSize = *bufferSize;
\r
8377 if ( stream_.doConvertBuffer[mode] ) {
\r
8379 bool makeBuffer = true;
\r
8380 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
\r
8381 if ( mode == INPUT ) {
\r
8382 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
\r
8383 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
\r
8384 if ( bufferBytes <= bytesOut ) makeBuffer = false;
\r
8388 if ( makeBuffer ) {
\r
8389 bufferBytes *= *bufferSize;
\r
8390 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
\r
8391 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
\r
8392 if ( stream_.deviceBuffer == NULL ) {
\r
8393 errorText_ = "RtApiPulse::probeDeviceOpen: error allocating device buffer memory.";
\r
8399 stream_.device[mode] = device;
\r
8401 // Setup the buffer conversion information structure.
\r
8402 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
\r
8404 if ( !stream_.apiHandle ) {
\r
8405 PulseAudioHandle *pah = new PulseAudioHandle;
\r
8407 errorText_ = "RtApiPulse::probeDeviceOpen: error allocating memory for handle.";
\r
8411 stream_.apiHandle = pah;
\r
8412 if ( pthread_cond_init( &pah->runnable_cv, NULL ) != 0 ) {
\r
8413 errorText_ = "RtApiPulse::probeDeviceOpen: error creating condition variable.";
\r
8417 pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
\r
8420 if ( !options->streamName.empty() ) streamName = options->streamName;
\r
8423 pa_buffer_attr buffer_attr;
\r
8424 buffer_attr.fragsize = bufferBytes;
\r
8425 buffer_attr.maxlength = -1;
\r
8427 pah->s_rec = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_RECORD, NULL, "Record", &ss, NULL, &buffer_attr, &error );
\r
8428 if ( !pah->s_rec ) {
\r
8429 errorText_ = "RtApiPulse::probeDeviceOpen: error connecting input to PulseAudio server.";
\r
8434 pah->s_play = pa_simple_new( NULL, "RtAudio", PA_STREAM_PLAYBACK, NULL, "Playback", &ss, NULL, NULL, &error );
\r
8435 if ( !pah->s_play ) {
\r
8436 errorText_ = "RtApiPulse::probeDeviceOpen: error connecting output to PulseAudio server.";
\r
8444 if ( stream_.mode == UNINITIALIZED )
\r
8445 stream_.mode = mode;
\r
8446 else if ( stream_.mode == mode )
\r
8449 stream_.mode = DUPLEX;
\r
8451 if ( !stream_.callbackInfo.isRunning ) {
\r
8452 stream_.callbackInfo.object = this;
\r
8453 stream_.callbackInfo.isRunning = true;
\r
8454 if ( pthread_create( &pah->thread, NULL, pulseaudio_callback, (void *)&stream_.callbackInfo) != 0 ) {
\r
8455 errorText_ = "RtApiPulse::probeDeviceOpen: error creating thread.";
\r
8460 stream_.state = STREAM_STOPPED;
\r
8464 if ( pah && stream_.callbackInfo.isRunning ) {
\r
8465 pthread_cond_destroy( &pah->runnable_cv );
\r
8467 stream_.apiHandle = 0;
\r
8470 for ( int i=0; i<2; i++ ) {
\r
8471 if ( stream_.userBuffer[i] ) {
\r
8472 free( stream_.userBuffer[i] );
\r
8473 stream_.userBuffer[i] = 0;
\r
8477 if ( stream_.deviceBuffer ) {
\r
8478 free( stream_.deviceBuffer );
\r
8479 stream_.deviceBuffer = 0;
\r
8485 //******************** End of __LINUX_PULSE__ *********************//
\r
8488 #if defined(__LINUX_OSS__)
\r
8490 #include <unistd.h>
\r
8491 #include <sys/ioctl.h>
\r
8492 #include <unistd.h>
\r
8493 #include <fcntl.h>
\r
8494 #include <sys/soundcard.h>
\r
8495 #include <errno.h>
\r
8498 static void *ossCallbackHandler(void * ptr);
\r
8500 // A structure to hold various information related to the OSS API
\r
8501 // implementation.
\r
8502 struct OssHandle {
\r
8503 int id[2]; // device ids
\r
8506 pthread_cond_t runnable;
\r
8509 :triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
\r
8512 RtApiOss :: RtApiOss()
\r
8514 // Nothing to do here.
\r
8517 RtApiOss :: ~RtApiOss()
\r
8519 if ( stream_.state != STREAM_CLOSED ) closeStream();
\r
8522 unsigned int RtApiOss :: getDeviceCount( void )
\r
8524 int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
\r
8525 if ( mixerfd == -1 ) {
\r
8526 errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'.";
\r
8527 error( RtAudioError::WARNING );
\r
8531 oss_sysinfo sysinfo;
\r
8532 if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) {
\r
8534 errorText_ = "RtApiOss::getDeviceCount: error getting sysinfo, OSS version >= 4.0 is required.";
\r
8535 error( RtAudioError::WARNING );
\r
8540 return sysinfo.numaudios;
\r
8543 RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device )
\r
8545 RtAudio::DeviceInfo info;
\r
8546 info.probed = false;
\r
8548 int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
\r
8549 if ( mixerfd == -1 ) {
\r
8550 errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'.";
\r
8551 error( RtAudioError::WARNING );
\r
8555 oss_sysinfo sysinfo;
\r
8556 int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
\r
8557 if ( result == -1 ) {
\r
8559 errorText_ = "RtApiOss::getDeviceInfo: error getting sysinfo, OSS version >= 4.0 is required.";
\r
8560 error( RtAudioError::WARNING );
\r
8564 unsigned nDevices = sysinfo.numaudios;
\r
8565 if ( nDevices == 0 ) {
\r
8567 errorText_ = "RtApiOss::getDeviceInfo: no devices found!";
\r
8568 error( RtAudioError::INVALID_USE );
\r
8572 if ( device >= nDevices ) {
\r
8574 errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!";
\r
8575 error( RtAudioError::INVALID_USE );
\r
8579 oss_audioinfo ainfo;
\r
8580 ainfo.dev = device;
\r
8581 result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
\r
8583 if ( result == -1 ) {
\r
8584 errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
\r
8585 errorText_ = errorStream_.str();
\r
8586 error( RtAudioError::WARNING );
\r
8591 if ( ainfo.caps & PCM_CAP_OUTPUT ) info.outputChannels = ainfo.max_channels;
\r
8592 if ( ainfo.caps & PCM_CAP_INPUT ) info.inputChannels = ainfo.max_channels;
\r
8593 if ( ainfo.caps & PCM_CAP_DUPLEX ) {
\r
8594 if ( info.outputChannels > 0 && info.inputChannels > 0 && ainfo.caps & PCM_CAP_DUPLEX )
\r
8595 info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
\r
8598 // Probe data formats ... do for input
\r
8599 unsigned long mask = ainfo.iformats;
\r
8600 if ( mask & AFMT_S16_LE || mask & AFMT_S16_BE )
\r
8601 info.nativeFormats |= RTAUDIO_SINT16;
\r
8602 if ( mask & AFMT_S8 )
\r
8603 info.nativeFormats |= RTAUDIO_SINT8;
\r
8604 if ( mask & AFMT_S32_LE || mask & AFMT_S32_BE )
\r
8605 info.nativeFormats |= RTAUDIO_SINT32;
\r
8606 if ( mask & AFMT_FLOAT )
\r
8607 info.nativeFormats |= RTAUDIO_FLOAT32;
\r
8608 if ( mask & AFMT_S24_LE || mask & AFMT_S24_BE )
\r
8609 info.nativeFormats |= RTAUDIO_SINT24;
\r
8611 // Check that we have at least one supported format
\r
8612 if ( info.nativeFormats == 0 ) {
\r
8613 errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio.";
\r
8614 errorText_ = errorStream_.str();
\r
8615 error( RtAudioError::WARNING );
\r
8619 // Probe the supported sample rates.
\r
8620 info.sampleRates.clear();
\r
8621 if ( ainfo.nrates ) {
\r
8622 for ( unsigned int i=0; i<ainfo.nrates; i++ ) {
\r
8623 for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
\r
8624 if ( ainfo.rates[i] == SAMPLE_RATES[k] ) {
\r
8625 info.sampleRates.push_back( SAMPLE_RATES[k] );
\r
8632 // Check min and max rate values;
\r
8633 for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
\r
8634 if ( ainfo.min_rate <= (int) SAMPLE_RATES[k] && ainfo.max_rate >= (int) SAMPLE_RATES[k] )
\r
8635 info.sampleRates.push_back( SAMPLE_RATES[k] );
\r
8639 if ( info.sampleRates.size() == 0 ) {
\r
8640 errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ").";
\r
8641 errorText_ = errorStream_.str();
\r
8642 error( RtAudioError::WARNING );
\r
8645 info.probed = true;
\r
8646 info.name = ainfo.name;
\r
8653 bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
\r
8654 unsigned int firstChannel, unsigned int sampleRate,
\r
8655 RtAudioFormat format, unsigned int *bufferSize,
\r
8656 RtAudio::StreamOptions *options )
\r
8658 int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
\r
8659 if ( mixerfd == -1 ) {
\r
8660 errorText_ = "RtApiOss::probeDeviceOpen: error opening '/dev/mixer'.";
\r
8664 oss_sysinfo sysinfo;
\r
8665 int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
\r
8666 if ( result == -1 ) {
\r
8668 errorText_ = "RtApiOss::probeDeviceOpen: error getting sysinfo, OSS version >= 4.0 is required.";
\r
8672 unsigned nDevices = sysinfo.numaudios;
\r
8673 if ( nDevices == 0 ) {
\r
8674 // This should not happen because a check is made before this function is called.
\r
8676 errorText_ = "RtApiOss::probeDeviceOpen: no devices found!";
\r
8680 if ( device >= nDevices ) {
\r
8681 // This should not happen because a check is made before this function is called.
\r
8683 errorText_ = "RtApiOss::probeDeviceOpen: device ID is invalid!";
\r
8687 oss_audioinfo ainfo;
\r
8688 ainfo.dev = device;
\r
8689 result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
\r
8691 if ( result == -1 ) {
\r
8692 errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
\r
8693 errorText_ = errorStream_.str();
\r
8697 // Check if device supports input or output
\r
8698 if ( ( mode == OUTPUT && !( ainfo.caps & PCM_CAP_OUTPUT ) ) ||
\r
8699 ( mode == INPUT && !( ainfo.caps & PCM_CAP_INPUT ) ) ) {
\r
8700 if ( mode == OUTPUT )
\r
8701 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support output.";
\r
8703 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support input.";
\r
8704 errorText_ = errorStream_.str();
\r
8709 OssHandle *handle = (OssHandle *) stream_.apiHandle;
\r
8710 if ( mode == OUTPUT )
\r
8711 flags |= O_WRONLY;
\r
8712 else { // mode == INPUT
\r
8713 if (stream_.mode == OUTPUT && stream_.device[0] == device) {
\r
8714 // We just set the same device for playback ... close and reopen for duplex (OSS only).
\r
8715 close( handle->id[0] );
\r
8716 handle->id[0] = 0;
\r
8717 if ( !( ainfo.caps & PCM_CAP_DUPLEX ) ) {
\r
8718 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support duplex mode.";
\r
8719 errorText_ = errorStream_.str();
\r
8722 // Check that the number previously set channels is the same.
\r
8723 if ( stream_.nUserChannels[0] != channels ) {
\r
8724 errorStream_ << "RtApiOss::probeDeviceOpen: input/output channels must be equal for OSS duplex device (" << ainfo.name << ").";
\r
8725 errorText_ = errorStream_.str();
\r
8731 flags |= O_RDONLY;
\r
8734 // Set exclusive access if specified.
\r
8735 if ( options && options->flags & RTAUDIO_HOG_DEVICE ) flags |= O_EXCL;
\r
8737 // Try to open the device.
\r
8739 fd = open( ainfo.devnode, flags, 0 );
\r
8741 if ( errno == EBUSY )
\r
8742 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") is busy.";
\r
8744 errorStream_ << "RtApiOss::probeDeviceOpen: error opening device (" << ainfo.name << ").";
\r
8745 errorText_ = errorStream_.str();
\r
8749 // For duplex operation, specifically set this mode (this doesn't seem to work).
\r
8751 if ( flags | O_RDWR ) {
\r
8752 result = ioctl( fd, SNDCTL_DSP_SETDUPLEX, NULL );
\r
8753 if ( result == -1) {
\r
8754 errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ").";
\r
8755 errorText_ = errorStream_.str();
\r
8761 // Check the device channel support.
\r
8762 stream_.nUserChannels[mode] = channels;
\r
8763 if ( ainfo.max_channels < (int)(channels + firstChannel) ) {
\r
8765 errorStream_ << "RtApiOss::probeDeviceOpen: the device (" << ainfo.name << ") does not support requested channel parameters.";
\r
8766 errorText_ = errorStream_.str();
\r
8770 // Set the number of channels.
\r
8771 int deviceChannels = channels + firstChannel;
\r
8772 result = ioctl( fd, SNDCTL_DSP_CHANNELS, &deviceChannels );
\r
8773 if ( result == -1 || deviceChannels < (int)(channels + firstChannel) ) {
\r
8775 errorStream_ << "RtApiOss::probeDeviceOpen: error setting channel parameters on device (" << ainfo.name << ").";
\r
8776 errorText_ = errorStream_.str();
\r
8779 stream_.nDeviceChannels[mode] = deviceChannels;
\r
8781 // Get the data format mask
\r
8783 result = ioctl( fd, SNDCTL_DSP_GETFMTS, &mask );
\r
8784 if ( result == -1 ) {
\r
8786 errorStream_ << "RtApiOss::probeDeviceOpen: error getting device (" << ainfo.name << ") data formats.";
\r
8787 errorText_ = errorStream_.str();
\r
8791 // Determine how to set the device format.
\r
8792 stream_.userFormat = format;
\r
8793 int deviceFormat = -1;
\r
8794 stream_.doByteSwap[mode] = false;
\r
8795 if ( format == RTAUDIO_SINT8 ) {
\r
8796 if ( mask & AFMT_S8 ) {
\r
8797 deviceFormat = AFMT_S8;
\r
8798 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
\r
8801 else if ( format == RTAUDIO_SINT16 ) {
\r
8802 if ( mask & AFMT_S16_NE ) {
\r
8803 deviceFormat = AFMT_S16_NE;
\r
8804 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
\r
8806 else if ( mask & AFMT_S16_OE ) {
\r
8807 deviceFormat = AFMT_S16_OE;
\r
8808 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
\r
8809 stream_.doByteSwap[mode] = true;
\r
8812 else if ( format == RTAUDIO_SINT24 ) {
\r
8813 if ( mask & AFMT_S24_NE ) {
\r
8814 deviceFormat = AFMT_S24_NE;
\r
8815 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
\r
8817 else if ( mask & AFMT_S24_OE ) {
\r
8818 deviceFormat = AFMT_S24_OE;
\r
8819 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
\r
8820 stream_.doByteSwap[mode] = true;
\r
8823 else if ( format == RTAUDIO_SINT32 ) {
\r
8824 if ( mask & AFMT_S32_NE ) {
\r
8825 deviceFormat = AFMT_S32_NE;
\r
8826 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
\r
8828 else if ( mask & AFMT_S32_OE ) {
\r
8829 deviceFormat = AFMT_S32_OE;
\r
8830 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
\r
8831 stream_.doByteSwap[mode] = true;
\r
8835 if ( deviceFormat == -1 ) {
\r
8836 // The user requested format is not natively supported by the device.
\r
8837 if ( mask & AFMT_S16_NE ) {
\r
8838 deviceFormat = AFMT_S16_NE;
\r
8839 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
\r
8841 else if ( mask & AFMT_S32_NE ) {
\r
8842 deviceFormat = AFMT_S32_NE;
\r
8843 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
\r
8845 else if ( mask & AFMT_S24_NE ) {
\r
8846 deviceFormat = AFMT_S24_NE;
\r
8847 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
\r
8849 else if ( mask & AFMT_S16_OE ) {
\r
8850 deviceFormat = AFMT_S16_OE;
\r
8851 stream_.deviceFormat[mode] = RTAUDIO_SINT16;
\r
8852 stream_.doByteSwap[mode] = true;
\r
8854 else if ( mask & AFMT_S32_OE ) {
\r
8855 deviceFormat = AFMT_S32_OE;
\r
8856 stream_.deviceFormat[mode] = RTAUDIO_SINT32;
\r
8857 stream_.doByteSwap[mode] = true;
\r
8859 else if ( mask & AFMT_S24_OE ) {
\r
8860 deviceFormat = AFMT_S24_OE;
\r
8861 stream_.deviceFormat[mode] = RTAUDIO_SINT24;
\r
8862 stream_.doByteSwap[mode] = true;
\r
8864 else if ( mask & AFMT_S8) {
\r
8865 deviceFormat = AFMT_S8;
\r
8866 stream_.deviceFormat[mode] = RTAUDIO_SINT8;
\r
8870 if ( stream_.deviceFormat[mode] == 0 ) {
\r
8871 // This really shouldn't happen ...
\r
8873 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") data format not supported by RtAudio.";
\r
8874 errorText_ = errorStream_.str();
\r
8878 // Set the data format.
\r
8879 int temp = deviceFormat;
\r
8880 result = ioctl( fd, SNDCTL_DSP_SETFMT, &deviceFormat );
\r
8881 if ( result == -1 || deviceFormat != temp ) {
\r
8883 errorStream_ << "RtApiOss::probeDeviceOpen: error setting data format on device (" << ainfo.name << ").";
\r
8884 errorText_ = errorStream_.str();
\r
8888 // Attempt to set the buffer size. According to OSS, the minimum
\r
8889 // number of buffers is two. The supposed minimum buffer size is 16
\r
8890 // bytes, so that will be our lower bound. The argument to this
\r
8891 // call is in the form 0xMMMMSSSS (hex), where the buffer size (in
\r
8892 // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM.
\r
8893 // We'll check the actual value used near the end of the setup
\r
8895 int ossBufferBytes = *bufferSize * formatBytes( stream_.deviceFormat[mode] ) * deviceChannels;
\r
8896 if ( ossBufferBytes < 16 ) ossBufferBytes = 16;
\r
8898 if ( options ) buffers = options->numberOfBuffers;
\r
8899 if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) buffers = 2;
\r
8900 if ( buffers < 2 ) buffers = 3;
\r
8901 temp = ((int) buffers << 16) + (int)( log10( (double)ossBufferBytes ) / log10( 2.0 ) );
\r
8902 result = ioctl( fd, SNDCTL_DSP_SETFRAGMENT, &temp );
\r
8903 if ( result == -1 ) {
\r
8905 errorStream_ << "RtApiOss::probeDeviceOpen: error setting buffer size on device (" << ainfo.name << ").";
\r
8906 errorText_ = errorStream_.str();
\r
8909 stream_.nBuffers = buffers;
\r
8911 // Save buffer size (in sample frames).
\r
8912 *bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels );
\r
8913 stream_.bufferSize = *bufferSize;
\r
8915 // Set the sample rate.
\r
8916 int srate = sampleRate;
\r
8917 result = ioctl( fd, SNDCTL_DSP_SPEED, &srate );
\r
8918 if ( result == -1 ) {
\r
8920 errorStream_ << "RtApiOss::probeDeviceOpen: error setting sample rate (" << sampleRate << ") on device (" << ainfo.name << ").";
\r
8921 errorText_ = errorStream_.str();
\r
8925 // Verify the sample rate setup worked.
\r
8926 if ( abs( srate - sampleRate ) > 100 ) {
\r
8928 errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ").";
\r
8929 errorText_ = errorStream_.str();
\r
8932 stream_.sampleRate = sampleRate;
\r
8934 if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device) {
\r
8935 // We're doing duplex setup here.
\r
8936 stream_.deviceFormat[0] = stream_.deviceFormat[1];
\r
8937 stream_.nDeviceChannels[0] = deviceChannels;
\r
8940 // Set interleaving parameters.
\r
8941 stream_.userInterleaved = true;
\r
8942 stream_.deviceInterleaved[mode] = true;
\r
8943 if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
\r
8944 stream_.userInterleaved = false;
\r
8946 // Set flags for buffer conversion
\r
8947 stream_.doConvertBuffer[mode] = false;
\r
8948 if ( stream_.userFormat != stream_.deviceFormat[mode] )
\r
8949 stream_.doConvertBuffer[mode] = true;
\r
8950 if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
\r
8951 stream_.doConvertBuffer[mode] = true;
\r
8952 if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
\r
8953 stream_.nUserChannels[mode] > 1 )
\r
8954 stream_.doConvertBuffer[mode] = true;
\r
8956 // Allocate the stream handles if necessary and then save.
\r
8957 if ( stream_.apiHandle == 0 ) {
\r
8959 handle = new OssHandle;
\r
8961 catch ( std::bad_alloc& ) {
\r
8962 errorText_ = "RtApiOss::probeDeviceOpen: error allocating OssHandle memory.";
\r
8966 if ( pthread_cond_init( &handle->runnable, NULL ) ) {
\r
8967 errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable.";
\r
8971 stream_.apiHandle = (void *) handle;
\r
8974 handle = (OssHandle *) stream_.apiHandle;
\r
8976 handle->id[mode] = fd;
\r
8978 // Allocate necessary internal buffers.
\r
8979 unsigned long bufferBytes;
\r
8980 bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
\r
8981 stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
\r
8982 if ( stream_.userBuffer[mode] == NULL ) {
\r
8983 errorText_ = "RtApiOss::probeDeviceOpen: error allocating user buffer memory.";
\r
8987 if ( stream_.doConvertBuffer[mode] ) {
\r
8989 bool makeBuffer = true;
\r
8990 bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
\r
8991 if ( mode == INPUT ) {
\r
8992 if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
\r
8993 unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
\r
8994 if ( bufferBytes <= bytesOut ) makeBuffer = false;
\r
8998 if ( makeBuffer ) {
\r
8999 bufferBytes *= *bufferSize;
\r
9000 if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
\r
9001 stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
\r
9002 if ( stream_.deviceBuffer == NULL ) {
\r
9003 errorText_ = "RtApiOss::probeDeviceOpen: error allocating device buffer memory.";
\r
9009 stream_.device[mode] = device;
\r
9010 stream_.state = STREAM_STOPPED;
\r
9012 // Setup the buffer conversion information structure.
\r
9013 if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
\r
9015 // Setup thread if necessary.
\r
9016 if ( stream_.mode == OUTPUT && mode == INPUT ) {
\r
9017 // We had already set up an output stream.
\r
9018 stream_.mode = DUPLEX;
\r
9019 if ( stream_.device[0] == device ) handle->id[0] = fd;
\r
9022 stream_.mode = mode;
\r
9024 // Setup callback thread.
\r
9025 stream_.callbackInfo.object = (void *) this;
\r
9027 // Set the thread attributes for joinable and realtime scheduling
\r
9028 // priority. The higher priority will only take affect if the
\r
9029 // program is run as root or suid.
\r
9030 pthread_attr_t attr;
\r
9031 pthread_attr_init( &attr );
\r
9032 pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
\r
9033 #ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
\r
9034 if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
\r
9035 struct sched_param param;
\r
9036 int priority = options->priority;
\r
9037 int min = sched_get_priority_min( SCHED_RR );
\r
9038 int max = sched_get_priority_max( SCHED_RR );
\r
9039 if ( priority < min ) priority = min;
\r
9040 else if ( priority > max ) priority = max;
\r
9041 param.sched_priority = priority;
\r
9042 pthread_attr_setschedparam( &attr, ¶m );
\r
9043 pthread_attr_setschedpolicy( &attr, SCHED_RR );
\r
9046 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
\r
9048 pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
\r
9051 stream_.callbackInfo.isRunning = true;
\r
9052 result = pthread_create( &stream_.callbackInfo.thread, &attr, ossCallbackHandler, &stream_.callbackInfo );
\r
9053 pthread_attr_destroy( &attr );
\r
9055 stream_.callbackInfo.isRunning = false;
\r
9056 errorText_ = "RtApiOss::error creating callback thread!";
\r
9065 pthread_cond_destroy( &handle->runnable );
\r
9066 if ( handle->id[0] ) close( handle->id[0] );
\r
9067 if ( handle->id[1] ) close( handle->id[1] );
\r
9069 stream_.apiHandle = 0;
\r
9072 for ( int i=0; i<2; i++ ) {
\r
9073 if ( stream_.userBuffer[i] ) {
\r
9074 free( stream_.userBuffer[i] );
\r
9075 stream_.userBuffer[i] = 0;
\r
9079 if ( stream_.deviceBuffer ) {
\r
9080 free( stream_.deviceBuffer );
\r
9081 stream_.deviceBuffer = 0;
\r
9087 void RtApiOss :: closeStream()
\r
9089 if ( stream_.state == STREAM_CLOSED ) {
\r
9090 errorText_ = "RtApiOss::closeStream(): no open stream to close!";
\r
9091 error( RtAudioError::WARNING );
\r
9095 OssHandle *handle = (OssHandle *) stream_.apiHandle;
\r
9096 stream_.callbackInfo.isRunning = false;
\r
9097 MUTEX_LOCK( &stream_.mutex );
\r
9098 if ( stream_.state == STREAM_STOPPED )
\r
9099 pthread_cond_signal( &handle->runnable );
\r
9100 MUTEX_UNLOCK( &stream_.mutex );
\r
9101 pthread_join( stream_.callbackInfo.thread, NULL );
\r
9103 if ( stream_.state == STREAM_RUNNING ) {
\r
9104 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
\r
9105 ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
\r
9107 ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
\r
9108 stream_.state = STREAM_STOPPED;
\r
9112 pthread_cond_destroy( &handle->runnable );
\r
9113 if ( handle->id[0] ) close( handle->id[0] );
\r
9114 if ( handle->id[1] ) close( handle->id[1] );
\r
9116 stream_.apiHandle = 0;
\r
9119 for ( int i=0; i<2; i++ ) {
\r
9120 if ( stream_.userBuffer[i] ) {
\r
9121 free( stream_.userBuffer[i] );
\r
9122 stream_.userBuffer[i] = 0;
\r
9126 if ( stream_.deviceBuffer ) {
\r
9127 free( stream_.deviceBuffer );
\r
9128 stream_.deviceBuffer = 0;
\r
9131 stream_.mode = UNINITIALIZED;
\r
9132 stream_.state = STREAM_CLOSED;
\r
9135 void RtApiOss :: startStream()
\r
9138 if ( stream_.state == STREAM_RUNNING ) {
\r
9139 errorText_ = "RtApiOss::startStream(): the stream is already running!";
\r
9140 error( RtAudioError::WARNING );
\r
9144 MUTEX_LOCK( &stream_.mutex );
\r
9146 stream_.state = STREAM_RUNNING;
\r
9148 // No need to do anything else here ... OSS automatically starts
\r
9149 // when fed samples.
\r
9151 MUTEX_UNLOCK( &stream_.mutex );
\r
9153 OssHandle *handle = (OssHandle *) stream_.apiHandle;
\r
9154 pthread_cond_signal( &handle->runnable );
\r
9157 void RtApiOss :: stopStream()
\r
9160 if ( stream_.state == STREAM_STOPPED ) {
\r
9161 errorText_ = "RtApiOss::stopStream(): the stream is already stopped!";
\r
9162 error( RtAudioError::WARNING );
\r
9166 MUTEX_LOCK( &stream_.mutex );
\r
9168 // The state might change while waiting on a mutex.
\r
9169 if ( stream_.state == STREAM_STOPPED ) {
\r
9170 MUTEX_UNLOCK( &stream_.mutex );
\r
9175 OssHandle *handle = (OssHandle *) stream_.apiHandle;
\r
9176 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
\r
9178 // Flush the output with zeros a few times.
\r
9181 RtAudioFormat format;
\r
9183 if ( stream_.doConvertBuffer[0] ) {
\r
9184 buffer = stream_.deviceBuffer;
\r
9185 samples = stream_.bufferSize * stream_.nDeviceChannels[0];
\r
9186 format = stream_.deviceFormat[0];
\r
9189 buffer = stream_.userBuffer[0];
\r
9190 samples = stream_.bufferSize * stream_.nUserChannels[0];
\r
9191 format = stream_.userFormat;
\r
9194 memset( buffer, 0, samples * formatBytes(format) );
\r
9195 for ( unsigned int i=0; i<stream_.nBuffers+1; i++ ) {
\r
9196 result = write( handle->id[0], buffer, samples * formatBytes(format) );
\r
9197 if ( result == -1 ) {
\r
9198 errorText_ = "RtApiOss::stopStream: audio write error.";
\r
9199 error( RtAudioError::WARNING );
\r
9203 result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
\r
9204 if ( result == -1 ) {
\r
9205 errorStream_ << "RtApiOss::stopStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
\r
9206 errorText_ = errorStream_.str();
\r
9209 handle->triggered = false;
\r
9212 if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
\r
9213 result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
\r
9214 if ( result == -1 ) {
\r
9215 errorStream_ << "RtApiOss::stopStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
\r
9216 errorText_ = errorStream_.str();
\r
9222 stream_.state = STREAM_STOPPED;
\r
9223 MUTEX_UNLOCK( &stream_.mutex );
\r
9225 if ( result != -1 ) return;
\r
9226 error( RtAudioError::SYSTEM_ERROR );
\r
9229 void RtApiOss :: abortStream()
\r
9232 if ( stream_.state == STREAM_STOPPED ) {
\r
9233 errorText_ = "RtApiOss::abortStream(): the stream is already stopped!";
\r
9234 error( RtAudioError::WARNING );
\r
9238 MUTEX_LOCK( &stream_.mutex );
\r
9240 // The state might change while waiting on a mutex.
\r
9241 if ( stream_.state == STREAM_STOPPED ) {
\r
9242 MUTEX_UNLOCK( &stream_.mutex );
\r
9247 OssHandle *handle = (OssHandle *) stream_.apiHandle;
\r
9248 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
\r
9249 result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
\r
9250 if ( result == -1 ) {
\r
9251 errorStream_ << "RtApiOss::abortStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
\r
9252 errorText_ = errorStream_.str();
\r
9255 handle->triggered = false;
\r
9258 if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
\r
9259 result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
\r
9260 if ( result == -1 ) {
\r
9261 errorStream_ << "RtApiOss::abortStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
\r
9262 errorText_ = errorStream_.str();
\r
9268 stream_.state = STREAM_STOPPED;
\r
9269 MUTEX_UNLOCK( &stream_.mutex );
\r
9271 if ( result != -1 ) return;
\r
9272 error( RtAudioError::SYSTEM_ERROR );
\r
9275 void RtApiOss :: callbackEvent()
\r
9277 OssHandle *handle = (OssHandle *) stream_.apiHandle;
\r
9278 if ( stream_.state == STREAM_STOPPED ) {
\r
9279 MUTEX_LOCK( &stream_.mutex );
\r
9280 pthread_cond_wait( &handle->runnable, &stream_.mutex );
\r
9281 if ( stream_.state != STREAM_RUNNING ) {
\r
9282 MUTEX_UNLOCK( &stream_.mutex );
\r
9285 MUTEX_UNLOCK( &stream_.mutex );
\r
9288 if ( stream_.state == STREAM_CLOSED ) {
\r
9289 errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!";
\r
9290 error( RtAudioError::WARNING );
\r
9294 // Invoke user callback to get fresh output data.
\r
9295 int doStopStream = 0;
\r
9296 RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
\r
9297 double streamTime = getStreamTime();
\r
9298 RtAudioStreamStatus status = 0;
\r
9299 if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
\r
9300 status |= RTAUDIO_OUTPUT_UNDERFLOW;
\r
9301 handle->xrun[0] = false;
\r
9303 if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
\r
9304 status |= RTAUDIO_INPUT_OVERFLOW;
\r
9305 handle->xrun[1] = false;
\r
9307 doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
\r
9308 stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
\r
9309 if ( doStopStream == 2 ) {
\r
9310 this->abortStream();
\r
9314 MUTEX_LOCK( &stream_.mutex );
\r
9316 // The state might change while waiting on a mutex.
\r
9317 if ( stream_.state == STREAM_STOPPED ) goto unlock;
\r
9322 RtAudioFormat format;
\r
9324 if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
\r
9326 // Setup parameters and do buffer conversion if necessary.
\r
9327 if ( stream_.doConvertBuffer[0] ) {
\r
9328 buffer = stream_.deviceBuffer;
\r
9329 convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
\r
9330 samples = stream_.bufferSize * stream_.nDeviceChannels[0];
\r
9331 format = stream_.deviceFormat[0];
\r
9334 buffer = stream_.userBuffer[0];
\r
9335 samples = stream_.bufferSize * stream_.nUserChannels[0];
\r
9336 format = stream_.userFormat;
\r
9339 // Do byte swapping if necessary.
\r
9340 if ( stream_.doByteSwap[0] )
\r
9341 byteSwapBuffer( buffer, samples, format );
\r
9343 if ( stream_.mode == DUPLEX && handle->triggered == false ) {
\r
9345 ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
\r
9346 result = write( handle->id[0], buffer, samples * formatBytes(format) );
\r
9347 trig = PCM_ENABLE_INPUT|PCM_ENABLE_OUTPUT;
\r
9348 ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
\r
9349 handle->triggered = true;
\r
9352 // Write samples to device.
\r
9353 result = write( handle->id[0], buffer, samples * formatBytes(format) );
\r
9355 if ( result == -1 ) {
\r
9356 // We'll assume this is an underrun, though there isn't a
\r
9357 // specific means for determining that.
\r
9358 handle->xrun[0] = true;
\r
9359 errorText_ = "RtApiOss::callbackEvent: audio write error.";
\r
9360 error( RtAudioError::WARNING );
\r
9361 // Continue on to input section.
\r
9365 if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
\r
9367 // Setup parameters.
\r
9368 if ( stream_.doConvertBuffer[1] ) {
\r
9369 buffer = stream_.deviceBuffer;
\r
9370 samples = stream_.bufferSize * stream_.nDeviceChannels[1];
\r
9371 format = stream_.deviceFormat[1];
\r
9374 buffer = stream_.userBuffer[1];
\r
9375 samples = stream_.bufferSize * stream_.nUserChannels[1];
\r
9376 format = stream_.userFormat;
\r
9379 // Read samples from device.
\r
9380 result = read( handle->id[1], buffer, samples * formatBytes(format) );
\r
9382 if ( result == -1 ) {
\r
9383 // We'll assume this is an overrun, though there isn't a
\r
9384 // specific means for determining that.
\r
9385 handle->xrun[1] = true;
\r
9386 errorText_ = "RtApiOss::callbackEvent: audio read error.";
\r
9387 error( RtAudioError::WARNING );
\r
9391 // Do byte swapping if necessary.
\r
9392 if ( stream_.doByteSwap[1] )
\r
9393 byteSwapBuffer( buffer, samples, format );
\r
9395 // Do buffer conversion if necessary.
\r
9396 if ( stream_.doConvertBuffer[1] )
\r
9397 convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
\r
9401 MUTEX_UNLOCK( &stream_.mutex );
\r
9403 RtApi::tickStreamTime();
\r
9404 if ( doStopStream == 1 ) this->stopStream();
\r
9407 static void *ossCallbackHandler( void *ptr )
\r
9409 CallbackInfo *info = (CallbackInfo *) ptr;
\r
9410 RtApiOss *object = (RtApiOss *) info->object;
\r
9411 bool *isRunning = &info->isRunning;
\r
9413 while ( *isRunning == true ) {
\r
9414 pthread_testcancel();
\r
9415 object->callbackEvent();
\r
9418 pthread_exit( NULL );
\r
9421 //******************** End of __LINUX_OSS__ *********************//
\r
9425 // *************************************************** //
\r
9427 // Protected common (OS-independent) RtAudio methods.
\r
9429 // *************************************************** //
\r
9431 // This method can be modified to control the behavior of error
\r
9432 // message printing.
\r
9433 void RtApi :: error( RtAudioError::Type type )
\r
9435 errorStream_.str(""); // clear the ostringstream
\r
9437 RtAudioErrorCallback errorCallback = (RtAudioErrorCallback) stream_.callbackInfo.errorCallback;
\r
9438 if ( errorCallback ) {
\r
9439 // abortStream() can generate new error messages. Ignore them. Just keep original one.
\r
9441 if ( firstErrorOccurred_ )
\r
9444 firstErrorOccurred_ = true;
\r
9445 const std::string errorMessage = errorText_;
\r
9447 if ( type != RtAudioError::WARNING && stream_.state != STREAM_STOPPED) {
\r
9448 stream_.callbackInfo.isRunning = false; // exit from the thread
\r
9452 errorCallback( type, errorMessage );
\r
9453 firstErrorOccurred_ = false;
\r
9457 if ( type == RtAudioError::WARNING && showWarnings_ == true )
\r
9458 std::cerr << '\n' << errorText_ << "\n\n";
\r
9459 else if ( type != RtAudioError::WARNING )
\r
9460 throw( RtAudioError( errorText_, type ) );
\r
9463 void RtApi :: verifyStream()
\r
9465 if ( stream_.state == STREAM_CLOSED ) {
\r
9466 errorText_ = "RtApi:: a stream is not open!";
\r
9467 error( RtAudioError::INVALID_USE );
\r
9471 void RtApi :: clearStreamInfo()
\r
9473 stream_.mode = UNINITIALIZED;
\r
9474 stream_.state = STREAM_CLOSED;
\r
9475 stream_.sampleRate = 0;
\r
9476 stream_.bufferSize = 0;
\r
9477 stream_.nBuffers = 0;
\r
9478 stream_.userFormat = 0;
\r
9479 stream_.userInterleaved = true;
\r
9480 stream_.streamTime = 0.0;
\r
9481 stream_.apiHandle = 0;
\r
9482 stream_.deviceBuffer = 0;
\r
9483 stream_.callbackInfo.callback = 0;
\r
9484 stream_.callbackInfo.userData = 0;
\r
9485 stream_.callbackInfo.isRunning = false;
\r
9486 stream_.callbackInfo.errorCallback = 0;
\r
9487 for ( int i=0; i<2; i++ ) {
\r
9488 stream_.device[i] = 11111;
\r
9489 stream_.doConvertBuffer[i] = false;
\r
9490 stream_.deviceInterleaved[i] = true;
\r
9491 stream_.doByteSwap[i] = false;
\r
9492 stream_.nUserChannels[i] = 0;
\r
9493 stream_.nDeviceChannels[i] = 0;
\r
9494 stream_.channelOffset[i] = 0;
\r
9495 stream_.deviceFormat[i] = 0;
\r
9496 stream_.latency[i] = 0;
\r
9497 stream_.userBuffer[i] = 0;
\r
9498 stream_.convertInfo[i].channels = 0;
\r
9499 stream_.convertInfo[i].inJump = 0;
\r
9500 stream_.convertInfo[i].outJump = 0;
\r
9501 stream_.convertInfo[i].inFormat = 0;
\r
9502 stream_.convertInfo[i].outFormat = 0;
\r
9503 stream_.convertInfo[i].inOffset.clear();
\r
9504 stream_.convertInfo[i].outOffset.clear();
\r
9508 unsigned int RtApi :: formatBytes( RtAudioFormat format )
\r
9510 if ( format == RTAUDIO_SINT16 )
\r
9512 else if ( format == RTAUDIO_SINT32 || format == RTAUDIO_FLOAT32 )
\r
9514 else if ( format == RTAUDIO_FLOAT64 )
\r
9516 else if ( format == RTAUDIO_SINT24 )
\r
9518 else if ( format == RTAUDIO_SINT8 )
\r
9521 errorText_ = "RtApi::formatBytes: undefined format.";
\r
9522 error( RtAudioError::WARNING );
\r
9527 void RtApi :: setConvertInfo( StreamMode mode, unsigned int firstChannel )
\r
9529 if ( mode == INPUT ) { // convert device to user buffer
\r
9530 stream_.convertInfo[mode].inJump = stream_.nDeviceChannels[1];
\r
9531 stream_.convertInfo[mode].outJump = stream_.nUserChannels[1];
\r
9532 stream_.convertInfo[mode].inFormat = stream_.deviceFormat[1];
\r
9533 stream_.convertInfo[mode].outFormat = stream_.userFormat;
\r
9535 else { // convert user to device buffer
\r
9536 stream_.convertInfo[mode].inJump = stream_.nUserChannels[0];
\r
9537 stream_.convertInfo[mode].outJump = stream_.nDeviceChannels[0];
\r
9538 stream_.convertInfo[mode].inFormat = stream_.userFormat;
\r
9539 stream_.convertInfo[mode].outFormat = stream_.deviceFormat[0];
\r
9542 if ( stream_.convertInfo[mode].inJump < stream_.convertInfo[mode].outJump )
\r
9543 stream_.convertInfo[mode].channels = stream_.convertInfo[mode].inJump;
\r
9545 stream_.convertInfo[mode].channels = stream_.convertInfo[mode].outJump;
\r
9547 // Set up the interleave/deinterleave offsets.
\r
9548 if ( stream_.deviceInterleaved[mode] != stream_.userInterleaved ) {
\r
9549 if ( ( mode == OUTPUT && stream_.deviceInterleaved[mode] ) ||
\r
9550 ( mode == INPUT && stream_.userInterleaved ) ) {
\r
9551 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
\r
9552 stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
\r
9553 stream_.convertInfo[mode].outOffset.push_back( k );
\r
9554 stream_.convertInfo[mode].inJump = 1;
\r
9558 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
\r
9559 stream_.convertInfo[mode].inOffset.push_back( k );
\r
9560 stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
\r
9561 stream_.convertInfo[mode].outJump = 1;
\r
9565 else { // no (de)interleaving
\r
9566 if ( stream_.userInterleaved ) {
\r
9567 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
\r
9568 stream_.convertInfo[mode].inOffset.push_back( k );
\r
9569 stream_.convertInfo[mode].outOffset.push_back( k );
\r
9573 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
\r
9574 stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
\r
9575 stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
\r
9576 stream_.convertInfo[mode].inJump = 1;
\r
9577 stream_.convertInfo[mode].outJump = 1;
\r
9582 // Add channel offset.
\r
9583 if ( firstChannel > 0 ) {
\r
9584 if ( stream_.deviceInterleaved[mode] ) {
\r
9585 if ( mode == OUTPUT ) {
\r
9586 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
\r
9587 stream_.convertInfo[mode].outOffset[k] += firstChannel;
\r
9590 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
\r
9591 stream_.convertInfo[mode].inOffset[k] += firstChannel;
\r
9595 if ( mode == OUTPUT ) {
\r
9596 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
\r
9597 stream_.convertInfo[mode].outOffset[k] += ( firstChannel * stream_.bufferSize );
\r
9600 for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
\r
9601 stream_.convertInfo[mode].inOffset[k] += ( firstChannel * stream_.bufferSize );
\r
9607 void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info )
\r
9609 // This function does format conversion, input/output channel compensation, and
\r
9610 // data interleaving/deinterleaving. 24-bit integers are assumed to occupy
\r
9611 // the lower three bytes of a 32-bit integer.
\r
9613 // Clear our device buffer when in/out duplex device channels are different
\r
9614 if ( outBuffer == stream_.deviceBuffer && stream_.mode == DUPLEX &&
\r
9615 ( stream_.nDeviceChannels[0] < stream_.nDeviceChannels[1] ) )
\r
9616 memset( outBuffer, 0, stream_.bufferSize * info.outJump * formatBytes( info.outFormat ) );
\r
9619 if (info.outFormat == RTAUDIO_FLOAT64) {
\r
9621 Float64 *out = (Float64 *)outBuffer;
\r
9623 if (info.inFormat == RTAUDIO_SINT8) {
\r
9624 signed char *in = (signed char *)inBuffer;
\r
9625 scale = 1.0 / 127.5;
\r
9626 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9627 for (j=0; j<info.channels; j++) {
\r
9628 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
\r
9629 out[info.outOffset[j]] += 0.5;
\r
9630 out[info.outOffset[j]] *= scale;
\r
9632 in += info.inJump;
\r
9633 out += info.outJump;
\r
9636 else if (info.inFormat == RTAUDIO_SINT16) {
\r
9637 Int16 *in = (Int16 *)inBuffer;
\r
9638 scale = 1.0 / 32767.5;
\r
9639 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9640 for (j=0; j<info.channels; j++) {
\r
9641 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
\r
9642 out[info.outOffset[j]] += 0.5;
\r
9643 out[info.outOffset[j]] *= scale;
\r
9645 in += info.inJump;
\r
9646 out += info.outJump;
\r
9649 else if (info.inFormat == RTAUDIO_SINT24) {
\r
9650 Int24 *in = (Int24 *)inBuffer;
\r
9651 scale = 1.0 / 8388607.5;
\r
9652 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9653 for (j=0; j<info.channels; j++) {
\r
9654 out[info.outOffset[j]] = (Float64) (in[info.inOffset[j]].asInt());
\r
9655 out[info.outOffset[j]] += 0.5;
\r
9656 out[info.outOffset[j]] *= scale;
\r
9658 in += info.inJump;
\r
9659 out += info.outJump;
\r
9662 else if (info.inFormat == RTAUDIO_SINT32) {
\r
9663 Int32 *in = (Int32 *)inBuffer;
\r
9664 scale = 1.0 / 2147483647.5;
\r
9665 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9666 for (j=0; j<info.channels; j++) {
\r
9667 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
\r
9668 out[info.outOffset[j]] += 0.5;
\r
9669 out[info.outOffset[j]] *= scale;
\r
9671 in += info.inJump;
\r
9672 out += info.outJump;
\r
9675 else if (info.inFormat == RTAUDIO_FLOAT32) {
\r
9676 Float32 *in = (Float32 *)inBuffer;
\r
9677 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9678 for (j=0; j<info.channels; j++) {
\r
9679 out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
\r
9681 in += info.inJump;
\r
9682 out += info.outJump;
\r
9685 else if (info.inFormat == RTAUDIO_FLOAT64) {
\r
9686 // Channel compensation and/or (de)interleaving only.
\r
9687 Float64 *in = (Float64 *)inBuffer;
\r
9688 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9689 for (j=0; j<info.channels; j++) {
\r
9690 out[info.outOffset[j]] = in[info.inOffset[j]];
\r
9692 in += info.inJump;
\r
9693 out += info.outJump;
\r
9697 else if (info.outFormat == RTAUDIO_FLOAT32) {
\r
9699 Float32 *out = (Float32 *)outBuffer;
\r
9701 if (info.inFormat == RTAUDIO_SINT8) {
\r
9702 signed char *in = (signed char *)inBuffer;
\r
9703 scale = (Float32) ( 1.0 / 127.5 );
\r
9704 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9705 for (j=0; j<info.channels; j++) {
\r
9706 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
\r
9707 out[info.outOffset[j]] += 0.5;
\r
9708 out[info.outOffset[j]] *= scale;
\r
9710 in += info.inJump;
\r
9711 out += info.outJump;
\r
9714 else if (info.inFormat == RTAUDIO_SINT16) {
\r
9715 Int16 *in = (Int16 *)inBuffer;
\r
9716 scale = (Float32) ( 1.0 / 32767.5 );
\r
9717 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9718 for (j=0; j<info.channels; j++) {
\r
9719 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
\r
9720 out[info.outOffset[j]] += 0.5;
\r
9721 out[info.outOffset[j]] *= scale;
\r
9723 in += info.inJump;
\r
9724 out += info.outJump;
\r
9727 else if (info.inFormat == RTAUDIO_SINT24) {
\r
9728 Int24 *in = (Int24 *)inBuffer;
\r
9729 scale = (Float32) ( 1.0 / 8388607.5 );
\r
9730 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9731 for (j=0; j<info.channels; j++) {
\r
9732 out[info.outOffset[j]] = (Float32) (in[info.inOffset[j]].asInt());
\r
9733 out[info.outOffset[j]] += 0.5;
\r
9734 out[info.outOffset[j]] *= scale;
\r
9736 in += info.inJump;
\r
9737 out += info.outJump;
\r
9740 else if (info.inFormat == RTAUDIO_SINT32) {
\r
9741 Int32 *in = (Int32 *)inBuffer;
\r
9742 scale = (Float32) ( 1.0 / 2147483647.5 );
\r
9743 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9744 for (j=0; j<info.channels; j++) {
\r
9745 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
\r
9746 out[info.outOffset[j]] += 0.5;
\r
9747 out[info.outOffset[j]] *= scale;
\r
9749 in += info.inJump;
\r
9750 out += info.outJump;
\r
9753 else if (info.inFormat == RTAUDIO_FLOAT32) {
\r
9754 // Channel compensation and/or (de)interleaving only.
\r
9755 Float32 *in = (Float32 *)inBuffer;
\r
9756 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9757 for (j=0; j<info.channels; j++) {
\r
9758 out[info.outOffset[j]] = in[info.inOffset[j]];
\r
9760 in += info.inJump;
\r
9761 out += info.outJump;
\r
9764 else if (info.inFormat == RTAUDIO_FLOAT64) {
\r
9765 Float64 *in = (Float64 *)inBuffer;
\r
9766 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9767 for (j=0; j<info.channels; j++) {
\r
9768 out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
\r
9770 in += info.inJump;
\r
9771 out += info.outJump;
\r
9775 else if (info.outFormat == RTAUDIO_SINT32) {
\r
9776 Int32 *out = (Int32 *)outBuffer;
\r
9777 if (info.inFormat == RTAUDIO_SINT8) {
\r
9778 signed char *in = (signed char *)inBuffer;
\r
9779 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9780 for (j=0; j<info.channels; j++) {
\r
9781 out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
\r
9782 out[info.outOffset[j]] <<= 24;
\r
9784 in += info.inJump;
\r
9785 out += info.outJump;
\r
9788 else if (info.inFormat == RTAUDIO_SINT16) {
\r
9789 Int16 *in = (Int16 *)inBuffer;
\r
9790 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9791 for (j=0; j<info.channels; j++) {
\r
9792 out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
\r
9793 out[info.outOffset[j]] <<= 16;
\r
9795 in += info.inJump;
\r
9796 out += info.outJump;
\r
9799 else if (info.inFormat == RTAUDIO_SINT24) {
\r
9800 Int24 *in = (Int24 *)inBuffer;
\r
9801 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9802 for (j=0; j<info.channels; j++) {
\r
9803 out[info.outOffset[j]] = (Int32) in[info.inOffset[j]].asInt();
\r
9804 out[info.outOffset[j]] <<= 8;
\r
9806 in += info.inJump;
\r
9807 out += info.outJump;
\r
9810 else if (info.inFormat == RTAUDIO_SINT32) {
\r
9811 // Channel compensation and/or (de)interleaving only.
\r
9812 Int32 *in = (Int32 *)inBuffer;
\r
9813 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9814 for (j=0; j<info.channels; j++) {
\r
9815 out[info.outOffset[j]] = in[info.inOffset[j]];
\r
9817 in += info.inJump;
\r
9818 out += info.outJump;
\r
9821 else if (info.inFormat == RTAUDIO_FLOAT32) {
\r
9822 Float32 *in = (Float32 *)inBuffer;
\r
9823 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9824 for (j=0; j<info.channels; j++) {
\r
9825 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
\r
9827 in += info.inJump;
\r
9828 out += info.outJump;
\r
9831 else if (info.inFormat == RTAUDIO_FLOAT64) {
\r
9832 Float64 *in = (Float64 *)inBuffer;
\r
9833 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9834 for (j=0; j<info.channels; j++) {
\r
9835 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
\r
9837 in += info.inJump;
\r
9838 out += info.outJump;
\r
9842 else if (info.outFormat == RTAUDIO_SINT24) {
\r
9843 Int24 *out = (Int24 *)outBuffer;
\r
9844 if (info.inFormat == RTAUDIO_SINT8) {
\r
9845 signed char *in = (signed char *)inBuffer;
\r
9846 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9847 for (j=0; j<info.channels; j++) {
\r
9848 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] << 16);
\r
9849 //out[info.outOffset[j]] <<= 16;
\r
9851 in += info.inJump;
\r
9852 out += info.outJump;
\r
9855 else if (info.inFormat == RTAUDIO_SINT16) {
\r
9856 Int16 *in = (Int16 *)inBuffer;
\r
9857 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9858 for (j=0; j<info.channels; j++) {
\r
9859 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] << 8);
\r
9860 //out[info.outOffset[j]] <<= 8;
\r
9862 in += info.inJump;
\r
9863 out += info.outJump;
\r
9866 else if (info.inFormat == RTAUDIO_SINT24) {
\r
9867 // Channel compensation and/or (de)interleaving only.
\r
9868 Int24 *in = (Int24 *)inBuffer;
\r
9869 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9870 for (j=0; j<info.channels; j++) {
\r
9871 out[info.outOffset[j]] = in[info.inOffset[j]];
\r
9873 in += info.inJump;
\r
9874 out += info.outJump;
\r
9877 else if (info.inFormat == RTAUDIO_SINT32) {
\r
9878 Int32 *in = (Int32 *)inBuffer;
\r
9879 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9880 for (j=0; j<info.channels; j++) {
\r
9881 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] >> 8);
\r
9882 //out[info.outOffset[j]] >>= 8;
\r
9884 in += info.inJump;
\r
9885 out += info.outJump;
\r
9888 else if (info.inFormat == RTAUDIO_FLOAT32) {
\r
9889 Float32 *in = (Float32 *)inBuffer;
\r
9890 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9891 for (j=0; j<info.channels; j++) {
\r
9892 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
\r
9894 in += info.inJump;
\r
9895 out += info.outJump;
\r
9898 else if (info.inFormat == RTAUDIO_FLOAT64) {
\r
9899 Float64 *in = (Float64 *)inBuffer;
\r
9900 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9901 for (j=0; j<info.channels; j++) {
\r
9902 out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
\r
9904 in += info.inJump;
\r
9905 out += info.outJump;
\r
9909 else if (info.outFormat == RTAUDIO_SINT16) {
\r
9910 Int16 *out = (Int16 *)outBuffer;
\r
9911 if (info.inFormat == RTAUDIO_SINT8) {
\r
9912 signed char *in = (signed char *)inBuffer;
\r
9913 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9914 for (j=0; j<info.channels; j++) {
\r
9915 out[info.outOffset[j]] = (Int16) in[info.inOffset[j]];
\r
9916 out[info.outOffset[j]] <<= 8;
\r
9918 in += info.inJump;
\r
9919 out += info.outJump;
\r
9922 else if (info.inFormat == RTAUDIO_SINT16) {
\r
9923 // Channel compensation and/or (de)interleaving only.
\r
9924 Int16 *in = (Int16 *)inBuffer;
\r
9925 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9926 for (j=0; j<info.channels; j++) {
\r
9927 out[info.outOffset[j]] = in[info.inOffset[j]];
\r
9929 in += info.inJump;
\r
9930 out += info.outJump;
\r
9933 else if (info.inFormat == RTAUDIO_SINT24) {
\r
9934 Int24 *in = (Int24 *)inBuffer;
\r
9935 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9936 for (j=0; j<info.channels; j++) {
\r
9937 out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]].asInt() >> 8);
\r
9939 in += info.inJump;
\r
9940 out += info.outJump;
\r
9943 else if (info.inFormat == RTAUDIO_SINT32) {
\r
9944 Int32 *in = (Int32 *)inBuffer;
\r
9945 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9946 for (j=0; j<info.channels; j++) {
\r
9947 out[info.outOffset[j]] = (Int16) ((in[info.inOffset[j]] >> 16) & 0x0000ffff);
\r
9949 in += info.inJump;
\r
9950 out += info.outJump;
\r
9953 else if (info.inFormat == RTAUDIO_FLOAT32) {
\r
9954 Float32 *in = (Float32 *)inBuffer;
\r
9955 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9956 for (j=0; j<info.channels; j++) {
\r
9957 out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
\r
9959 in += info.inJump;
\r
9960 out += info.outJump;
\r
9963 else if (info.inFormat == RTAUDIO_FLOAT64) {
\r
9964 Float64 *in = (Float64 *)inBuffer;
\r
9965 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9966 for (j=0; j<info.channels; j++) {
\r
9967 out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
\r
9969 in += info.inJump;
\r
9970 out += info.outJump;
\r
9974 else if (info.outFormat == RTAUDIO_SINT8) {
\r
9975 signed char *out = (signed char *)outBuffer;
\r
9976 if (info.inFormat == RTAUDIO_SINT8) {
\r
9977 // Channel compensation and/or (de)interleaving only.
\r
9978 signed char *in = (signed char *)inBuffer;
\r
9979 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9980 for (j=0; j<info.channels; j++) {
\r
9981 out[info.outOffset[j]] = in[info.inOffset[j]];
\r
9983 in += info.inJump;
\r
9984 out += info.outJump;
\r
9987 if (info.inFormat == RTAUDIO_SINT16) {
\r
9988 Int16 *in = (Int16 *)inBuffer;
\r
9989 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
9990 for (j=0; j<info.channels; j++) {
\r
9991 out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 8) & 0x00ff);
\r
9993 in += info.inJump;
\r
9994 out += info.outJump;
\r
9997 else if (info.inFormat == RTAUDIO_SINT24) {
\r
9998 Int24 *in = (Int24 *)inBuffer;
\r
9999 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
10000 for (j=0; j<info.channels; j++) {
\r
10001 out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]].asInt() >> 16);
\r
10003 in += info.inJump;
\r
10004 out += info.outJump;
\r
10007 else if (info.inFormat == RTAUDIO_SINT32) {
\r
10008 Int32 *in = (Int32 *)inBuffer;
\r
10009 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
10010 for (j=0; j<info.channels; j++) {
\r
10011 out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 24) & 0x000000ff);
\r
10013 in += info.inJump;
\r
10014 out += info.outJump;
\r
10017 else if (info.inFormat == RTAUDIO_FLOAT32) {
\r
10018 Float32 *in = (Float32 *)inBuffer;
\r
10019 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
10020 for (j=0; j<info.channels; j++) {
\r
10021 out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
\r
10023 in += info.inJump;
\r
10024 out += info.outJump;
\r
10027 else if (info.inFormat == RTAUDIO_FLOAT64) {
\r
10028 Float64 *in = (Float64 *)inBuffer;
\r
10029 for (unsigned int i=0; i<stream_.bufferSize; i++) {
\r
10030 for (j=0; j<info.channels; j++) {
\r
10031 out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
\r
10033 in += info.inJump;
\r
10034 out += info.outJump;
\r
10040 //static inline uint16_t bswap_16(uint16_t x) { return (x>>8) | (x<<8); }
\r
10041 //static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); }
\r
10042 //static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); }
\r
10044 void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format )
\r
10046 register char val;
\r
10047 register char *ptr;
\r
10050 if ( format == RTAUDIO_SINT16 ) {
\r
10051 for ( unsigned int i=0; i<samples; i++ ) {
\r
10052 // Swap 1st and 2nd bytes.
\r
10054 *(ptr) = *(ptr+1);
\r
10057 // Increment 2 bytes.
\r
10061 else if ( format == RTAUDIO_SINT32 ||
\r
10062 format == RTAUDIO_FLOAT32 ) {
\r
10063 for ( unsigned int i=0; i<samples; i++ ) {
\r
10064 // Swap 1st and 4th bytes.
\r
10066 *(ptr) = *(ptr+3);
\r
10069 // Swap 2nd and 3rd bytes.
\r
10072 *(ptr) = *(ptr+1);
\r
10075 // Increment 3 more bytes.
\r
10079 else if ( format == RTAUDIO_SINT24 ) {
\r
10080 for ( unsigned int i=0; i<samples; i++ ) {
\r
10081 // Swap 1st and 3rd bytes.
\r
10083 *(ptr) = *(ptr+2);
\r
10086 // Increment 2 more bytes.
\r
10090 else if ( format == RTAUDIO_FLOAT64 ) {
\r
10091 for ( unsigned int i=0; i<samples; i++ ) {
\r
10092 // Swap 1st and 8th bytes
\r
10094 *(ptr) = *(ptr+7);
\r
10097 // Swap 2nd and 7th bytes
\r
10100 *(ptr) = *(ptr+5);
\r
10103 // Swap 3rd and 6th bytes
\r
10106 *(ptr) = *(ptr+3);
\r
10109 // Swap 4th and 5th bytes
\r
10112 *(ptr) = *(ptr+1);
\r
10115 // Increment 5 more bytes.
\r
10121 // Indentation settings for Vim and Emacs
\r
10123 // Local Variables:
\r
10124 // c-basic-offset: 2
\r
10125 // indent-tabs-mode: nil
\r
10128 // vim: et sts=2 sw=2
\r
projects / rtaudio-cdist.git / blob