1 /************************************************************************/
3 \brief Realtime audio i/o C++ classes.
5 RtAudio provides a common API (Application Programming Interface)
6 for realtime audio input/output across Linux (native ALSA, Jack,
7 and OSS), Macintosh OS X (CoreAudio and Jack), and Windows
8 (DirectSound, ASIO and WASAPI) operating systems.
10 RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/
12 RtAudio: realtime audio i/o C++ classes
13 Copyright (c) 2001-2016 Gary P. Scavone
15 Permission is hereby granted, free of charge, to any person
16 obtaining a copy of this software and associated documentation files
17 (the "Software"), to deal in the Software without restriction,
18 including without limitation the rights to use, copy, modify, merge,
19 publish, distribute, sublicense, and/or sell copies of the Software,
20 and to permit persons to whom the Software is furnished to do so,
21 subject to the following conditions:
23 The above copyright notice and this permission notice shall be
24 included in all copies or substantial portions of the Software.
26 Any person wishing to distribute modifications to the Software is
27 asked to send the modifications to the original developer so that
28 they can be incorporated into the canonical version. This is,
29 however, not a binding provision of this license.
31 THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
32 EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
33 MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
34 IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
35 ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
36 CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
37 WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
39 /************************************************************************/
48 #define RTAUDIO_VERSION "4.1.2"
55 /*! \typedef typedef unsigned long RtAudioFormat;
56 \brief RtAudio data format type.
58 Support for signed integers and floats. Audio data fed to/from an
59 RtAudio stream is assumed to ALWAYS be in host byte order. The
60 internal routines will automatically take care of any necessary
61 byte-swapping between the host format and the soundcard. Thus,
62 endian-ness is not a concern in the following format definitions.
64 - \e RTAUDIO_SINT8: 8-bit signed integer.
65 - \e RTAUDIO_SINT16: 16-bit signed integer.
66 - \e RTAUDIO_SINT24: 24-bit signed integer.
67 - \e RTAUDIO_SINT32: 32-bit signed integer.
68 - \e RTAUDIO_FLOAT32: Normalized between plus/minus 1.0.
69 - \e RTAUDIO_FLOAT64: Normalized between plus/minus 1.0.
71 typedef unsigned long RtAudioFormat;
72 static const RtAudioFormat RTAUDIO_SINT8 = 0x1; // 8-bit signed integer.
73 static const RtAudioFormat RTAUDIO_SINT16 = 0x2; // 16-bit signed integer.
74 static const RtAudioFormat RTAUDIO_SINT24 = 0x4; // 24-bit signed integer.
75 static const RtAudioFormat RTAUDIO_SINT32 = 0x8; // 32-bit signed integer.
76 static const RtAudioFormat RTAUDIO_FLOAT32 = 0x10; // Normalized between plus/minus 1.0.
77 static const RtAudioFormat RTAUDIO_FLOAT64 = 0x20; // Normalized between plus/minus 1.0.
79 /*! \typedef typedef unsigned long RtAudioStreamFlags;
80 \brief RtAudio stream option flags.
82 The following flags can be OR'ed together to allow a client to
83 make changes to the default stream behavior:
85 - \e RTAUDIO_NONINTERLEAVED: Use non-interleaved buffers (default = interleaved).
86 - \e RTAUDIO_MINIMIZE_LATENCY: Attempt to set stream parameters for lowest possible latency.
87 - \e RTAUDIO_HOG_DEVICE: Attempt grab device for exclusive use.
88 - \e RTAUDIO_ALSA_USE_DEFAULT: Use the "default" PCM device (ALSA only).
89 - \e RTAUDIO_JACK_DONT_CONNECT: Do not automatically connect ports (JACK only).
91 By default, RtAudio streams pass and receive audio data from the
92 client in an interleaved format. By passing the
93 RTAUDIO_NONINTERLEAVED flag to the openStream() function, audio
94 data will instead be presented in non-interleaved buffers. In
95 this case, each buffer argument in the RtAudioCallback function
96 will point to a single array of data, with \c nFrames samples for
97 each channel concatenated back-to-back. For example, the first
98 sample of data for the second channel would be located at index \c
99 nFrames (assuming the \c buffer pointer was recast to the correct
100 data type for the stream).
102 Certain audio APIs offer a number of parameters that influence the
103 I/O latency of a stream. By default, RtAudio will attempt to set
104 these parameters internally for robust (glitch-free) performance
105 (though some APIs, like Windows Direct Sound, make this difficult).
106 By passing the RTAUDIO_MINIMIZE_LATENCY flag to the openStream()
107 function, internal stream settings will be influenced in an attempt
108 to minimize stream latency, though possibly at the expense of stream
111 If the RTAUDIO_HOG_DEVICE flag is set, RtAudio will attempt to
112 open the input and/or output stream device(s) for exclusive use.
113 Note that this is not possible with all supported audio APIs.
115 If the RTAUDIO_SCHEDULE_REALTIME flag is set, RtAudio will attempt
116 to select realtime scheduling (round-robin) for the callback thread.
118 If the RTAUDIO_ALSA_USE_DEFAULT flag is set, RtAudio will attempt to
119 open the "default" PCM device when using the ALSA API. Note that this
120 will override any specified input or output device id.
122 If the RTAUDIO_JACK_DONT_CONNECT flag is set, RtAudio will not attempt
123 to automatically connect the ports of the client to the audio device.
125 typedef unsigned int RtAudioStreamFlags;
126 static const RtAudioStreamFlags RTAUDIO_NONINTERLEAVED = 0x1; // Use non-interleaved buffers (default = interleaved).
127 static const RtAudioStreamFlags RTAUDIO_MINIMIZE_LATENCY = 0x2; // Attempt to set stream parameters for lowest possible latency.
128 static const RtAudioStreamFlags RTAUDIO_HOG_DEVICE = 0x4; // Attempt grab device and prevent use by others.
129 static const RtAudioStreamFlags RTAUDIO_SCHEDULE_REALTIME = 0x8; // Try to select realtime scheduling for callback thread.
130 static const RtAudioStreamFlags RTAUDIO_ALSA_USE_DEFAULT = 0x10; // Use the "default" PCM device (ALSA only).
131 static const RtAudioStreamFlags RTAUDIO_JACK_DONT_CONNECT = 0x20; // Do not automatically connect ports (JACK only).
133 /*! \typedef typedef unsigned long RtAudioStreamStatus;
134 \brief RtAudio stream status (over- or underflow) flags.
136 Notification of a stream over- or underflow is indicated by a
137 non-zero stream \c status argument in the RtAudioCallback function.
138 The stream status can be one of the following two options,
139 depending on whether the stream is open for output and/or input:
141 - \e RTAUDIO_INPUT_OVERFLOW: Input data was discarded because of an overflow condition at the driver.
142 - \e RTAUDIO_OUTPUT_UNDERFLOW: The output buffer ran low, likely producing a break in the output sound.
144 typedef unsigned int RtAudioStreamStatus;
145 static const RtAudioStreamStatus RTAUDIO_INPUT_OVERFLOW = 0x1; // Input data was discarded because of an overflow condition at the driver.
146 static const RtAudioStreamStatus RTAUDIO_OUTPUT_UNDERFLOW = 0x2; // The output buffer ran low, likely causing a gap in the output sound.
148 //! RtAudio callback function prototype.
150 All RtAudio clients must create a function of type RtAudioCallback
151 to read and/or write data from/to the audio stream. When the
152 underlying audio system is ready for new input or output data, this
153 function will be invoked.
155 \param outputBuffer For output (or duplex) streams, the client
156 should write \c nFrames of audio sample frames into this
157 buffer. This argument should be recast to the datatype
158 specified when the stream was opened. For input-only
159 streams, this argument will be NULL.
161 \param inputBuffer For input (or duplex) streams, this buffer will
162 hold \c nFrames of input audio sample frames. This
163 argument should be recast to the datatype specified when the
164 stream was opened. For output-only streams, this argument
167 \param nFrames The number of sample frames of input or output
168 data in the buffers. The actual buffer size in bytes is
169 dependent on the data type and number of channels in use.
171 \param streamTime The number of seconds that have elapsed since the
174 \param status If non-zero, this argument indicates a data overflow
175 or underflow condition for the stream. The particular
176 condition can be determined by comparison with the
177 RtAudioStreamStatus flags.
179 \param userData A pointer to optional data provided by the client
180 when opening the stream (default = NULL).
182 To continue normal stream operation, the RtAudioCallback function
183 should return a value of zero. To stop the stream and drain the
184 output buffer, the function should return a value of one. To abort
185 the stream immediately, the client should return a value of two.
187 typedef int (*RtAudioCallback)( void *outputBuffer, void *inputBuffer,
188 unsigned int nFrames,
190 RtAudioStreamStatus status,
193 /************************************************************************/
194 /*! \class RtAudioError
195 \brief Exception handling class for RtAudio.
197 The RtAudioError class is quite simple but it does allow errors to be
198 "caught" by RtAudioError::Type. See the RtAudio documentation to know
199 which methods can throw an RtAudioError.
201 /************************************************************************/
203 class RtAudioError : public std::exception
206 //! Defined RtAudioError types.
208 WARNING, /*!< A non-critical error. */
209 DEBUG_WARNING, /*!< A non-critical error which might be useful for debugging. */
210 UNSPECIFIED, /*!< The default, unspecified error type. */
211 NO_DEVICES_FOUND, /*!< No devices found on system. */
212 INVALID_DEVICE, /*!< An invalid device ID was specified. */
213 MEMORY_ERROR, /*!< An error occured during memory allocation. */
214 INVALID_PARAMETER, /*!< An invalid parameter was specified to a function. */
215 INVALID_USE, /*!< The function was called incorrectly. */
216 DRIVER_ERROR, /*!< A system driver error occured. */
217 SYSTEM_ERROR, /*!< A system error occured. */
218 THREAD_ERROR /*!< A thread error occured. */
222 RtAudioError( const std::string& message, Type type = RtAudioError::UNSPECIFIED ) throw() : message_(message), type_(type) {}
225 virtual ~RtAudioError( void ) throw() {}
227 //! Prints thrown error message to stderr.
228 virtual void printMessage( void ) const throw() { std::cerr << '\n' << message_ << "\n\n"; }
230 //! Returns the thrown error message type.
231 virtual const Type& getType(void) const throw() { return type_; }
233 //! Returns the thrown error message string.
234 virtual const std::string& getMessage(void) const throw() { return message_; }
236 //! Returns the thrown error message as a c-style string.
237 virtual const char* what( void ) const throw() { return message_.c_str(); }
240 std::string message_;
244 //! RtAudio error callback function prototype.
246 \param type Type of error.
247 \param errorText Error description.
249 typedef void (*RtAudioErrorCallback)( RtAudioError::Type type, const std::string &errorText );
251 // **************************************************************** //
253 // RtAudio class declaration.
255 // RtAudio is a "controller" used to select an available audio i/o
256 // interface. It presents a common API for the user to call but all
257 // functionality is implemented by the class RtApi and its
258 // subclasses. RtAudio creates an instance of an RtApi subclass
259 // based on the user's API choice. If no choice is made, RtAudio
260 // attempts to make a "logical" API selection.
262 // **************************************************************** //
270 //! Audio API specifier arguments.
272 UNSPECIFIED, /*!< Search for a working compiled API. */
273 LINUX_ALSA, /*!< The Advanced Linux Sound Architecture API. */
274 LINUX_PULSE, /*!< The Linux PulseAudio API. */
275 LINUX_OSS, /*!< The Linux Open Sound System API. */
276 UNIX_JACK, /*!< The Jack Low-Latency Audio Server API. */
277 MACOSX_CORE, /*!< Macintosh OS-X Core Audio API. */
278 WINDOWS_WASAPI, /*!< The Microsoft WASAPI API. */
279 WINDOWS_ASIO, /*!< The Steinberg Audio Stream I/O API. */
280 WINDOWS_DS, /*!< The Microsoft Direct Sound API. */
281 RTAUDIO_DUMMY /*!< A compilable but non-functional API. */
284 //! The public device information structure for returning queried values.
286 bool probed; /*!< true if the device capabilities were successfully probed. */
287 std::string name; /*!< Character string device identifier. */
288 unsigned int outputChannels; /*!< Maximum output channels supported by device. */
289 unsigned int inputChannels; /*!< Maximum input channels supported by device. */
290 unsigned int duplexChannels; /*!< Maximum simultaneous input/output channels supported by device. */
291 bool isDefaultOutput; /*!< true if this is the default output device. */
292 bool isDefaultInput; /*!< true if this is the default input device. */
293 std::vector<unsigned int> sampleRates; /*!< Supported sample rates (queried from list of standard rates). */
294 unsigned int preferredSampleRate; /*!< Preferred sample rate, eg. for WASAPI the system sample rate. */
295 RtAudioFormat nativeFormats; /*!< Bit mask of supported data formats. */
297 // Default constructor.
299 :probed(false), outputChannels(0), inputChannels(0), duplexChannels(0),
300 isDefaultOutput(false), isDefaultInput(false), preferredSampleRate(0), nativeFormats(0) {}
303 //! The structure for specifying input or ouput stream parameters.
304 struct StreamParameters {
305 unsigned int deviceId; /*!< Device index (0 to getDeviceCount() - 1). */
306 unsigned int nChannels; /*!< Number of channels. */
307 unsigned int firstChannel; /*!< First channel index on device (default = 0). */
309 // Default constructor.
311 : deviceId(0), nChannels(0), firstChannel(0) {}
314 //! The structure for specifying stream options.
316 The following flags can be OR'ed together to allow a client to
317 make changes to the default stream behavior:
319 - \e RTAUDIO_NONINTERLEAVED: Use non-interleaved buffers (default = interleaved).
320 - \e RTAUDIO_MINIMIZE_LATENCY: Attempt to set stream parameters for lowest possible latency.
321 - \e RTAUDIO_HOG_DEVICE: Attempt grab device for exclusive use.
322 - \e RTAUDIO_SCHEDULE_REALTIME: Attempt to select realtime scheduling for callback thread.
323 - \e RTAUDIO_ALSA_USE_DEFAULT: Use the "default" PCM device (ALSA only).
325 By default, RtAudio streams pass and receive audio data from the
326 client in an interleaved format. By passing the
327 RTAUDIO_NONINTERLEAVED flag to the openStream() function, audio
328 data will instead be presented in non-interleaved buffers. In
329 this case, each buffer argument in the RtAudioCallback function
330 will point to a single array of data, with \c nFrames samples for
331 each channel concatenated back-to-back. For example, the first
332 sample of data for the second channel would be located at index \c
333 nFrames (assuming the \c buffer pointer was recast to the correct
334 data type for the stream).
336 Certain audio APIs offer a number of parameters that influence the
337 I/O latency of a stream. By default, RtAudio will attempt to set
338 these parameters internally for robust (glitch-free) performance
339 (though some APIs, like Windows Direct Sound, make this difficult).
340 By passing the RTAUDIO_MINIMIZE_LATENCY flag to the openStream()
341 function, internal stream settings will be influenced in an attempt
342 to minimize stream latency, though possibly at the expense of stream
345 If the RTAUDIO_HOG_DEVICE flag is set, RtAudio will attempt to
346 open the input and/or output stream device(s) for exclusive use.
347 Note that this is not possible with all supported audio APIs.
349 If the RTAUDIO_SCHEDULE_REALTIME flag is set, RtAudio will attempt
350 to select realtime scheduling (round-robin) for the callback thread.
351 The \c priority parameter will only be used if the RTAUDIO_SCHEDULE_REALTIME
352 flag is set. It defines the thread's realtime priority.
354 If the RTAUDIO_ALSA_USE_DEFAULT flag is set, RtAudio will attempt to
355 open the "default" PCM device when using the ALSA API. Note that this
356 will override any specified input or output device id.
358 The \c numberOfBuffers parameter can be used to control stream
359 latency in the Windows DirectSound, Linux OSS, and Linux Alsa APIs
360 only. A value of two is usually the smallest allowed. Larger
361 numbers can potentially result in more robust stream performance,
362 though likely at the cost of stream latency. The value set by the
363 user is replaced during execution of the RtAudio::openStream()
364 function by the value actually used by the system.
366 The \c streamName parameter can be used to set the client name
367 when using the Jack API. By default, the client name is set to
368 RtApiJack. However, if you wish to create multiple instances of
369 RtAudio with Jack, each instance must have a unique client name.
371 struct StreamOptions {
372 RtAudioStreamFlags flags; /*!< A bit-mask of stream flags (RTAUDIO_NONINTERLEAVED, RTAUDIO_MINIMIZE_LATENCY, RTAUDIO_HOG_DEVICE, RTAUDIO_ALSA_USE_DEFAULT). */
373 unsigned int numberOfBuffers; /*!< Number of stream buffers. */
374 std::string streamName; /*!< A stream name (currently used only in Jack). */
375 int priority; /*!< Scheduling priority of callback thread (only used with flag RTAUDIO_SCHEDULE_REALTIME). */
377 // Default constructor.
379 : flags(0), numberOfBuffers(0), priority(0) {}
382 //! A static function to determine the current RtAudio version.
383 static std::string getVersion( void ) throw();
385 //! A static function to determine the available compiled audio APIs.
387 The values returned in the std::vector can be compared against
388 the enumerated list values. Note that there can be more than one
389 API compiled for certain operating systems.
391 static void getCompiledApi( std::vector<RtAudio::Api> &apis ) throw();
393 //! The class constructor.
395 The constructor performs minor initialization tasks. An exception
396 can be thrown if no API support is compiled.
398 If no API argument is specified and multiple API support has been
399 compiled, the default order of use is JACK, ALSA, OSS (Linux
400 systems) and ASIO, DS (Windows systems).
402 RtAudio( RtAudio::Api api=UNSPECIFIED );
406 If a stream is running or open, it will be stopped and closed
411 //! Returns the audio API specifier for the current instance of RtAudio.
412 RtAudio::Api getCurrentApi( void ) throw();
414 //! A public function that queries for the number of audio devices available.
416 This function performs a system query of available devices each time it
417 is called, thus supporting devices connected \e after instantiation. If
418 a system error occurs during processing, a warning will be issued.
420 unsigned int getDeviceCount( void ) throw();
422 //! Return an RtAudio::DeviceInfo structure for a specified device number.
425 Any device integer between 0 and getDeviceCount() - 1 is valid.
426 If an invalid argument is provided, an RtAudioError (type = INVALID_USE)
427 will be thrown. If a device is busy or otherwise unavailable, the
428 structure member "probed" will have a value of "false" and all
429 other members are undefined. If the specified device is the
430 current default input or output device, the corresponding
431 "isDefault" member will have a value of "true".
433 RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
435 //! A function that returns the index of the default output device.
437 If the underlying audio API does not provide a "default
438 device", or if no devices are available, the return value will be
439 0. Note that this is a valid device identifier and it is the
440 client's responsibility to verify that a device is available
441 before attempting to open a stream.
443 unsigned int getDefaultOutputDevice( void ) throw();
445 //! A function that returns the index of the default input device.
447 If the underlying audio API does not provide a "default
448 device", or if no devices are available, the return value will be
449 0. Note that this is a valid device identifier and it is the
450 client's responsibility to verify that a device is available
451 before attempting to open a stream.
453 unsigned int getDefaultInputDevice( void ) throw();
455 //! A public function for opening a stream with the specified parameters.
457 An RtAudioError (type = SYSTEM_ERROR) is thrown if a stream cannot be
458 opened with the specified parameters or an error occurs during
459 processing. An RtAudioError (type = INVALID_USE) is thrown if any
460 invalid device ID or channel number parameters are specified.
462 \param outputParameters Specifies output stream parameters to use
463 when opening a stream, including a device ID, number of channels,
464 and starting channel number. For input-only streams, this
465 argument should be NULL. The device ID is an index value between
466 0 and getDeviceCount() - 1.
467 \param inputParameters Specifies input stream parameters to use
468 when opening a stream, including a device ID, number of channels,
469 and starting channel number. For output-only streams, this
470 argument should be NULL. The device ID is an index value between
471 0 and getDeviceCount() - 1.
472 \param format An RtAudioFormat specifying the desired sample data format.
473 \param sampleRate The desired sample rate (sample frames per second).
474 \param *bufferFrames A pointer to a value indicating the desired
475 internal buffer size in sample frames. The actual value
476 used by the device is returned via the same pointer. A
477 value of zero can be specified, in which case the lowest
478 allowable value is determined.
479 \param callback A client-defined function that will be invoked
480 when input data is available and/or output data is needed.
481 \param userData An optional pointer to data that can be accessed
482 from within the callback function.
483 \param options An optional pointer to a structure containing various
484 global stream options, including a list of OR'ed RtAudioStreamFlags
485 and a suggested number of stream buffers that can be used to
486 control stream latency. More buffers typically result in more
487 robust performance, though at a cost of greater latency. If a
488 value of zero is specified, a system-specific median value is
489 chosen. If the RTAUDIO_MINIMIZE_LATENCY flag bit is set, the
490 lowest allowable value is used. The actual value used is
491 returned via the structure argument. The parameter is API dependent.
492 \param errorCallback A client-defined function that will be invoked
493 when an error has occured.
495 void openStream( RtAudio::StreamParameters *outputParameters,
496 RtAudio::StreamParameters *inputParameters,
497 RtAudioFormat format, unsigned int sampleRate,
498 unsigned int *bufferFrames, RtAudioCallback callback,
499 void *userData = NULL, RtAudio::StreamOptions *options = NULL, RtAudioErrorCallback errorCallback = NULL );
501 //! A function that closes a stream and frees any associated stream memory.
503 If a stream is not open, this function issues a warning and
504 returns (no exception is thrown).
506 void closeStream( void ) throw();
508 //! A function that starts a stream.
510 An RtAudioError (type = SYSTEM_ERROR) is thrown if an error occurs
511 during processing. An RtAudioError (type = INVALID_USE) is thrown if a
512 stream is not open. A warning is issued if the stream is already
515 void startStream( void );
517 //! Stop a stream, allowing any samples remaining in the output queue to be played.
519 An RtAudioError (type = SYSTEM_ERROR) is thrown if an error occurs
520 during processing. An RtAudioError (type = INVALID_USE) is thrown if a
521 stream is not open. A warning is issued if the stream is already
524 void stopStream( void );
526 //! Stop a stream, discarding any samples remaining in the input/output queue.
528 An RtAudioError (type = SYSTEM_ERROR) is thrown if an error occurs
529 during processing. An RtAudioError (type = INVALID_USE) is thrown if a
530 stream is not open. A warning is issued if the stream is already
533 void abortStream( void );
535 //! Returns true if a stream is open and false if not.
536 bool isStreamOpen( void ) const throw();
538 //! Returns true if the stream is running and false if it is stopped or not open.
539 bool isStreamRunning( void ) const throw();
541 //! Returns the number of elapsed seconds since the stream was started.
543 If a stream is not open, an RtAudioError (type = INVALID_USE) will be thrown.
545 double getStreamTime( void );
547 //! Set the stream time to a time in seconds greater than or equal to 0.0.
549 If a stream is not open, an RtAudioError (type = INVALID_USE) will be thrown.
551 void setStreamTime( double time );
553 //! Returns the internal stream latency in sample frames.
555 The stream latency refers to delay in audio input and/or output
556 caused by internal buffering by the audio system and/or hardware.
557 For duplex streams, the returned value will represent the sum of
558 the input and output latencies. If a stream is not open, an
559 RtAudioError (type = INVALID_USE) will be thrown. If the API does not
560 report latency, the return value will be zero.
562 long getStreamLatency( void );
564 //! Returns actual sample rate in use by the stream.
566 On some systems, the sample rate used may be slightly different
567 than that specified in the stream parameters. If a stream is not
568 open, an RtAudioError (type = INVALID_USE) will be thrown.
570 unsigned int getStreamSampleRate( void );
572 //! Specify whether warning messages should be printed to stderr.
573 void showWarnings( bool value = true ) throw();
577 void openRtApi( RtAudio::Api api );
581 // Operating system dependent thread functionality.
582 #if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) || defined(__WINDOWS_WASAPI__)
590 typedef uintptr_t ThreadHandle;
591 typedef CRITICAL_SECTION StreamMutex;
593 #elif defined(__LINUX_ALSA__) || defined(__LINUX_PULSE__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__)
594 // Using pthread library for various flavors of unix.
597 typedef pthread_t ThreadHandle;
598 typedef pthread_mutex_t StreamMutex;
600 #else // Setup for "dummy" behavior
602 #define __RTAUDIO_DUMMY__
603 typedef int ThreadHandle;
604 typedef int StreamMutex;
608 // This global structure type is used to pass callback information
609 // between the private RtAudio stream structure and global callback
610 // handling functions.
611 struct CallbackInfo {
612 void *object; // Used as a "this" pointer.
617 void *apiInfo; // void pointer for API specific callback information
622 // Default constructor.
624 :object(0), callback(0), userData(0), errorCallback(0), apiInfo(0), isRunning(false), doRealtime(false) {}
627 // **************************************************************** //
629 // RtApi class declaration.
631 // Subclasses of RtApi contain all API- and OS-specific code necessary
632 // to fully implement the RtAudio API.
634 // Note that RtApi is an abstract base class and cannot be
635 // explicitly instantiated. The class RtAudio will create an
636 // instance of an RtApi subclass (RtApiOss, RtApiAlsa,
637 // RtApiJack, RtApiCore, RtApiDs, or RtApiAsio).
639 // **************************************************************** //
641 #pragma pack(push, 1)
650 S24& operator = ( const int& i ) {
651 c3[0] = (i & 0x000000ff);
652 c3[1] = (i & 0x0000ff00) >> 8;
653 c3[2] = (i & 0x00ff0000) >> 16;
657 S24( const S24& v ) { *this = v; }
658 S24( const double& d ) { *this = (int) d; }
659 S24( const float& f ) { *this = (int) f; }
660 S24( const signed short& s ) { *this = (int) s; }
661 S24( const char& c ) { *this = (int) c; }
664 int i = c3[0] | (c3[1] << 8) | (c3[2] << 16);
665 if (i & 0x800000) i |= ~0xffffff;
671 #if defined( HAVE_GETTIMEOFDAY )
672 #include <sys/time.h>
683 virtual RtAudio::Api getCurrentApi( void ) = 0;
684 virtual unsigned int getDeviceCount( void ) = 0;
685 virtual RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) = 0;
686 virtual unsigned int getDefaultInputDevice( void );
687 virtual unsigned int getDefaultOutputDevice( void );
688 void openStream( RtAudio::StreamParameters *outputParameters,
689 RtAudio::StreamParameters *inputParameters,
690 RtAudioFormat format, unsigned int sampleRate,
691 unsigned int *bufferFrames, RtAudioCallback callback,
692 void *userData, RtAudio::StreamOptions *options,
693 RtAudioErrorCallback errorCallback );
694 virtual void closeStream( void );
695 virtual void startStream( void ) = 0;
696 virtual void stopStream( void ) = 0;
697 virtual void abortStream( void ) = 0;
698 long getStreamLatency( void );
699 unsigned int getStreamSampleRate( void );
700 virtual double getStreamTime( void );
701 virtual void setStreamTime( double time );
702 bool isStreamOpen( void ) const { return stream_.state != STREAM_CLOSED; }
703 bool isStreamRunning( void ) const { return stream_.state == STREAM_RUNNING; }
704 void showWarnings( bool value ) { showWarnings_ = value; }
709 static const unsigned int MAX_SAMPLE_RATES;
710 static const unsigned int SAMPLE_RATES[];
712 enum { FAILURE, SUCCESS };
728 // A protected structure used for buffer conversion.
732 RtAudioFormat inFormat, outFormat;
733 std::vector<int> inOffset;
734 std::vector<int> outOffset;
737 // A protected structure for audio streams.
739 unsigned int device[2]; // Playback and record, respectively.
740 void *apiHandle; // void pointer for API specific stream handle information
741 StreamMode mode; // OUTPUT, INPUT, or DUPLEX.
742 StreamState state; // STOPPED, RUNNING, or CLOSED
743 char *userBuffer[2]; // Playback and record, respectively.
745 bool doConvertBuffer[2]; // Playback and record, respectively.
746 bool userInterleaved;
747 bool deviceInterleaved[2]; // Playback and record, respectively.
748 bool doByteSwap[2]; // Playback and record, respectively.
749 unsigned int sampleRate;
750 unsigned int bufferSize;
751 unsigned int nBuffers;
752 unsigned int nUserChannels[2]; // Playback and record, respectively.
753 unsigned int nDeviceChannels[2]; // Playback and record channels, respectively.
754 unsigned int channelOffset[2]; // Playback and record, respectively.
755 unsigned long latency[2]; // Playback and record, respectively.
756 RtAudioFormat userFormat;
757 RtAudioFormat deviceFormat[2]; // Playback and record, respectively.
759 CallbackInfo callbackInfo;
760 ConvertInfo convertInfo[2];
761 double streamTime; // Number of elapsed seconds since the stream started.
763 #if defined(HAVE_GETTIMEOFDAY)
764 struct timeval lastTickTimestamp;
768 :apiHandle(0), deviceBuffer(0) { device[0] = 11111; device[1] = 11111; }
772 typedef signed short Int16;
773 typedef signed int Int32;
774 typedef float Float32;
775 typedef double Float64;
777 std::ostringstream errorStream_;
778 std::string errorText_;
781 bool firstErrorOccurred_;
784 Protected, api-specific method that attempts to open a device
785 with the given parameters. This function MUST be implemented by
786 all subclasses. If an error is encountered during the probe, a
787 "warning" message is reported and FAILURE is returned. A
788 successful probe is indicated by a return value of SUCCESS.
790 virtual bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
791 unsigned int firstChannel, unsigned int sampleRate,
792 RtAudioFormat format, unsigned int *bufferSize,
793 RtAudio::StreamOptions *options );
795 //! A protected function used to increment the stream time.
796 void tickStreamTime( void );
798 //! Protected common method to clear an RtApiStream structure.
799 void clearStreamInfo();
802 Protected common method that throws an RtAudioError (type =
803 INVALID_USE) if a stream is not open.
805 void verifyStream( void );
807 //! Protected common error method to allow global control over error handling.
808 void error( RtAudioError::Type type );
811 Protected method used to perform format, channel number, and/or interleaving
812 conversions between the user and device buffers.
814 void convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info );
816 //! Protected common method used to perform byte-swapping on buffers.
817 void byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format );
819 //! Protected common method that returns the number of bytes for a given format.
820 unsigned int formatBytes( RtAudioFormat format );
822 //! Protected common method that sets up the parameters for buffer conversion.
823 void setConvertInfo( StreamMode mode, unsigned int firstChannel );
826 // **************************************************************** //
828 // Inline RtAudio definitions.
830 // **************************************************************** //
832 inline RtAudio::Api RtAudio :: getCurrentApi( void ) throw() { return rtapi_->getCurrentApi(); }
833 inline unsigned int RtAudio :: getDeviceCount( void ) throw() { return rtapi_->getDeviceCount(); }
834 inline RtAudio::DeviceInfo RtAudio :: getDeviceInfo( unsigned int device ) { return rtapi_->getDeviceInfo( device ); }
835 inline unsigned int RtAudio :: getDefaultInputDevice( void ) throw() { return rtapi_->getDefaultInputDevice(); }
836 inline unsigned int RtAudio :: getDefaultOutputDevice( void ) throw() { return rtapi_->getDefaultOutputDevice(); }
837 inline void RtAudio :: closeStream( void ) throw() { return rtapi_->closeStream(); }
838 inline void RtAudio :: startStream( void ) { return rtapi_->startStream(); }
839 inline void RtAudio :: stopStream( void ) { return rtapi_->stopStream(); }
840 inline void RtAudio :: abortStream( void ) { return rtapi_->abortStream(); }
841 inline bool RtAudio :: isStreamOpen( void ) const throw() { return rtapi_->isStreamOpen(); }
842 inline bool RtAudio :: isStreamRunning( void ) const throw() { return rtapi_->isStreamRunning(); }
843 inline long RtAudio :: getStreamLatency( void ) { return rtapi_->getStreamLatency(); }
844 inline unsigned int RtAudio :: getStreamSampleRate( void ) { return rtapi_->getStreamSampleRate(); }
845 inline double RtAudio :: getStreamTime( void ) { return rtapi_->getStreamTime(); }
846 inline void RtAudio :: setStreamTime( double time ) { return rtapi_->setStreamTime( time ); }
847 inline void RtAudio :: showWarnings( bool value ) throw() { rtapi_->showWarnings( value ); }
849 // RtApi Subclass prototypes.
851 #if defined(__MACOSX_CORE__)
853 #include <CoreAudio/AudioHardware.h>
855 class RtApiCore: public RtApi
861 RtAudio::Api getCurrentApi( void ) { return RtAudio::MACOSX_CORE; }
862 unsigned int getDeviceCount( void );
863 RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
864 unsigned int getDefaultOutputDevice( void );
865 unsigned int getDefaultInputDevice( void );
866 void closeStream( void );
867 void startStream( void );
868 void stopStream( void );
869 void abortStream( void );
870 long getStreamLatency( void );
872 // This function is intended for internal use only. It must be
873 // public because it is called by the internal callback handler,
874 // which is not a member of RtAudio. External use of this function
875 // will most likely produce highly undesireable results!
876 bool callbackEvent( AudioDeviceID deviceId,
877 const AudioBufferList *inBufferList,
878 const AudioBufferList *outBufferList );
882 bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
883 unsigned int firstChannel, unsigned int sampleRate,
884 RtAudioFormat format, unsigned int *bufferSize,
885 RtAudio::StreamOptions *options );
886 static const char* getErrorCode( OSStatus code );
891 #if defined(__UNIX_JACK__)
893 class RtApiJack: public RtApi
899 RtAudio::Api getCurrentApi( void ) { return RtAudio::UNIX_JACK; }
900 unsigned int getDeviceCount( void );
901 RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
902 void closeStream( void );
903 void startStream( void );
904 void stopStream( void );
905 void abortStream( void );
906 long getStreamLatency( void );
908 // This function is intended for internal use only. It must be
909 // public because it is called by the internal callback handler,
910 // which is not a member of RtAudio. External use of this function
911 // will most likely produce highly undesireable results!
912 bool callbackEvent( unsigned long nframes );
916 bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
917 unsigned int firstChannel, unsigned int sampleRate,
918 RtAudioFormat format, unsigned int *bufferSize,
919 RtAudio::StreamOptions *options );
921 bool shouldAutoconnect_;
926 #if defined(__WINDOWS_ASIO__)
928 class RtApiAsio: public RtApi
934 RtAudio::Api getCurrentApi( void ) { return RtAudio::WINDOWS_ASIO; }
935 unsigned int getDeviceCount( void );
936 RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
937 void closeStream( void );
938 void startStream( void );
939 void stopStream( void );
940 void abortStream( void );
941 long getStreamLatency( void );
943 // This function is intended for internal use only. It must be
944 // public because it is called by the internal callback handler,
945 // which is not a member of RtAudio. External use of this function
946 // will most likely produce highly undesireable results!
947 bool callbackEvent( long bufferIndex );
951 std::vector<RtAudio::DeviceInfo> devices_;
952 void saveDeviceInfo( void );
954 bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
955 unsigned int firstChannel, unsigned int sampleRate,
956 RtAudioFormat format, unsigned int *bufferSize,
957 RtAudio::StreamOptions *options );
962 #if defined(__WINDOWS_DS__)
964 class RtApiDs: public RtApi
970 RtAudio::Api getCurrentApi( void ) { return RtAudio::WINDOWS_DS; }
971 unsigned int getDeviceCount( void );
972 unsigned int getDefaultOutputDevice( void );
973 unsigned int getDefaultInputDevice( void );
974 RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
975 void closeStream( void );
976 void startStream( void );
977 void stopStream( void );
978 void abortStream( void );
979 long getStreamLatency( void );
981 // This function is intended for internal use only. It must be
982 // public because it is called by the internal callback handler,
983 // which is not a member of RtAudio. External use of this function
984 // will most likely produce highly undesireable results!
985 void callbackEvent( void );
991 long duplexPrerollBytes;
992 std::vector<struct DsDevice> dsDevices;
993 bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
994 unsigned int firstChannel, unsigned int sampleRate,
995 RtAudioFormat format, unsigned int *bufferSize,
996 RtAudio::StreamOptions *options );
1001 #if defined(__WINDOWS_WASAPI__)
1003 struct IMMDeviceEnumerator;
1005 class RtApiWasapi : public RtApi
1011 RtAudio::Api getCurrentApi( void ) { return RtAudio::WINDOWS_WASAPI; }
1012 unsigned int getDeviceCount( void );
1013 RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
1014 unsigned int getDefaultOutputDevice( void );
1015 unsigned int getDefaultInputDevice( void );
1016 void closeStream( void );
1017 void startStream( void );
1018 void stopStream( void );
1019 void abortStream( void );
1022 bool coInitialized_;
1023 IMMDeviceEnumerator* deviceEnumerator_;
1025 bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
1026 unsigned int firstChannel, unsigned int sampleRate,
1027 RtAudioFormat format, unsigned int* bufferSize,
1028 RtAudio::StreamOptions* options );
1030 static DWORD WINAPI runWasapiThread( void* wasapiPtr );
1031 static DWORD WINAPI stopWasapiThread( void* wasapiPtr );
1032 static DWORD WINAPI abortWasapiThread( void* wasapiPtr );
1033 void wasapiThread();
1038 #if defined(__LINUX_ALSA__)
1040 class RtApiAlsa: public RtApi
1046 RtAudio::Api getCurrentApi() { return RtAudio::LINUX_ALSA; }
1047 unsigned int getDeviceCount( void );
1048 RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
1049 void closeStream( void );
1050 void startStream( void );
1051 void stopStream( void );
1052 void abortStream( void );
1054 // This function is intended for internal use only. It must be
1055 // public because it is called by the internal callback handler,
1056 // which is not a member of RtAudio. External use of this function
1057 // will most likely produce highly undesireable results!
1058 void callbackEvent( void );
1062 std::vector<RtAudio::DeviceInfo> devices_;
1063 void saveDeviceInfo( void );
1064 bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
1065 unsigned int firstChannel, unsigned int sampleRate,
1066 RtAudioFormat format, unsigned int *bufferSize,
1067 RtAudio::StreamOptions *options );
1072 #if defined(__LINUX_PULSE__)
1074 class RtApiPulse: public RtApi
1078 RtAudio::Api getCurrentApi() { return RtAudio::LINUX_PULSE; }
1079 unsigned int getDeviceCount( void );
1080 RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
1081 void closeStream( void );
1082 void startStream( void );
1083 void stopStream( void );
1084 void abortStream( void );
1086 // This function is intended for internal use only. It must be
1087 // public because it is called by the internal callback handler,
1088 // which is not a member of RtAudio. External use of this function
1089 // will most likely produce highly undesireable results!
1090 void callbackEvent( void );
1094 std::vector<RtAudio::DeviceInfo> devices_;
1095 void saveDeviceInfo( void );
1096 bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
1097 unsigned int firstChannel, unsigned int sampleRate,
1098 RtAudioFormat format, unsigned int *bufferSize,
1099 RtAudio::StreamOptions *options );
1104 #if defined(__LINUX_OSS__)
1106 class RtApiOss: public RtApi
1112 RtAudio::Api getCurrentApi() { return RtAudio::LINUX_OSS; }
1113 unsigned int getDeviceCount( void );
1114 RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
1115 void closeStream( void );
1116 void startStream( void );
1117 void stopStream( void );
1118 void abortStream( void );
1120 // This function is intended for internal use only. It must be
1121 // public because it is called by the internal callback handler,
1122 // which is not a member of RtAudio. External use of this function
1123 // will most likely produce highly undesireable results!
1124 void callbackEvent( void );
1128 bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
1129 unsigned int firstChannel, unsigned int sampleRate,
1130 RtAudioFormat format, unsigned int *bufferSize,
1131 RtAudio::StreamOptions *options );
1136 #if defined(__RTAUDIO_DUMMY__)
1138 class RtApiDummy: public RtApi
1142 RtApiDummy() { errorText_ = "RtApiDummy: This class provides no functionality."; error( RtAudioError::WARNING ); }
1143 RtAudio::Api getCurrentApi( void ) { return RtAudio::RTAUDIO_DUMMY; }
1144 unsigned int getDeviceCount( void ) { return 0; }
1145 RtAudio::DeviceInfo getDeviceInfo( unsigned int /*device*/ ) { RtAudio::DeviceInfo info; return info; }
1146 void closeStream( void ) {}
1147 void startStream( void ) {}
1148 void stopStream( void ) {}
1149 void abortStream( void ) {}
1153 bool probeDeviceOpen( unsigned int /*device*/, StreamMode /*mode*/, unsigned int /*channels*/,
1154 unsigned int /*firstChannel*/, unsigned int /*sampleRate*/,
1155 RtAudioFormat /*format*/, unsigned int * /*bufferSize*/,
1156 RtAudio::StreamOptions * /*options*/ ) { return false; }
1163 // Indentation settings for Vim and Emacs
1166 // c-basic-offset: 2
1167 // indent-tabs-mode: nil
1170 // vim: et sts=2 sw=2