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 GitHub site: https://github.com/thestk/rtaudio
11 RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/
13 RtAudio: realtime audio i/o C++ classes
14 Copyright (c) 2001-2019 Gary P. Scavone
16 Permission is hereby granted, free of charge, to any person
17 obtaining a copy of this software and associated documentation files
18 (the "Software"), to deal in the Software without restriction,
19 including without limitation the rights to use, copy, modify, merge,
20 publish, distribute, sublicense, and/or sell copies of the Software,
21 and to permit persons to whom the Software is furnished to do so,
22 subject to the following conditions:
24 The above copyright notice and this permission notice shall be
25 included in all copies or substantial portions of the Software.
27 Any person wishing to distribute modifications to the Software is
28 asked to send the modifications to the original developer so that
29 they can be incorporated into the canonical version. This is,
30 however, not a binding provision of this license.
32 THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
33 EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
34 MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
35 IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
36 ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
37 CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
38 WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
40 /************************************************************************/
49 #define RTAUDIO_VERSION "6.0.0beta1"
51 #if defined _WIN32 || defined __CYGWIN__
52 #if defined(RTAUDIO_EXPORT)
53 #define RTAUDIO_DLL_PUBLIC __declspec(dllexport)
55 #define RTAUDIO_DLL_PUBLIC
59 #define RTAUDIO_DLL_PUBLIC __attribute__( (visibility( "default" )) )
61 #define RTAUDIO_DLL_PUBLIC
69 /*! \typedef typedef unsigned long RtAudioFormat;
70 \brief RtAudio data format type.
72 Support for signed integers and floats. Audio data fed to/from an
73 RtAudio stream is assumed to ALWAYS be in host byte order. The
74 internal routines will automatically take care of any necessary
75 byte-swapping between the host format and the soundcard. Thus,
76 endian-ness is not a concern in the following format definitions.
78 - \e RTAUDIO_SINT8: 8-bit signed integer.
79 - \e RTAUDIO_SINT16: 16-bit signed integer.
80 - \e RTAUDIO_SINT24: 24-bit signed integer.
81 - \e RTAUDIO_SINT32: 32-bit signed integer.
82 - \e RTAUDIO_FLOAT32: Normalized between plus/minus 1.0.
83 - \e RTAUDIO_FLOAT64: Normalized between plus/minus 1.0.
85 typedef unsigned long RtAudioFormat;
86 static const RtAudioFormat RTAUDIO_SINT8 = 0x1; // 8-bit signed integer.
87 static const RtAudioFormat RTAUDIO_SINT16 = 0x2; // 16-bit signed integer.
88 static const RtAudioFormat RTAUDIO_SINT24 = 0x4; // 24-bit signed integer.
89 static const RtAudioFormat RTAUDIO_SINT32 = 0x8; // 32-bit signed integer.
90 static const RtAudioFormat RTAUDIO_FLOAT32 = 0x10; // Normalized between plus/minus 1.0.
91 static const RtAudioFormat RTAUDIO_FLOAT64 = 0x20; // Normalized between plus/minus 1.0.
93 /*! \typedef typedef unsigned long RtAudioStreamFlags;
94 \brief RtAudio stream option flags.
96 The following flags can be OR'ed together to allow a client to
97 make changes to the default stream behavior:
99 - \e RTAUDIO_NONINTERLEAVED: Use non-interleaved buffers (default = interleaved).
100 - \e RTAUDIO_MINIMIZE_LATENCY: Attempt to set stream parameters for lowest possible latency.
101 - \e RTAUDIO_HOG_DEVICE: Attempt grab device for exclusive use.
102 - \e RTAUDIO_ALSA_USE_DEFAULT: Use the "default" PCM device (ALSA only).
103 - \e RTAUDIO_JACK_DONT_CONNECT: Do not automatically connect ports (JACK only).
105 By default, RtAudio streams pass and receive audio data from the
106 client in an interleaved format. By passing the
107 RTAUDIO_NONINTERLEAVED flag to the openStream() function, audio
108 data will instead be presented in non-interleaved buffers. In
109 this case, each buffer argument in the RtAudioCallback function
110 will point to a single array of data, with \c nFrames samples for
111 each channel concatenated back-to-back. For example, the first
112 sample of data for the second channel would be located at index \c
113 nFrames (assuming the \c buffer pointer was recast to the correct
114 data type for the stream).
116 Certain audio APIs offer a number of parameters that influence the
117 I/O latency of a stream. By default, RtAudio will attempt to set
118 these parameters internally for robust (glitch-free) performance
119 (though some APIs, like Windows DirectSound, make this difficult).
120 By passing the RTAUDIO_MINIMIZE_LATENCY flag to the openStream()
121 function, internal stream settings will be influenced in an attempt
122 to minimize stream latency, though possibly at the expense of stream
125 If the RTAUDIO_HOG_DEVICE flag is set, RtAudio will attempt to
126 open the input and/or output stream device(s) for exclusive use.
127 Note that this is not possible with all supported audio APIs.
129 If the RTAUDIO_SCHEDULE_REALTIME flag is set, RtAudio will attempt
130 to select realtime scheduling (round-robin) for the callback thread.
132 If the RTAUDIO_ALSA_USE_DEFAULT flag is set, RtAudio will attempt to
133 open the "default" PCM device when using the ALSA API. Note that this
134 will override any specified input or output device id.
136 If the RTAUDIO_JACK_DONT_CONNECT flag is set, RtAudio will not attempt
137 to automatically connect the ports of the client to the audio device.
139 typedef unsigned int RtAudioStreamFlags;
140 static const RtAudioStreamFlags RTAUDIO_NONINTERLEAVED = 0x1; // Use non-interleaved buffers (default = interleaved).
141 static const RtAudioStreamFlags RTAUDIO_MINIMIZE_LATENCY = 0x2; // Attempt to set stream parameters for lowest possible latency.
142 static const RtAudioStreamFlags RTAUDIO_HOG_DEVICE = 0x4; // Attempt grab device and prevent use by others.
143 static const RtAudioStreamFlags RTAUDIO_SCHEDULE_REALTIME = 0x8; // Try to select realtime scheduling for callback thread.
144 static const RtAudioStreamFlags RTAUDIO_ALSA_USE_DEFAULT = 0x10; // Use the "default" PCM device (ALSA only).
145 static const RtAudioStreamFlags RTAUDIO_JACK_DONT_CONNECT = 0x20; // Do not automatically connect ports (JACK only).
147 /*! \typedef typedef unsigned long RtAudioStreamStatus;
148 \brief RtAudio stream status (over- or underflow) flags.
150 Notification of a stream over- or underflow is indicated by a
151 non-zero stream \c status argument in the RtAudioCallback function.
152 The stream status can be one of the following two options,
153 depending on whether the stream is open for output and/or input:
155 - \e RTAUDIO_INPUT_OVERFLOW: Input data was discarded because of an overflow condition at the driver.
156 - \e RTAUDIO_OUTPUT_UNDERFLOW: The output buffer ran low, likely producing a break in the output sound.
158 typedef unsigned int RtAudioStreamStatus;
159 static const RtAudioStreamStatus RTAUDIO_INPUT_OVERFLOW = 0x1; // Input data was discarded because of an overflow condition at the driver.
160 static const RtAudioStreamStatus RTAUDIO_OUTPUT_UNDERFLOW = 0x2; // The output buffer ran low, likely causing a gap in the output sound.
162 //! RtAudio callback function prototype.
164 All RtAudio clients must create a function of type RtAudioCallback
165 to read and/or write data from/to the audio stream. When the
166 underlying audio system is ready for new input or output data, this
167 function will be invoked.
169 \param outputBuffer For output (or duplex) streams, the client
170 should write \c nFrames of audio sample frames into this
171 buffer. This argument should be recast to the datatype
172 specified when the stream was opened. For input-only
173 streams, this argument will be NULL.
175 \param inputBuffer For input (or duplex) streams, this buffer will
176 hold \c nFrames of input audio sample frames. This
177 argument should be recast to the datatype specified when the
178 stream was opened. For output-only streams, this argument
181 \param nFrames The number of sample frames of input or output
182 data in the buffers. The actual buffer size in bytes is
183 dependent on the data type and number of channels in use.
185 \param streamTime The number of seconds that have elapsed since the
188 \param status If non-zero, this argument indicates a data overflow
189 or underflow condition for the stream. The particular
190 condition can be determined by comparison with the
191 RtAudioStreamStatus flags.
193 \param userData A pointer to optional data provided by the client
194 when opening the stream (default = NULL).
197 To continue normal stream operation, the RtAudioCallback function
198 should return a value of zero. To stop the stream and drain the
199 output buffer, the function should return a value of one. To abort
200 the stream immediately, the client should return a value of two.
202 typedef int (*RtAudioCallback)( void *outputBuffer, void *inputBuffer,
203 unsigned int nFrames,
205 RtAudioStreamStatus status,
208 enum RtAudioErrorType {
209 RTAUDIO_NO_ERROR, /*!< No error. */
210 RTAUDIO_WARNING, /*!< A non-critical error. */
211 RTAUDIO_UNKNOWN_ERROR, /*!< An unspecified error type. */
212 RTAUDIO_NO_DEVICES_FOUND, /*!< No devices found on system. */
213 RTAUDIO_INVALID_DEVICE, /*!< An invalid device ID was specified. */
214 RTAUDIO_DEVICE_DISCONNECT, /*!< A device in use was disconnected. */
215 RTAUDIO_MEMORY_ERROR, /*!< An error occured during memory allocation. */
216 RTAUDIO_INVALID_PARAMETER, /*!< An invalid parameter was specified to a function. */
217 RTAUDIO_INVALID_USE, /*!< The function was called incorrectly. */
218 RTAUDIO_DRIVER_ERROR, /*!< A system driver error occurred. */
219 RTAUDIO_SYSTEM_ERROR, /*!< A system error occurred. */
220 RTAUDIO_THREAD_ERROR /*!< A thread error occurred. */
223 //! RtAudio error callback function prototype.
225 \param type Type of error.
226 \param errorText Error description.
228 typedef void (*RtAudioErrorCallback)( RtAudioErrorType type, const std::string &errorText );
230 // **************************************************************** //
232 // RtAudio class declaration.
234 // RtAudio is a "controller" used to select an available audio i/o
235 // interface. It presents a common API for the user to call but all
236 // functionality is implemented by the class RtApi and its
237 // subclasses. RtAudio creates an instance of an RtApi subclass
238 // based on the user's API choice. If no choice is made, RtAudio
239 // attempts to make a "logical" API selection.
241 // **************************************************************** //
245 class RTAUDIO_DLL_PUBLIC RtAudio
249 //! Audio API specifier arguments.
251 UNSPECIFIED, /*!< Search for a working compiled API. */
252 LINUX_ALSA, /*!< The Advanced Linux Sound Architecture API. */
253 LINUX_PULSE, /*!< The Linux PulseAudio API. */
254 LINUX_OSS, /*!< The Linux Open Sound System API. */
255 UNIX_JACK, /*!< The Jack Low-Latency Audio Server API. */
256 MACOSX_CORE, /*!< Macintosh OS-X Core Audio API. */
257 WINDOWS_WASAPI, /*!< The Microsoft WASAPI API. */
258 WINDOWS_ASIO, /*!< The Steinberg Audio Stream I/O API. */
259 WINDOWS_DS, /*!< The Microsoft DirectSound API. */
260 RTAUDIO_DUMMY, /*!< A compilable but non-functional API. */
261 NUM_APIS /*!< Number of values in this enum. */
264 //! The public device information structure for returning queried values.
266 bool probed; /*!< true if the device capabilities were successfully probed. */
267 std::string name; /*!< Character string device identifier. */
268 unsigned int outputChannels; /*!< Maximum output channels supported by device. */
269 unsigned int inputChannels; /*!< Maximum input channels supported by device. */
270 unsigned int duplexChannels; /*!< Maximum simultaneous input/output channels supported by device. */
271 bool isDefaultOutput; /*!< true if this is the default output device. */
272 bool isDefaultInput; /*!< true if this is the default input device. */
273 std::vector<unsigned int> sampleRates; /*!< Supported sample rates (queried from list of standard rates). */
274 unsigned int currentSampleRate; /*!< Current sample rate, system sample rate as currently configured. */
275 unsigned int preferredSampleRate; /*!< Preferred sample rate, e.g. for WASAPI the system sample rate. */
276 RtAudioFormat nativeFormats; /*!< Bit mask of supported data formats. */
278 // Default constructor.
280 :probed(false), outputChannels(0), inputChannels(0), duplexChannels(0),
281 isDefaultOutput(false), isDefaultInput(false), currentSampleRate(0), preferredSampleRate(0), nativeFormats(0) {}
284 //! The structure for specifying input or ouput stream parameters.
285 struct StreamParameters {
286 unsigned int deviceId; /*!< Device index (0 to getDeviceCount() - 1). */
287 unsigned int nChannels; /*!< Number of channels. */
288 unsigned int firstChannel; /*!< First channel index on device (default = 0). */
290 // Default constructor.
292 : deviceId(0), nChannels(0), firstChannel(0) {}
295 //! The structure for specifying stream options.
297 The following flags can be OR'ed together to allow a client to
298 make changes to the default stream behavior:
300 - \e RTAUDIO_NONINTERLEAVED: Use non-interleaved buffers (default = interleaved).
301 - \e RTAUDIO_MINIMIZE_LATENCY: Attempt to set stream parameters for lowest possible latency.
302 - \e RTAUDIO_HOG_DEVICE: Attempt grab device for exclusive use.
303 - \e RTAUDIO_SCHEDULE_REALTIME: Attempt to select realtime scheduling for callback thread.
304 - \e RTAUDIO_ALSA_USE_DEFAULT: Use the "default" PCM device (ALSA only).
306 By default, RtAudio streams pass and receive audio data from the
307 client in an interleaved format. By passing the
308 RTAUDIO_NONINTERLEAVED flag to the openStream() function, audio
309 data will instead be presented in non-interleaved buffers. In
310 this case, each buffer argument in the RtAudioCallback function
311 will point to a single array of data, with \c nFrames samples for
312 each channel concatenated back-to-back. For example, the first
313 sample of data for the second channel would be located at index \c
314 nFrames (assuming the \c buffer pointer was recast to the correct
315 data type for the stream).
317 Certain audio APIs offer a number of parameters that influence the
318 I/O latency of a stream. By default, RtAudio will attempt to set
319 these parameters internally for robust (glitch-free) performance
320 (though some APIs, like Windows DirectSound, make this difficult).
321 By passing the RTAUDIO_MINIMIZE_LATENCY flag to the openStream()
322 function, internal stream settings will be influenced in an attempt
323 to minimize stream latency, though possibly at the expense of stream
326 If the RTAUDIO_HOG_DEVICE flag is set, RtAudio will attempt to
327 open the input and/or output stream device(s) for exclusive use.
328 Note that this is not possible with all supported audio APIs.
330 If the RTAUDIO_SCHEDULE_REALTIME flag is set, RtAudio will attempt
331 to select realtime scheduling (round-robin) for the callback thread.
332 The \c priority parameter will only be used if the RTAUDIO_SCHEDULE_REALTIME
333 flag is set. It defines the thread's realtime priority.
335 If the RTAUDIO_ALSA_USE_DEFAULT flag is set, RtAudio will attempt to
336 open the "default" PCM device when using the ALSA API. Note that this
337 will override any specified input or output device id.
339 The \c numberOfBuffers parameter can be used to control stream
340 latency in the Windows DirectSound, Linux OSS, and Linux Alsa APIs
341 only. A value of two is usually the smallest allowed. Larger
342 numbers can potentially result in more robust stream performance,
343 though likely at the cost of stream latency. The value set by the
344 user is replaced during execution of the RtAudio::openStream()
345 function by the value actually used by the system.
347 The \c streamName parameter can be used to set the client name
348 when using the Jack API. By default, the client name is set to
349 RtApiJack. However, if you wish to create multiple instances of
350 RtAudio with Jack, each instance must have a unique client name.
352 struct StreamOptions {
353 RtAudioStreamFlags flags; /*!< A bit-mask of stream flags (RTAUDIO_NONINTERLEAVED, RTAUDIO_MINIMIZE_LATENCY, RTAUDIO_HOG_DEVICE, RTAUDIO_ALSA_USE_DEFAULT). */
354 unsigned int numberOfBuffers; /*!< Number of stream buffers. */
355 std::string streamName; /*!< A stream name (currently used only in Jack). */
356 int priority; /*!< Scheduling priority of callback thread (only used with flag RTAUDIO_SCHEDULE_REALTIME). */
358 // Default constructor.
360 : flags(0), numberOfBuffers(0), priority(0) {}
363 //! A static function to determine the current RtAudio version.
364 static std::string getVersion( void );
366 //! A static function to determine the available compiled audio APIs.
368 The values returned in the std::vector can be compared against
369 the enumerated list values. Note that there can be more than one
370 API compiled for certain operating systems.
372 static void getCompiledApi( std::vector<RtAudio::Api> &apis );
374 //! Return the name of a specified compiled audio API.
376 This obtains a short lower-case name used for identification purposes.
377 This value is guaranteed to remain identical across library versions.
378 If the API is unknown, this function will return the empty string.
380 static std::string getApiName( RtAudio::Api api );
382 //! Return the display name of a specified compiled audio API.
384 This obtains a long name used for display purposes.
385 If the API is unknown, this function will return the empty string.
387 static std::string getApiDisplayName( RtAudio::Api api );
389 //! Return the compiled audio API having the given name.
391 A case insensitive comparison will check the specified name
392 against the list of compiled APIs, and return the one which
393 matches. On failure, the function returns UNSPECIFIED.
395 static RtAudio::Api getCompiledApiByName( const std::string &name );
397 //! The class constructor.
399 The constructor attempts to create an RtApi instance.
401 If an API argument is specified but that API has not been
402 compiled, a warning is issued and an instance of an available API
403 is created. If no compiled API is found, the routine will abort
404 (though this should be impossible because RtDummy is the default
405 if no API-specific preprocessor definition is provided to the
406 compiler). If no API argument is specified and multiple API
407 support has been compiled, the default order of use is JACK, ALSA,
408 OSS (Linux systems) and ASIO, DS (Windows systems).
410 An optional errorCallback function can be specified to
411 subsequently receive warning and error messages.
413 RtAudio( RtAudio::Api api=UNSPECIFIED, RtAudioErrorCallback errorCallback=0 );
417 If a stream is running or open, it will be stopped and closed
422 //! Returns the audio API specifier for the current instance of RtAudio.
423 RtAudio::Api getCurrentApi( void );
425 //! A public function that queries for the number of audio devices available.
427 This function performs a system query of available devices each time it
428 is called, thus supporting devices connected \e after instantiation. If
429 a system error occurs during processing, a warning will be issued.
431 unsigned int getDeviceCount( void );
433 //! Return an RtAudio::DeviceInfo structure for a specified device number.
435 Any device integer between 0 and getDeviceCount() - 1 is valid.
436 If an invalid argument is provided, an RTAUDIO_INVALID_USE
437 will be passed to the user-provided errorCallback function (or
438 otherwise printed to stderr), the structure member "probed" will
439 have a value of "false" and all other members will be undefined.
440 If a device is busy or otherwise unavailable, the structure member
441 "probed" will have a value of "false" and all other members will
442 be undefined. If the specified device is the current default
443 input or output device, the corresponding "isDefault" member will
444 have a value of "true".
446 RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
448 //! A function that returns the index of the default output device.
450 If the underlying audio API does not provide a "default
451 device", or if no devices are available, the return value will be
452 0. Note that this is a valid device identifier and it is the
453 client's responsibility to verify that a device is available
454 before attempting to open a stream.
456 unsigned int getDefaultOutputDevice( void );
458 //! A function that returns the index of the default input device.
460 If the underlying audio API does not provide a "default
461 device", or if no devices are available, the return value will be
462 0. Note that this is a valid device identifier and it is the
463 client's responsibility to verify that a device is available
464 before attempting to open a stream.
466 unsigned int getDefaultInputDevice( void );
468 //! A public function for opening a stream with the specified parameters.
470 An RTAUDIO_SYSTEM_ERROR is returned if a stream cannot be
471 opened with the specified parameters or an error occurs during
472 processing. An RTAUDIO_INVALID_USE is returned if a stream
473 is already open or any invalid stream parameters are specified.
475 \param outputParameters Specifies output stream parameters to use
476 when opening a stream, including a device ID, number of channels,
477 and starting channel number. For input-only streams, this
478 argument should be NULL. The device ID is an index value between
479 0 and getDeviceCount() - 1.
480 \param inputParameters Specifies input stream parameters to use
481 when opening a stream, including a device ID, number of channels,
482 and starting channel number. For output-only streams, this
483 argument should be NULL. The device ID is an index value between
484 0 and getDeviceCount() - 1.
485 \param format An RtAudioFormat specifying the desired sample data format.
486 \param sampleRate The desired sample rate (sample frames per second).
487 \param bufferFrames A pointer to a value indicating the desired
488 internal buffer size in sample frames. The actual value
489 used by the device is returned via the same pointer. A
490 value of zero can be specified, in which case the lowest
491 allowable value is determined.
492 \param callback A client-defined function that will be invoked
493 when input data is available and/or output data is needed.
494 \param userData An optional pointer to data that can be accessed
495 from within the callback function.
496 \param options An optional pointer to a structure containing various
497 global stream options, including a list of OR'ed RtAudioStreamFlags
498 and a suggested number of stream buffers that can be used to
499 control stream latency. More buffers typically result in more
500 robust performance, though at a cost of greater latency. If a
501 value of zero is specified, a system-specific median value is
502 chosen. If the RTAUDIO_MINIMIZE_LATENCY flag bit is set, the
503 lowest allowable value is used. The actual value used is
504 returned via the structure argument. The parameter is API dependent.
506 RtAudioErrorType openStream( RtAudio::StreamParameters *outputParameters,
507 RtAudio::StreamParameters *inputParameters,
508 RtAudioFormat format, unsigned int sampleRate,
509 unsigned int *bufferFrames, RtAudioCallback callback,
510 void *userData = NULL, RtAudio::StreamOptions *options = NULL );
512 //! A function that closes a stream and frees any associated stream memory.
514 If a stream is not open, an RTAUDIO_WARNING will be passed to the
515 user-provided errorCallback function (or otherwise printed to
518 void closeStream( void );
520 //! A function that starts a stream.
522 An RTAUDIO_SYSTEM_ERROR is returned if an error occurs during
523 processing. An RTAUDIO_WARNING is returned if a stream is not open
524 or is already running.
526 RtAudioErrorType startStream( void );
528 //! Stop a stream, allowing any samples remaining in the output queue to be played.
530 An RTAUDIO_SYSTEM_ERROR is returned if an error occurs during
531 processing. An RTAUDIO_WARNING is returned if a stream is not
532 open or is already stopped.
534 RtAudioErrorType stopStream( void );
536 //! Stop a stream, discarding any samples remaining in the input/output queue.
538 An RTAUDIO_SYSTEM_ERROR is returned if an error occurs during
539 processing. An RTAUDIO_WARNING is returned if a stream is not
540 open or is already stopped.
542 RtAudioErrorType abortStream( void );
544 //! Returns true if a stream is open and false if not.
545 bool isStreamOpen( void ) const;
547 //! Returns true if the stream is running and false if it is stopped or not open.
548 bool isStreamRunning( void ) const;
550 //! Returns the number of seconds of processed data since the stream was started.
552 The stream time is calculated from the number of sample frames
553 processed by the underlying audio system, which will increment by
554 units of the audio buffer size. It is not an absolute running
555 time. If a stream is not open, the returned value may not be
558 double getStreamTime( void );
560 //! Set the stream time to a time in seconds greater than or equal to 0.0.
561 void setStreamTime( double time );
563 //! Returns the internal stream latency in sample frames.
565 The stream latency refers to delay in audio input and/or output
566 caused by internal buffering by the audio system and/or hardware.
567 For duplex streams, the returned value will represent the sum of
568 the input and output latencies. If a stream is not open, the
569 returned value will be invalid. If the API does not report
570 latency, the return value will be zero.
572 long getStreamLatency( void );
574 //! Returns actual sample rate in use by the (open) stream.
576 On some systems, the sample rate used may be slightly different
577 than that specified in the stream parameters. If a stream is not
578 open, a value of zero is returned.
580 unsigned int getStreamSampleRate( void );
582 //! Set a client-defined function that will be invoked when an error or warning occurs.
583 void setErrorCallback( RtAudioErrorCallback errorCallback );
585 //! Specify whether warning messages should be output or not.
587 The default behaviour is for warning messages to be output,
588 either to a client-defined error callback function (if specified)
591 void showWarnings( bool value = true );
595 void openRtApi( RtAudio::Api api );
599 // Operating system dependent thread functionality.
600 #if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) || defined(__WINDOWS_WASAPI__)
609 typedef uintptr_t ThreadHandle;
610 typedef CRITICAL_SECTION StreamMutex;
612 #elif defined(__LINUX_ALSA__) || defined(__LINUX_PULSE__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__)
613 // Using pthread library for various flavors of unix.
616 typedef pthread_t ThreadHandle;
617 typedef pthread_mutex_t StreamMutex;
619 #else // Setup for "dummy" behavior
621 #define __RTAUDIO_DUMMY__
622 typedef int ThreadHandle;
623 typedef int StreamMutex;
627 // This global structure type is used to pass callback information
628 // between the private RtAudio stream structure and global callback
629 // handling functions.
630 struct CallbackInfo {
631 void *object; // Used as a "this" pointer.
635 void *apiInfo; // void pointer for API specific callback information
639 bool deviceDisconnected;
641 // Default constructor.
643 :object(0), callback(0), userData(0), apiInfo(0), isRunning(false), doRealtime(false), priority(0), deviceDisconnected(false) {}
646 // **************************************************************** //
648 // RtApi class declaration.
650 // Subclasses of RtApi contain all API- and OS-specific code necessary
651 // to fully implement the RtAudio API.
653 // Note that RtApi is an abstract base class and cannot be
654 // explicitly instantiated. The class RtAudio will create an
655 // instance of an RtApi subclass (RtApiOss, RtApiAlsa,
656 // RtApiJack, RtApiCore, RtApiDs, or RtApiAsio).
658 // **************************************************************** //
660 #pragma pack(push, 1)
669 S24& operator = ( const int& i ) {
670 c3[0] = (i & 0x000000ff);
671 c3[1] = (i & 0x0000ff00) >> 8;
672 c3[2] = (i & 0x00ff0000) >> 16;
676 S24( const double& d ) { *this = (int) d; }
677 S24( const float& f ) { *this = (int) f; }
678 S24( const signed short& s ) { *this = (int) s; }
679 S24( const char& c ) { *this = (int) c; }
682 int i = c3[0] | (c3[1] << 8) | (c3[2] << 16);
683 if (i & 0x800000) i |= ~0xffffff;
689 #if defined( HAVE_GETTIMEOFDAY )
690 #include <sys/time.h>
695 class RTAUDIO_DLL_PUBLIC RtApi
701 virtual RtAudio::Api getCurrentApi( void ) = 0;
702 virtual unsigned int getDeviceCount( void ) = 0;
703 virtual RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) = 0;
704 virtual unsigned int getDefaultInputDevice( void );
705 virtual unsigned int getDefaultOutputDevice( void );
706 RtAudioErrorType openStream( RtAudio::StreamParameters *outputParameters,
707 RtAudio::StreamParameters *inputParameters,
708 RtAudioFormat format, unsigned int sampleRate,
709 unsigned int *bufferFrames, RtAudioCallback callback,
710 void *userData, RtAudio::StreamOptions *options );
711 virtual void closeStream( void );
712 virtual RtAudioErrorType startStream( void ) = 0;
713 virtual RtAudioErrorType stopStream( void ) = 0;
714 virtual RtAudioErrorType abortStream( void ) = 0;
715 long getStreamLatency( void );
716 unsigned int getStreamSampleRate( void );
717 virtual double getStreamTime( void ) const { return stream_.streamTime; }
718 virtual void setStreamTime( double time );
719 bool isStreamOpen( void ) const { return stream_.state != STREAM_CLOSED; }
720 bool isStreamRunning( void ) const { return stream_.state == STREAM_RUNNING; }
721 void setErrorCallback( RtAudioErrorCallback errorCallback ) { errorCallback_ = errorCallback; }
722 void showWarnings( bool value ) { showWarnings_ = value; }
727 static const unsigned int MAX_SAMPLE_RATES;
728 static const unsigned int SAMPLE_RATES[];
730 enum { FAILURE, SUCCESS };
746 // A protected structure used for buffer conversion.
750 RtAudioFormat inFormat, outFormat;
751 std::vector<int> inOffset;
752 std::vector<int> outOffset;
755 // A protected structure for audio streams.
757 unsigned int device[2]; // Playback and record, respectively.
758 void *apiHandle; // void pointer for API specific stream handle information
759 StreamMode mode; // OUTPUT, INPUT, or DUPLEX.
760 StreamState state; // STOPPED, RUNNING, or CLOSED
761 char *userBuffer[2]; // Playback and record, respectively.
763 bool doConvertBuffer[2]; // Playback and record, respectively.
764 bool userInterleaved;
765 bool deviceInterleaved[2]; // Playback and record, respectively.
766 bool doByteSwap[2]; // Playback and record, respectively.
767 unsigned int sampleRate;
768 unsigned int bufferSize;
769 unsigned int nBuffers;
770 unsigned int nUserChannels[2]; // Playback and record, respectively.
771 unsigned int nDeviceChannels[2]; // Playback and record channels, respectively.
772 unsigned int channelOffset[2]; // Playback and record, respectively.
773 unsigned long latency[2]; // Playback and record, respectively.
774 RtAudioFormat userFormat;
775 RtAudioFormat deviceFormat[2]; // Playback and record, respectively.
777 CallbackInfo callbackInfo;
778 ConvertInfo convertInfo[2];
779 double streamTime; // Number of elapsed seconds since the stream started.
781 #if defined(HAVE_GETTIMEOFDAY)
782 struct timeval lastTickTimestamp;
786 :apiHandle(0), deviceBuffer(0) { device[0] = 11111; device[1] = 11111; }
790 typedef signed short Int16;
791 typedef signed int Int32;
792 typedef float Float32;
793 typedef double Float64;
795 std::ostringstream errorStream_;
796 std::string errorText_;
797 RtAudioErrorCallback errorCallback_;
802 Protected, api-specific method that attempts to open a device
803 with the given parameters. This function MUST be implemented by
804 all subclasses. If an error is encountered during the probe, a
805 "warning" message is reported and FAILURE is returned. A
806 successful probe is indicated by a return value of SUCCESS.
808 virtual bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
809 unsigned int firstChannel, unsigned int sampleRate,
810 RtAudioFormat format, unsigned int *bufferSize,
811 RtAudio::StreamOptions *options );
813 //! A protected function used to increment the stream time.
814 void tickStreamTime( void );
816 //! Protected common method to clear an RtApiStream structure.
817 void clearStreamInfo();
819 //! Protected common error method to allow global control over error handling.
820 RtAudioErrorType error( RtAudioErrorType type );
823 Protected method used to perform format, channel number, and/or interleaving
824 conversions between the user and device buffers.
826 void convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info );
828 //! Protected common method used to perform byte-swapping on buffers.
829 void byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format );
831 //! Protected common method that returns the number of bytes for a given format.
832 unsigned int formatBytes( RtAudioFormat format );
834 //! Protected common method that sets up the parameters for buffer conversion.
835 void setConvertInfo( StreamMode mode, unsigned int firstChannel );
838 // **************************************************************** //
840 // Inline RtAudio definitions.
842 // **************************************************************** //
844 inline RtAudio::Api RtAudio :: getCurrentApi( void ) { return rtapi_->getCurrentApi(); }
845 inline unsigned int RtAudio :: getDeviceCount( void ) { return rtapi_->getDeviceCount(); }
846 inline RtAudio::DeviceInfo RtAudio :: getDeviceInfo( unsigned int device ) { return rtapi_->getDeviceInfo( device ); }
847 inline unsigned int RtAudio :: getDefaultInputDevice( void ) { return rtapi_->getDefaultInputDevice(); }
848 inline unsigned int RtAudio :: getDefaultOutputDevice( void ) { return rtapi_->getDefaultOutputDevice(); }
849 inline void RtAudio :: closeStream( void ) { return rtapi_->closeStream(); }
850 inline RtAudioErrorType RtAudio :: startStream( void ) { return rtapi_->startStream(); }
851 inline RtAudioErrorType RtAudio :: stopStream( void ) { return rtapi_->stopStream(); }
852 inline RtAudioErrorType RtAudio :: abortStream( void ) { return rtapi_->abortStream(); }
853 inline bool RtAudio :: isStreamOpen( void ) const { return rtapi_->isStreamOpen(); }
854 inline bool RtAudio :: isStreamRunning( void ) const { return rtapi_->isStreamRunning(); }
855 inline long RtAudio :: getStreamLatency( void ) { return rtapi_->getStreamLatency(); }
856 inline unsigned int RtAudio :: getStreamSampleRate( void ) { return rtapi_->getStreamSampleRate(); }
857 inline double RtAudio :: getStreamTime( void ) { return rtapi_->getStreamTime(); }
858 inline void RtAudio :: setStreamTime( double time ) { return rtapi_->setStreamTime( time ); }
859 inline void RtAudio :: setErrorCallback( RtAudioErrorCallback errorCallback ) { rtapi_->setErrorCallback( errorCallback ); }
860 inline void RtAudio :: showWarnings( bool value ) { rtapi_->showWarnings( value ); }
862 // RtApi Subclass prototypes.
864 #if defined(__MACOSX_CORE__)
866 #include <CoreAudio/AudioHardware.h>
868 class RtApiCore: public RtApi
874 RtAudio::Api getCurrentApi( void ) { return RtAudio::MACOSX_CORE; }
875 unsigned int getDeviceCount( void );
876 RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
877 unsigned int getDefaultOutputDevice( void );
878 unsigned int getDefaultInputDevice( void );
879 void closeStream( void );
880 RtAudioErrorType startStream( void );
881 RtAudioErrorType stopStream( void );
882 RtAudioErrorType abortStream( void );
884 // This function is intended for internal use only. It must be
885 // public because it is called by the internal callback handler,
886 // which is not a member of RtAudio. External use of this function
887 // will most likely produce highly undesireable results!
888 bool callbackEvent( AudioDeviceID deviceId,
889 const AudioBufferList *inBufferList,
890 const AudioBufferList *outBufferList );
894 bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
895 unsigned int firstChannel, unsigned int sampleRate,
896 RtAudioFormat format, unsigned int *bufferSize,
897 RtAudio::StreamOptions *options );
898 static const char* getErrorCode( OSStatus code );
903 #if defined(__UNIX_JACK__)
905 class RtApiJack: public RtApi
911 RtAudio::Api getCurrentApi( void ) { return RtAudio::UNIX_JACK; }
912 unsigned int getDeviceCount( void );
913 RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
914 void closeStream( void );
915 RtAudioErrorType startStream( void );
916 RtAudioErrorType stopStream( void );
917 RtAudioErrorType abortStream( void );
919 // This function is intended for internal use only. It must be
920 // public because it is called by the internal callback handler,
921 // which is not a member of RtAudio. External use of this function
922 // will most likely produce highly undesireable results!
923 bool callbackEvent( unsigned long nframes );
927 bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
928 unsigned int firstChannel, unsigned int sampleRate,
929 RtAudioFormat format, unsigned int *bufferSize,
930 RtAudio::StreamOptions *options );
932 bool shouldAutoconnect_;
937 #if defined(__WINDOWS_ASIO__)
939 class RtApiAsio: public RtApi
945 RtAudio::Api getCurrentApi( void ) { return RtAudio::WINDOWS_ASIO; }
946 unsigned int getDeviceCount( void );
947 RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
948 void closeStream( void );
949 void startStream( void );
950 void stopStream( void );
951 void abortStream( void );
953 // This function is intended for internal use only. It must be
954 // public because it is called by the internal callback handler,
955 // which is not a member of RtAudio. External use of this function
956 // will most likely produce highly undesireable results!
957 bool callbackEvent( long bufferIndex );
961 std::vector<RtAudio::DeviceInfo> devices_;
962 void saveDeviceInfo( void );
964 bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
965 unsigned int firstChannel, unsigned int sampleRate,
966 RtAudioFormat format, unsigned int *bufferSize,
967 RtAudio::StreamOptions *options );
972 #if defined(__WINDOWS_DS__)
974 class RtApiDs: public RtApi
980 RtAudio::Api getCurrentApi( void ) { return RtAudio::WINDOWS_DS; }
981 unsigned int getDeviceCount( void );
982 unsigned int getDefaultOutputDevice( void );
983 unsigned int getDefaultInputDevice( void );
984 RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
985 void closeStream( void );
986 void startStream( void );
987 void stopStream( void );
988 void abortStream( void );
990 // This function is intended for internal use only. It must be
991 // public because it is called by the internal callback handler,
992 // which is not a member of RtAudio. External use of this function
993 // will most likely produce highly undesireable results!
994 void callbackEvent( void );
1000 long duplexPrerollBytes;
1001 std::vector<struct DsDevice> dsDevices;
1002 bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
1003 unsigned int firstChannel, unsigned int sampleRate,
1004 RtAudioFormat format, unsigned int *bufferSize,
1005 RtAudio::StreamOptions *options );
1010 #if defined(__WINDOWS_WASAPI__)
1012 struct IMMDeviceEnumerator;
1014 class RtApiWasapi : public RtApi
1018 virtual ~RtApiWasapi();
1020 RtAudio::Api getCurrentApi( void ) { return RtAudio::WINDOWS_WASAPI; }
1021 unsigned int getDeviceCount( void );
1022 RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
1023 unsigned int getDefaultOutputDevice( void );
1024 unsigned int getDefaultInputDevice( void );
1025 void closeStream( void );
1026 void startStream( void );
1027 void stopStream( void );
1028 void abortStream( void );
1031 bool coInitialized_;
1032 IMMDeviceEnumerator* deviceEnumerator_;
1034 bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
1035 unsigned int firstChannel, unsigned int sampleRate,
1036 RtAudioFormat format, unsigned int* bufferSize,
1037 RtAudio::StreamOptions* options );
1039 static DWORD WINAPI runWasapiThread( void* wasapiPtr );
1040 static DWORD WINAPI stopWasapiThread( void* wasapiPtr );
1041 static DWORD WINAPI abortWasapiThread( void* wasapiPtr );
1042 void wasapiThread();
1047 #if defined(__LINUX_ALSA__)
1049 class RtApiAlsa: public RtApi
1055 RtAudio::Api getCurrentApi() { return RtAudio::LINUX_ALSA; }
1056 unsigned int getDeviceCount( void );
1057 RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
1058 void closeStream( void );
1059 void startStream( void );
1060 void stopStream( void );
1061 void abortStream( void );
1063 // This function is intended for internal use only. It must be
1064 // public because it is called by the internal callback handler,
1065 // which is not a member of RtAudio. External use of this function
1066 // will most likely produce highly undesireable results!
1067 void callbackEvent( void );
1071 std::vector<RtAudio::DeviceInfo> devices_;
1072 void saveDeviceInfo( void );
1073 bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
1074 unsigned int firstChannel, unsigned int sampleRate,
1075 RtAudioFormat format, unsigned int *bufferSize,
1076 RtAudio::StreamOptions *options );
1081 #if defined(__LINUX_PULSE__)
1083 class RtApiPulse: public RtApi
1087 RtAudio::Api getCurrentApi() { return RtAudio::LINUX_PULSE; }
1088 unsigned int getDeviceCount( void );
1089 RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
1090 void closeStream( void );
1091 void startStream( void );
1092 void stopStream( void );
1093 void abortStream( void );
1095 // This function is intended for internal use only. It must be
1096 // public because it is called by the internal callback handler,
1097 // which is not a member of RtAudio. External use of this function
1098 // will most likely produce highly undesireable results!
1099 void callbackEvent( void );
1103 std::vector<RtAudio::DeviceInfo> devices_;
1104 void saveDeviceInfo( void );
1105 bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
1106 unsigned int firstChannel, unsigned int sampleRate,
1107 RtAudioFormat format, unsigned int *bufferSize,
1108 RtAudio::StreamOptions *options );
1113 #if defined(__LINUX_OSS__)
1115 class RtApiOss: public RtApi
1121 RtAudio::Api getCurrentApi() { return RtAudio::LINUX_OSS; }
1122 unsigned int getDeviceCount( void );
1123 RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
1124 void closeStream( void );
1125 void startStream( void );
1126 void stopStream( void );
1127 void abortStream( void );
1129 // This function is intended for internal use only. It must be
1130 // public because it is called by the internal callback handler,
1131 // which is not a member of RtAudio. External use of this function
1132 // will most likely produce highly undesireable results!
1133 void callbackEvent( void );
1137 bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
1138 unsigned int firstChannel, unsigned int sampleRate,
1139 RtAudioFormat format, unsigned int *bufferSize,
1140 RtAudio::StreamOptions *options );
1145 #if defined(__RTAUDIO_DUMMY__)
1147 class RtApiDummy: public RtApi
1151 RtApiDummy() { errorText_ = "RtApiDummy: This class provides no functionality."; error( RTAUDIO_WARNING ); }
1152 RtAudio::Api getCurrentApi( void ) { return RtAudio::RTAUDIO_DUMMY; }
1153 unsigned int getDeviceCount( void ) { return 0; }
1154 RtAudio::DeviceInfo getDeviceInfo( unsigned int /*device*/ ) { RtAudio::DeviceInfo info; return info; }
1155 void closeStream( void ) {}
1156 RtAudioErrorType startStream( void ) { return RTAUDIO_NO_ERROR; }
1157 RtAudioErrorType stopStream( void ) { return RTAUDIO_NO_ERROR; }
1158 RtAudioErrorType abortStream( void ) { return RTAUDIO_NO_ERROR; }
1162 bool probeDeviceOpen( unsigned int /*device*/, StreamMode /*mode*/, unsigned int /*channels*/,
1163 unsigned int /*firstChannel*/, unsigned int /*sampleRate*/,
1164 RtAudioFormat /*format*/, unsigned int * /*bufferSize*/,
1165 RtAudio::StreamOptions * /*options*/ ) { return false; }
1172 // Indentation settings for Vim and Emacs
1175 // c-basic-offset: 2
1176 // indent-tabs-mode: nil
1179 // vim: et sts=2 sw=2