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 and ASIO) operating systems.
10 RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/
12 RtAudio: realtime audio i/o C++ classes
13 Copyright (c) 2001-2012 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 /************************************************************************/
45 // RtAudio: Version 4.0.11
54 /*! \typedef typedef unsigned long RtAudioFormat;
55 \brief RtAudio data format type.
57 Support for signed integers and floats. Audio data fed to/from an
58 RtAudio stream is assumed to ALWAYS be in host byte order. The
59 internal routines will automatically take care of any necessary
60 byte-swapping between the host format and the soundcard. Thus,
61 endian-ness is not a concern in the following format definitions.
62 Note that 24-bit data is expected to be encapsulated in a 32-bit
65 - \e RTAUDIO_SINT8: 8-bit signed integer.
66 - \e RTAUDIO_SINT16: 16-bit signed integer.
67 - \e RTAUDIO_SINT24: Lower 3 bytes of 32-bit signed integer.
68 - \e RTAUDIO_SINT32: 32-bit signed integer.
69 - \e RTAUDIO_FLOAT32: Normalized between plus/minus 1.0.
70 - \e RTAUDIO_FLOAT64: Normalized between plus/minus 1.0.
72 typedef unsigned long RtAudioFormat;
73 static const RtAudioFormat RTAUDIO_SINT8 = 0x1; // 8-bit signed integer.
74 static const RtAudioFormat RTAUDIO_SINT16 = 0x2; // 16-bit signed integer.
75 static const RtAudioFormat RTAUDIO_SINT24 = 0x4; // Lower 3 bytes of 32-bit signed integer.
76 static const RtAudioFormat RTAUDIO_SINT32 = 0x8; // 32-bit signed integer.
77 static const RtAudioFormat RTAUDIO_FLOAT32 = 0x10; // Normalized between plus/minus 1.0.
78 static const RtAudioFormat RTAUDIO_FLOAT64 = 0x20; // Normalized between plus/minus 1.0.
80 /*! \typedef typedef unsigned long RtAudioStreamFlags;
81 \brief RtAudio stream option flags.
83 The following flags can be OR'ed together to allow a client to
84 make changes to the default stream behavior:
86 - \e RTAUDIO_NONINTERLEAVED: Use non-interleaved buffers (default = interleaved).
87 - \e RTAUDIO_MINIMIZE_LATENCY: Attempt to set stream parameters for lowest possible latency.
88 - \e RTAUDIO_HOG_DEVICE: Attempt grab device for exclusive use.
89 - \e RTAUDIO_ALSA_USE_DEFAULT: Use the "default" PCM device (ALSA 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 typedef unsigned int RtAudioStreamFlags;
123 static const RtAudioStreamFlags RTAUDIO_NONINTERLEAVED = 0x1; // Use non-interleaved buffers (default = interleaved).
124 static const RtAudioStreamFlags RTAUDIO_MINIMIZE_LATENCY = 0x2; // Attempt to set stream parameters for lowest possible latency.
125 static const RtAudioStreamFlags RTAUDIO_HOG_DEVICE = 0x4; // Attempt grab device and prevent use by others.
126 static const RtAudioStreamFlags RTAUDIO_SCHEDULE_REALTIME = 0x8; // Try to select realtime scheduling for callback thread.
127 static const RtAudioStreamFlags RTAUDIO_ALSA_USE_DEFAULT = 0x10; // Use the "default" PCM device (ALSA only).
129 /*! \typedef typedef unsigned long RtAudioStreamStatus;
130 \brief RtAudio stream status (over- or underflow) flags.
132 Notification of a stream over- or underflow is indicated by a
133 non-zero stream \c status argument in the RtAudioCallback function.
134 The stream status can be one of the following two options,
135 depending on whether the stream is open for output and/or input:
137 - \e RTAUDIO_INPUT_OVERFLOW: Input data was discarded because of an overflow condition at the driver.
138 - \e RTAUDIO_OUTPUT_UNDERFLOW: The output buffer ran low, likely producing a break in the output sound.
140 typedef unsigned int RtAudioStreamStatus;
141 static const RtAudioStreamStatus RTAUDIO_INPUT_OVERFLOW = 0x1; // Input data was discarded because of an overflow condition at the driver.
142 static const RtAudioStreamStatus RTAUDIO_OUTPUT_UNDERFLOW = 0x2; // The output buffer ran low, likely causing a gap in the output sound.
144 //! RtAudio callback function prototype.
146 All RtAudio clients must create a function of type RtAudioCallback
147 to read and/or write data from/to the audio stream. When the
148 underlying audio system is ready for new input or output data, this
149 function will be invoked.
151 \param outputBuffer For output (or duplex) streams, the client
152 should write \c nFrames of audio sample frames into this
153 buffer. This argument should be recast to the datatype
154 specified when the stream was opened. For input-only
155 streams, this argument will be NULL.
157 \param inputBuffer For input (or duplex) streams, this buffer will
158 hold \c nFrames of input audio sample frames. This
159 argument should be recast to the datatype specified when the
160 stream was opened. For output-only streams, this argument
163 \param nFrames The number of sample frames of input or output
164 data in the buffers. The actual buffer size in bytes is
165 dependent on the data type and number of channels in use.
167 \param streamTime The number of seconds that have elapsed since the
170 \param status If non-zero, this argument indicates a data overflow
171 or underflow condition for the stream. The particular
172 condition can be determined by comparison with the
173 RtAudioStreamStatus flags.
175 \param userData A pointer to optional data provided by the client
176 when opening the stream (default = NULL).
178 To continue normal stream operation, the RtAudioCallback function
179 should return a value of zero. To stop the stream and drain the
180 output buffer, the function should return a value of one. To abort
181 the stream immediately, the client should return a value of two.
183 typedef int (*RtAudioCallback)( void *outputBuffer, void *inputBuffer,
184 unsigned int nFrames,
186 RtAudioStreamStatus status,
190 // **************************************************************** //
192 // RtAudio class declaration.
194 // RtAudio is a "controller" used to select an available audio i/o
195 // interface. It presents a common API for the user to call but all
196 // functionality is implemented by the class RtApi and its
197 // subclasses. RtAudio creates an instance of an RtApi subclass
198 // based on the user's API choice. If no choice is made, RtAudio
199 // attempts to make a "logical" API selection.
201 // **************************************************************** //
209 //! Audio API specifier arguments.
211 UNSPECIFIED, /*!< Search for a working compiled API. */
212 LINUX_ALSA, /*!< The Advanced Linux Sound Architecture API. */
213 LINUX_PULSE, /*!< The Linux PulseAudio API. */
214 LINUX_OSS, /*!< The Linux Open Sound System API. */
215 UNIX_JACK, /*!< The Jack Low-Latency Audio Server API. */
216 MACOSX_CORE, /*!< Macintosh OS-X Core Audio API. */
217 WINDOWS_ASIO, /*!< The Steinberg Audio Stream I/O API. */
218 WINDOWS_DS, /*!< The Microsoft Direct Sound API. */
219 RTAUDIO_DUMMY /*!< A compilable but non-functional API. */
222 //! The public device information structure for returning queried values.
224 bool probed; /*!< true if the device capabilities were successfully probed. */
225 std::string name; /*!< Character string device identifier. */
226 unsigned int outputChannels; /*!< Maximum output channels supported by device. */
227 unsigned int inputChannels; /*!< Maximum input channels supported by device. */
228 unsigned int duplexChannels; /*!< Maximum simultaneous input/output channels supported by device. */
229 bool isDefaultOutput; /*!< true if this is the default output device. */
230 bool isDefaultInput; /*!< true if this is the default input device. */
231 std::vector<unsigned int> sampleRates; /*!< Supported sample rates (queried from list of standard rates). */
232 RtAudioFormat nativeFormats; /*!< Bit mask of supported data formats. */
234 // Default constructor.
236 :probed(false), outputChannels(0), inputChannels(0), duplexChannels(0),
237 isDefaultOutput(false), isDefaultInput(false), nativeFormats(0) {}
240 //! The structure for specifying input or ouput stream parameters.
241 struct StreamParameters {
242 unsigned int deviceId; /*!< Device index (0 to getDeviceCount() - 1). */
243 unsigned int nChannels; /*!< Number of channels. */
244 unsigned int firstChannel; /*!< First channel index on device (default = 0). */
246 // Default constructor.
248 : deviceId(0), nChannels(0), firstChannel(0) {}
251 //! The structure for specifying stream options.
253 The following flags can be OR'ed together to allow a client to
254 make changes to the default stream behavior:
256 - \e RTAUDIO_NONINTERLEAVED: Use non-interleaved buffers (default = interleaved).
257 - \e RTAUDIO_MINIMIZE_LATENCY: Attempt to set stream parameters for lowest possible latency.
258 - \e RTAUDIO_HOG_DEVICE: Attempt grab device for exclusive use.
259 - \e RTAUDIO_SCHEDULE_REALTIME: Attempt to select realtime scheduling for callback thread.
260 - \e RTAUDIO_ALSA_USE_DEFAULT: Use the "default" PCM device (ALSA only).
262 By default, RtAudio streams pass and receive audio data from the
263 client in an interleaved format. By passing the
264 RTAUDIO_NONINTERLEAVED flag to the openStream() function, audio
265 data will instead be presented in non-interleaved buffers. In
266 this case, each buffer argument in the RtAudioCallback function
267 will point to a single array of data, with \c nFrames samples for
268 each channel concatenated back-to-back. For example, the first
269 sample of data for the second channel would be located at index \c
270 nFrames (assuming the \c buffer pointer was recast to the correct
271 data type for the stream).
273 Certain audio APIs offer a number of parameters that influence the
274 I/O latency of a stream. By default, RtAudio will attempt to set
275 these parameters internally for robust (glitch-free) performance
276 (though some APIs, like Windows Direct Sound, make this difficult).
277 By passing the RTAUDIO_MINIMIZE_LATENCY flag to the openStream()
278 function, internal stream settings will be influenced in an attempt
279 to minimize stream latency, though possibly at the expense of stream
282 If the RTAUDIO_HOG_DEVICE flag is set, RtAudio will attempt to
283 open the input and/or output stream device(s) for exclusive use.
284 Note that this is not possible with all supported audio APIs.
286 If the RTAUDIO_SCHEDULE_REALTIME flag is set, RtAudio will attempt
287 to select realtime scheduling (round-robin) for the callback thread.
288 The \c priority parameter will only be used if the RTAUDIO_SCHEDULE_REALTIME
289 flag is set. It defines the thread's realtime priority.
291 If the RTAUDIO_ALSA_USE_DEFAULT flag is set, RtAudio will attempt to
292 open the "default" PCM device when using the ALSA API. Note that this
293 will override any specified input or output device id.
295 The \c numberOfBuffers parameter can be used to control stream
296 latency in the Windows DirectSound, Linux OSS, and Linux Alsa APIs
297 only. A value of two is usually the smallest allowed. Larger
298 numbers can potentially result in more robust stream performance,
299 though likely at the cost of stream latency. The value set by the
300 user is replaced during execution of the RtAudio::openStream()
301 function by the value actually used by the system.
303 The \c streamName parameter can be used to set the client name
304 when using the Jack API. By default, the client name is set to
305 RtApiJack. However, if you wish to create multiple instances of
306 RtAudio with Jack, each instance must have a unique client name.
308 struct StreamOptions {
309 RtAudioStreamFlags flags; /*!< A bit-mask of stream flags (RTAUDIO_NONINTERLEAVED, RTAUDIO_MINIMIZE_LATENCY, RTAUDIO_HOG_DEVICE, RTAUDIO_ALSA_USE_DEFAULT). */
310 unsigned int numberOfBuffers; /*!< Number of stream buffers. */
311 std::string streamName; /*!< A stream name (currently used only in Jack). */
312 int priority; /*!< Scheduling priority of callback thread (only used with flag RTAUDIO_SCHEDULE_REALTIME). */
314 // Default constructor.
316 : flags(0), numberOfBuffers(0), priority(0) {}
319 //! A static function to determine the available compiled audio APIs.
321 The values returned in the std::vector can be compared against
322 the enumerated list values. Note that there can be more than one
323 API compiled for certain operating systems.
325 static void getCompiledApi( std::vector<RtAudio::Api> &apis ) throw();
327 //! The class constructor.
329 The constructor performs minor initialization tasks. No exceptions
332 If no API argument is specified and multiple API support has been
333 compiled, the default order of use is JACK, ALSA, OSS (Linux
334 systems) and ASIO, DS (Windows systems).
336 RtAudio( RtAudio::Api api=UNSPECIFIED ) throw();
340 If a stream is running or open, it will be stopped and closed
345 //! Returns the audio API specifier for the current instance of RtAudio.
346 RtAudio::Api getCurrentApi( void ) throw();
348 //! A public function that queries for the number of audio devices available.
350 This function performs a system query of available devices each time it
351 is called, thus supporting devices connected \e after instantiation. If
352 a system error occurs during processing, a warning will be issued.
354 unsigned int getDeviceCount( void ) throw();
356 //! Return an RtAudio::DeviceInfo structure for a specified device number.
359 Any device integer between 0 and getDeviceCount() - 1 is valid.
360 If an invalid argument is provided, an RtError (type = INVALID_USE)
361 will be thrown. If a device is busy or otherwise unavailable, the
362 structure member "probed" will have a value of "false" and all
363 other members are undefined. If the specified device is the
364 current default input or output device, the corresponding
365 "isDefault" member will have a value of "true".
367 RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
369 //! A function that returns the index of the default output device.
371 If the underlying audio API does not provide a "default
372 device", or if no devices are available, the return value will be
373 0. Note that this is a valid device identifier and it is the
374 client's responsibility to verify that a device is available
375 before attempting to open a stream.
377 unsigned int getDefaultOutputDevice( void ) throw();
379 //! A function that returns the index of the default input device.
381 If the underlying audio API does not provide a "default
382 device", or if no devices are available, the return value will be
383 0. Note that this is a valid device identifier and it is the
384 client's responsibility to verify that a device is available
385 before attempting to open a stream.
387 unsigned int getDefaultInputDevice( void ) throw();
389 //! A public function for opening a stream with the specified parameters.
391 An RtError (type = SYSTEM_ERROR) is thrown if a stream cannot be
392 opened with the specified parameters or an error occurs during
393 processing. An RtError (type = INVALID_USE) is thrown if any
394 invalid device ID or channel number parameters are specified.
396 \param outputParameters Specifies output stream parameters to use
397 when opening a stream, including a device ID, number of channels,
398 and starting channel number. For input-only streams, this
399 argument should be NULL. The device ID is an index value between
400 0 and getDeviceCount() - 1.
401 \param inputParameters Specifies input stream parameters to use
402 when opening a stream, including a device ID, number of channels,
403 and starting channel number. For output-only streams, this
404 argument should be NULL. The device ID is an index value between
405 0 and getDeviceCount() - 1.
406 \param format An RtAudioFormat specifying the desired sample data format.
407 \param sampleRate The desired sample rate (sample frames per second).
408 \param *bufferFrames A pointer to a value indicating the desired
409 internal buffer size in sample frames. The actual value
410 used by the device is returned via the same pointer. A
411 value of zero can be specified, in which case the lowest
412 allowable value is determined.
413 \param callback A client-defined function that will be invoked
414 when input data is available and/or output data is needed.
415 \param userData An optional pointer to data that can be accessed
416 from within the callback function.
417 \param options An optional pointer to a structure containing various
418 global stream options, including a list of OR'ed RtAudioStreamFlags
419 and a suggested number of stream buffers that can be used to
420 control stream latency. More buffers typically result in more
421 robust performance, though at a cost of greater latency. If a
422 value of zero is specified, a system-specific median value is
423 chosen. If the RTAUDIO_MINIMIZE_LATENCY flag bit is set, the
424 lowest allowable value is used. The actual value used is
425 returned via the structure argument. The parameter is API dependent.
427 void openStream( RtAudio::StreamParameters *outputParameters,
428 RtAudio::StreamParameters *inputParameters,
429 RtAudioFormat format, unsigned int sampleRate,
430 unsigned int *bufferFrames, RtAudioCallback callback,
431 void *userData = NULL, RtAudio::StreamOptions *options = NULL );
433 //! A function that closes a stream and frees any associated stream memory.
435 If a stream is not open, this function issues a warning and
436 returns (no exception is thrown).
438 void closeStream( void ) throw();
440 //! A function that starts a stream.
442 An RtError (type = SYSTEM_ERROR) is thrown if an error occurs
443 during processing. An RtError (type = INVALID_USE) is thrown if a
444 stream is not open. A warning is issued if the stream is already
447 void startStream( void );
449 //! Stop a stream, allowing any samples remaining in the output queue to be played.
451 An RtError (type = SYSTEM_ERROR) is thrown if an error occurs
452 during processing. An RtError (type = INVALID_USE) is thrown if a
453 stream is not open. A warning is issued if the stream is already
456 void stopStream( void );
458 //! Stop a stream, discarding any samples remaining in the input/output queue.
460 An RtError (type = SYSTEM_ERROR) is thrown if an error occurs
461 during processing. An RtError (type = INVALID_USE) is thrown if a
462 stream is not open. A warning is issued if the stream is already
465 void abortStream( void );
467 //! Returns true if a stream is open and false if not.
468 bool isStreamOpen( void ) const throw();
470 //! Returns true if the stream is running and false if it is stopped or not open.
471 bool isStreamRunning( void ) const throw();
473 //! Returns the number of elapsed seconds since the stream was started.
475 If a stream is not open, an RtError (type = INVALID_USE) will be thrown.
477 double getStreamTime( void );
479 //! Returns the internal stream latency in sample frames.
481 The stream latency refers to delay in audio input and/or output
482 caused by internal buffering by the audio system and/or hardware.
483 For duplex streams, the returned value will represent the sum of
484 the input and output latencies. If a stream is not open, an
485 RtError (type = INVALID_USE) will be thrown. If the API does not
486 report latency, the return value will be zero.
488 long getStreamLatency( void );
490 //! Returns actual sample rate in use by the stream.
492 On some systems, the sample rate used may be slightly different
493 than that specified in the stream parameters. If a stream is not
494 open, an RtError (type = INVALID_USE) will be thrown.
496 unsigned int getStreamSampleRate( void );
498 //! Specify whether warning messages should be printed to stderr.
499 void showWarnings( bool value = true ) throw();
503 void openRtApi( RtAudio::Api api );
507 // Operating system dependent thread functionality.
508 #if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__)
512 typedef unsigned long ThreadHandle;
513 typedef CRITICAL_SECTION StreamMutex;
515 #elif defined(__LINUX_ALSA__) || defined(__LINUX_PULSE__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__)
516 // Using pthread library for various flavors of unix.
519 typedef pthread_t ThreadHandle;
520 typedef pthread_mutex_t StreamMutex;
522 #else // Setup for "dummy" behavior
524 #define __RTAUDIO_DUMMY__
525 typedef int ThreadHandle;
526 typedef int StreamMutex;
530 // This global structure type is used to pass callback information
531 // between the private RtAudio stream structure and global callback
532 // handling functions.
533 struct CallbackInfo {
534 void *object; // Used as a "this" pointer.
538 void *apiInfo; // void pointer for API specific callback information
543 // Default constructor.
545 :object(0), callback(0), userData(0), apiInfo(0), isRunning(false), doRealtime(false) {}
548 // **************************************************************** //
550 // RtApi class declaration.
552 // Subclasses of RtApi contain all API- and OS-specific code necessary
553 // to fully implement the RtAudio API.
555 // Note that RtApi is an abstract base class and cannot be
556 // explicitly instantiated. The class RtAudio will create an
557 // instance of an RtApi subclass (RtApiOss, RtApiAlsa,
558 // RtApiJack, RtApiCore, RtApiDs, or RtApiAsio).
560 // **************************************************************** //
562 #pragma pack(push, 1)
571 S24& operator = ( const int& i ) {
572 c3[0] = (i & 0x000000ff);
573 c3[1] = (i & 0x0000ff00) >> 8;
574 c3[2] = (i & 0x00ff0000) >> 16;
578 S24( const S24& v ) { *this = v; }
579 S24( const double& d ) { *this = (int) d; }
580 S24( const float& f ) { *this = (int) f; }
581 S24( const signed short& s ) { *this = (int) s; }
582 S24( const char& c ) { *this = (int) c; }
585 int i = c3[0] | (c3[1] << 8) | (c3[2] << 16);
586 if (i & 0x800000) i |= ~0xffffff;
592 #if defined( HAVE_GETTIMEOFDAY )
593 #include <sys/time.h>
604 virtual RtAudio::Api getCurrentApi( void ) = 0;
605 virtual unsigned int getDeviceCount( void ) = 0;
606 virtual RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) = 0;
607 virtual unsigned int getDefaultInputDevice( void );
608 virtual unsigned int getDefaultOutputDevice( void );
609 void openStream( RtAudio::StreamParameters *outputParameters,
610 RtAudio::StreamParameters *inputParameters,
611 RtAudioFormat format, unsigned int sampleRate,
612 unsigned int *bufferFrames, RtAudioCallback callback,
613 void *userData, RtAudio::StreamOptions *options );
614 virtual void closeStream( void );
615 virtual void startStream( void ) = 0;
616 virtual void stopStream( void ) = 0;
617 virtual void abortStream( void ) = 0;
618 long getStreamLatency( void );
619 unsigned int getStreamSampleRate( void );
620 virtual double getStreamTime( void );
621 bool isStreamOpen( void ) const { return stream_.state != STREAM_CLOSED; };
622 bool isStreamRunning( void ) const { return stream_.state == STREAM_RUNNING; };
623 void showWarnings( bool value ) { showWarnings_ = value; };
628 static const unsigned int MAX_SAMPLE_RATES;
629 static const unsigned int SAMPLE_RATES[];
631 enum { FAILURE, SUCCESS };
647 // A protected structure used for buffer conversion.
651 RtAudioFormat inFormat, outFormat;
652 std::vector<int> inOffset;
653 std::vector<int> outOffset;
656 // A protected structure for audio streams.
658 unsigned int device[2]; // Playback and record, respectively.
659 void *apiHandle; // void pointer for API specific stream handle information
660 StreamMode mode; // OUTPUT, INPUT, or DUPLEX.
661 StreamState state; // STOPPED, RUNNING, or CLOSED
662 char *userBuffer[2]; // Playback and record, respectively.
664 bool doConvertBuffer[2]; // Playback and record, respectively.
665 bool userInterleaved;
666 bool deviceInterleaved[2]; // Playback and record, respectively.
667 bool doByteSwap[2]; // Playback and record, respectively.
668 unsigned int sampleRate;
669 unsigned int bufferSize;
670 unsigned int nBuffers;
671 unsigned int nUserChannels[2]; // Playback and record, respectively.
672 unsigned int nDeviceChannels[2]; // Playback and record channels, respectively.
673 unsigned int channelOffset[2]; // Playback and record, respectively.
674 unsigned long latency[2]; // Playback and record, respectively.
675 RtAudioFormat userFormat;
676 RtAudioFormat deviceFormat[2]; // Playback and record, respectively.
678 CallbackInfo callbackInfo;
679 ConvertInfo convertInfo[2];
680 double streamTime; // Number of elapsed seconds since the stream started.
682 #if defined(HAVE_GETTIMEOFDAY)
683 struct timeval lastTickTimestamp;
687 :apiHandle(0), deviceBuffer(0) { device[0] = 11111; device[1] = 11111; }
691 typedef signed short Int16;
692 typedef signed int Int32;
693 typedef float Float32;
694 typedef double Float64;
696 std::ostringstream errorStream_;
697 std::string errorText_;
702 Protected, api-specific method that attempts to open a device
703 with the given parameters. This function MUST be implemented by
704 all subclasses. If an error is encountered during the probe, a
705 "warning" message is reported and FAILURE is returned. A
706 successful probe is indicated by a return value of SUCCESS.
708 virtual bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
709 unsigned int firstChannel, unsigned int sampleRate,
710 RtAudioFormat format, unsigned int *bufferSize,
711 RtAudio::StreamOptions *options );
713 //! A protected function used to increment the stream time.
714 void tickStreamTime( void );
716 //! Protected common method to clear an RtApiStream structure.
717 void clearStreamInfo();
720 Protected common method that throws an RtError (type =
721 INVALID_USE) if a stream is not open.
723 void verifyStream( void );
725 //! Protected common error method to allow global control over error handling.
726 void error( RtError::Type type );
729 Protected method used to perform format, channel number, and/or interleaving
730 conversions between the user and device buffers.
732 void convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info );
734 //! Protected common method used to perform byte-swapping on buffers.
735 void byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format );
737 //! Protected common method that returns the number of bytes for a given format.
738 unsigned int formatBytes( RtAudioFormat format );
740 //! Protected common method that sets up the parameters for buffer conversion.
741 void setConvertInfo( StreamMode mode, unsigned int firstChannel );
744 // **************************************************************** //
746 // Inline RtAudio definitions.
748 // **************************************************************** //
750 inline RtAudio::Api RtAudio :: getCurrentApi( void ) throw() { return rtapi_->getCurrentApi(); }
751 inline unsigned int RtAudio :: getDeviceCount( void ) throw() { return rtapi_->getDeviceCount(); }
752 inline RtAudio::DeviceInfo RtAudio :: getDeviceInfo( unsigned int device ) { return rtapi_->getDeviceInfo( device ); }
753 inline unsigned int RtAudio :: getDefaultInputDevice( void ) throw() { return rtapi_->getDefaultInputDevice(); }
754 inline unsigned int RtAudio :: getDefaultOutputDevice( void ) throw() { return rtapi_->getDefaultOutputDevice(); }
755 inline void RtAudio :: closeStream( void ) throw() { return rtapi_->closeStream(); }
756 inline void RtAudio :: startStream( void ) { return rtapi_->startStream(); }
757 inline void RtAudio :: stopStream( void ) { return rtapi_->stopStream(); }
758 inline void RtAudio :: abortStream( void ) { return rtapi_->abortStream(); }
759 inline bool RtAudio :: isStreamOpen( void ) const throw() { return rtapi_->isStreamOpen(); }
760 inline bool RtAudio :: isStreamRunning( void ) const throw() { return rtapi_->isStreamRunning(); }
761 inline long RtAudio :: getStreamLatency( void ) { return rtapi_->getStreamLatency(); }
762 inline unsigned int RtAudio :: getStreamSampleRate( void ) { return rtapi_->getStreamSampleRate(); };
763 inline double RtAudio :: getStreamTime( void ) { return rtapi_->getStreamTime(); }
764 inline void RtAudio :: showWarnings( bool value ) throw() { rtapi_->showWarnings( value ); }
766 // RtApi Subclass prototypes.
768 #if defined(__MACOSX_CORE__)
770 #include <CoreAudio/AudioHardware.h>
772 class RtApiCore: public RtApi
778 RtAudio::Api getCurrentApi( void ) { return RtAudio::MACOSX_CORE; };
779 unsigned int getDeviceCount( void );
780 RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
781 unsigned int getDefaultOutputDevice( void );
782 unsigned int getDefaultInputDevice( void );
783 void closeStream( void );
784 void startStream( void );
785 void stopStream( void );
786 void abortStream( void );
787 long getStreamLatency( void );
789 // This function is intended for internal use only. It must be
790 // public because it is called by the internal callback handler,
791 // which is not a member of RtAudio. External use of this function
792 // will most likely produce highly undesireable results!
793 bool callbackEvent( AudioDeviceID deviceId,
794 const AudioBufferList *inBufferList,
795 const AudioBufferList *outBufferList );
799 bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
800 unsigned int firstChannel, unsigned int sampleRate,
801 RtAudioFormat format, unsigned int *bufferSize,
802 RtAudio::StreamOptions *options );
803 static const char* getErrorCode( OSStatus code );
808 #if defined(__UNIX_JACK__)
810 class RtApiJack: public RtApi
816 RtAudio::Api getCurrentApi( void ) { return RtAudio::UNIX_JACK; };
817 unsigned int getDeviceCount( void );
818 RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
819 void closeStream( void );
820 void startStream( void );
821 void stopStream( void );
822 void abortStream( void );
823 long getStreamLatency( void );
825 // This function is intended for internal use only. It must be
826 // public because it is called by the internal callback handler,
827 // which is not a member of RtAudio. External use of this function
828 // will most likely produce highly undesireable results!
829 bool callbackEvent( unsigned long nframes );
833 bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
834 unsigned int firstChannel, unsigned int sampleRate,
835 RtAudioFormat format, unsigned int *bufferSize,
836 RtAudio::StreamOptions *options );
841 #if defined(__WINDOWS_ASIO__)
843 class RtApiAsio: public RtApi
849 RtAudio::Api getCurrentApi( void ) { return RtAudio::WINDOWS_ASIO; };
850 unsigned int getDeviceCount( void );
851 RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
852 void closeStream( void );
853 void startStream( void );
854 void stopStream( void );
855 void abortStream( void );
856 long getStreamLatency( void );
858 // This function is intended for internal use only. It must be
859 // public because it is called by the internal callback handler,
860 // which is not a member of RtAudio. External use of this function
861 // will most likely produce highly undesireable results!
862 bool callbackEvent( long bufferIndex );
866 std::vector<RtAudio::DeviceInfo> devices_;
867 void saveDeviceInfo( void );
869 bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
870 unsigned int firstChannel, unsigned int sampleRate,
871 RtAudioFormat format, unsigned int *bufferSize,
872 RtAudio::StreamOptions *options );
877 #if defined(__WINDOWS_DS__)
879 class RtApiDs: public RtApi
885 RtAudio::Api getCurrentApi( void ) { return RtAudio::WINDOWS_DS; };
886 unsigned int getDeviceCount( void );
887 unsigned int getDefaultOutputDevice( void );
888 unsigned int getDefaultInputDevice( void );
889 RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
890 void closeStream( void );
891 void startStream( void );
892 void stopStream( void );
893 void abortStream( void );
894 long getStreamLatency( void );
896 // This function is intended for internal use only. It must be
897 // public because it is called by the internal callback handler,
898 // which is not a member of RtAudio. External use of this function
899 // will most likely produce highly undesireable results!
900 void callbackEvent( void );
906 long duplexPrerollBytes;
907 bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
908 unsigned int firstChannel, unsigned int sampleRate,
909 RtAudioFormat format, unsigned int *bufferSize,
910 RtAudio::StreamOptions *options );
915 #if defined(__LINUX_ALSA__)
917 class RtApiAlsa: public RtApi
923 RtAudio::Api getCurrentApi() { return RtAudio::LINUX_ALSA; };
924 unsigned int getDeviceCount( void );
925 RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
926 void closeStream( void );
927 void startStream( void );
928 void stopStream( void );
929 void abortStream( void );
931 // This function is intended for internal use only. It must be
932 // public because it is called by the internal callback handler,
933 // which is not a member of RtAudio. External use of this function
934 // will most likely produce highly undesireable results!
935 void callbackEvent( void );
939 std::vector<RtAudio::DeviceInfo> devices_;
940 void saveDeviceInfo( void );
941 bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
942 unsigned int firstChannel, unsigned int sampleRate,
943 RtAudioFormat format, unsigned int *bufferSize,
944 RtAudio::StreamOptions *options );
949 #if defined(__LINUX_PULSE__)
951 class RtApiPulse: public RtApi
955 RtAudio::Api getCurrentApi() { return RtAudio::LINUX_PULSE; };
956 unsigned int getDeviceCount( void );
957 RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
958 void closeStream( void );
959 void startStream( void );
960 void stopStream( void );
961 void abortStream( void );
963 // This function is intended for internal use only. It must be
964 // public because it is called by the internal callback handler,
965 // which is not a member of RtAudio. External use of this function
966 // will most likely produce highly undesireable results!
967 void callbackEvent( void );
971 std::vector<RtAudio::DeviceInfo> devices_;
972 void saveDeviceInfo( void );
973 bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
974 unsigned int firstChannel, unsigned int sampleRate,
975 RtAudioFormat format, unsigned int *bufferSize,
976 RtAudio::StreamOptions *options );
981 #if defined(__LINUX_OSS__)
983 class RtApiOss: public RtApi
989 RtAudio::Api getCurrentApi() { return RtAudio::LINUX_OSS; };
990 unsigned int getDeviceCount( void );
991 RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
992 void closeStream( void );
993 void startStream( void );
994 void stopStream( void );
995 void abortStream( void );
997 // This function is intended for internal use only. It must be
998 // public because it is called by the internal callback handler,
999 // which is not a member of RtAudio. External use of this function
1000 // will most likely produce highly undesireable results!
1001 void callbackEvent( void );
1005 bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
1006 unsigned int firstChannel, unsigned int sampleRate,
1007 RtAudioFormat format, unsigned int *bufferSize,
1008 RtAudio::StreamOptions *options );
1013 #if defined(__RTAUDIO_DUMMY__)
1015 class RtApiDummy: public RtApi
1019 RtApiDummy() { errorText_ = "RtApiDummy: This class provides no functionality."; error( RtError::WARNING ); };
1020 RtAudio::Api getCurrentApi( void ) { return RtAudio::RTAUDIO_DUMMY; };
1021 unsigned int getDeviceCount( void ) { return 0; };
1022 RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) { RtAudio::DeviceInfo info; return info; };
1023 void closeStream( void ) {};
1024 void startStream( void ) {};
1025 void stopStream( void ) {};
1026 void abortStream( void ) {};
1030 bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
1031 unsigned int firstChannel, unsigned int sampleRate,
1032 RtAudioFormat format, unsigned int *bufferSize,
1033 RtAudio::StreamOptions *options ) { return false; };
1040 // Indentation settings for Vim and Emacs
1043 // c-basic-offset: 2
1044 // indent-tabs-mode: nil
1047 // vim: et sts=2 sw=2