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_OSS, /*!< The Linux Open Sound System API. */
214 UNIX_JACK, /*!< The Jack Low-Latency Audio Server API. */
215 MACOSX_CORE, /*!< Macintosh OS-X Core Audio API. */
216 WINDOWS_ASIO, /*!< The Steinberg Audio Stream I/O API. */
217 WINDOWS_DS, /*!< The Microsoft Direct Sound API. */
218 RTAUDIO_DUMMY /*!< A compilable but non-functional API. */
221 //! The public device information structure for returning queried values.
223 bool probed; /*!< true if the device capabilities were successfully probed. */
224 std::string name; /*!< Character string device identifier. */
225 unsigned int outputChannels; /*!< Maximum output channels supported by device. */
226 unsigned int inputChannels; /*!< Maximum input channels supported by device. */
227 unsigned int duplexChannels; /*!< Maximum simultaneous input/output channels supported by device. */
228 bool isDefaultOutput; /*!< true if this is the default output device. */
229 bool isDefaultInput; /*!< true if this is the default input device. */
230 std::vector<unsigned int> sampleRates; /*!< Supported sample rates (queried from list of standard rates). */
231 RtAudioFormat nativeFormats; /*!< Bit mask of supported data formats. */
233 // Default constructor.
235 :probed(false), outputChannels(0), inputChannels(0), duplexChannels(0),
236 isDefaultOutput(false), isDefaultInput(false), nativeFormats(0) {}
239 //! The structure for specifying input or ouput stream parameters.
240 struct StreamParameters {
241 unsigned int deviceId; /*!< Device index (0 to getDeviceCount() - 1). */
242 unsigned int nChannels; /*!< Number of channels. */
243 unsigned int firstChannel; /*!< First channel index on device (default = 0). */
245 // Default constructor.
247 : deviceId(0), nChannels(0), firstChannel(0) {}
250 //! The structure for specifying stream options.
252 The following flags can be OR'ed together to allow a client to
253 make changes to the default stream behavior:
255 - \e RTAUDIO_NONINTERLEAVED: Use non-interleaved buffers (default = interleaved).
256 - \e RTAUDIO_MINIMIZE_LATENCY: Attempt to set stream parameters for lowest possible latency.
257 - \e RTAUDIO_HOG_DEVICE: Attempt grab device for exclusive use.
258 - \e RTAUDIO_SCHEDULE_REALTIME: Attempt to select realtime scheduling for callback thread.
259 - \e RTAUDIO_ALSA_USE_DEFAULT: Use the "default" PCM device (ALSA only).
261 By default, RtAudio streams pass and receive audio data from the
262 client in an interleaved format. By passing the
263 RTAUDIO_NONINTERLEAVED flag to the openStream() function, audio
264 data will instead be presented in non-interleaved buffers. In
265 this case, each buffer argument in the RtAudioCallback function
266 will point to a single array of data, with \c nFrames samples for
267 each channel concatenated back-to-back. For example, the first
268 sample of data for the second channel would be located at index \c
269 nFrames (assuming the \c buffer pointer was recast to the correct
270 data type for the stream).
272 Certain audio APIs offer a number of parameters that influence the
273 I/O latency of a stream. By default, RtAudio will attempt to set
274 these parameters internally for robust (glitch-free) performance
275 (though some APIs, like Windows Direct Sound, make this difficult).
276 By passing the RTAUDIO_MINIMIZE_LATENCY flag to the openStream()
277 function, internal stream settings will be influenced in an attempt
278 to minimize stream latency, though possibly at the expense of stream
281 If the RTAUDIO_HOG_DEVICE flag is set, RtAudio will attempt to
282 open the input and/or output stream device(s) for exclusive use.
283 Note that this is not possible with all supported audio APIs.
285 If the RTAUDIO_SCHEDULE_REALTIME flag is set, RtAudio will attempt
286 to select realtime scheduling (round-robin) for the callback thread.
287 The \c priority parameter will only be used if the RTAUDIO_SCHEDULE_REALTIME
288 flag is set. It defines the thread's realtime priority.
290 If the RTAUDIO_ALSA_USE_DEFAULT flag is set, RtAudio will attempt to
291 open the "default" PCM device when using the ALSA API. Note that this
292 will override any specified input or output device id.
294 The \c numberOfBuffers parameter can be used to control stream
295 latency in the Windows DirectSound, Linux OSS, and Linux Alsa APIs
296 only. A value of two is usually the smallest allowed. Larger
297 numbers can potentially result in more robust stream performance,
298 though likely at the cost of stream latency. The value set by the
299 user is replaced during execution of the RtAudio::openStream()
300 function by the value actually used by the system.
302 The \c streamName parameter can be used to set the client name
303 when using the Jack API. By default, the client name is set to
304 RtApiJack. However, if you wish to create multiple instances of
305 RtAudio with Jack, each instance must have a unique client name.
307 struct StreamOptions {
308 RtAudioStreamFlags flags; /*!< A bit-mask of stream flags (RTAUDIO_NONINTERLEAVED, RTAUDIO_MINIMIZE_LATENCY, RTAUDIO_HOG_DEVICE, RTAUDIO_ALSA_USE_DEFAULT). */
309 unsigned int numberOfBuffers; /*!< Number of stream buffers. */
310 std::string streamName; /*!< A stream name (currently used only in Jack). */
311 int priority; /*!< Scheduling priority of callback thread (only used with flag RTAUDIO_SCHEDULE_REALTIME). */
313 // Default constructor.
315 : flags(0), numberOfBuffers(0), priority(0) {}
318 //! A static function to determine the available compiled audio APIs.
320 The values returned in the std::vector can be compared against
321 the enumerated list values. Note that there can be more than one
322 API compiled for certain operating systems.
324 static void getCompiledApi( std::vector<RtAudio::Api> &apis ) throw();
326 //! The class constructor.
328 The constructor performs minor initialization tasks. No exceptions
331 If no API argument is specified and multiple API support has been
332 compiled, the default order of use is JACK, ALSA, OSS (Linux
333 systems) and ASIO, DS (Windows systems).
335 RtAudio( RtAudio::Api api=UNSPECIFIED ) throw();
339 If a stream is running or open, it will be stopped and closed
344 //! Returns the audio API specifier for the current instance of RtAudio.
345 RtAudio::Api getCurrentApi( void ) throw();
347 //! A public function that queries for the number of audio devices available.
349 This function performs a system query of available devices each time it
350 is called, thus supporting devices connected \e after instantiation. If
351 a system error occurs during processing, a warning will be issued.
353 unsigned int getDeviceCount( void ) throw();
355 //! Return an RtAudio::DeviceInfo structure for a specified device number.
358 Any device integer between 0 and getDeviceCount() - 1 is valid.
359 If an invalid argument is provided, an RtError (type = INVALID_USE)
360 will be thrown. If a device is busy or otherwise unavailable, the
361 structure member "probed" will have a value of "false" and all
362 other members are undefined. If the specified device is the
363 current default input or output device, the corresponding
364 "isDefault" member will have a value of "true".
366 RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
368 //! A function that returns the index of the default output device.
370 If the underlying audio API does not provide a "default
371 device", or if no devices are available, the return value will be
372 0. Note that this is a valid device identifier and it is the
373 client's responsibility to verify that a device is available
374 before attempting to open a stream.
376 unsigned int getDefaultOutputDevice( void ) throw();
378 //! A function that returns the index of the default input device.
380 If the underlying audio API does not provide a "default
381 device", or if no devices are available, the return value will be
382 0. Note that this is a valid device identifier and it is the
383 client's responsibility to verify that a device is available
384 before attempting to open a stream.
386 unsigned int getDefaultInputDevice( void ) throw();
388 //! A public function for opening a stream with the specified parameters.
390 An RtError (type = SYSTEM_ERROR) is thrown if a stream cannot be
391 opened with the specified parameters or an error occurs during
392 processing. An RtError (type = INVALID_USE) is thrown if any
393 invalid device ID or channel number parameters are specified.
395 \param outputParameters Specifies output stream parameters to use
396 when opening a stream, including a device ID, number of channels,
397 and starting channel number. For input-only streams, this
398 argument should be NULL. The device ID is an index value between
399 0 and getDeviceCount() - 1.
400 \param inputParameters Specifies input stream parameters to use
401 when opening a stream, including a device ID, number of channels,
402 and starting channel number. For output-only streams, this
403 argument should be NULL. The device ID is an index value between
404 0 and getDeviceCount() - 1.
405 \param format An RtAudioFormat specifying the desired sample data format.
406 \param sampleRate The desired sample rate (sample frames per second).
407 \param *bufferFrames A pointer to a value indicating the desired
408 internal buffer size in sample frames. The actual value
409 used by the device is returned via the same pointer. A
410 value of zero can be specified, in which case the lowest
411 allowable value is determined.
412 \param callback A client-defined function that will be invoked
413 when input data is available and/or output data is needed.
414 \param userData An optional pointer to data that can be accessed
415 from within the callback function.
416 \param options An optional pointer to a structure containing various
417 global stream options, including a list of OR'ed RtAudioStreamFlags
418 and a suggested number of stream buffers that can be used to
419 control stream latency. More buffers typically result in more
420 robust performance, though at a cost of greater latency. If a
421 value of zero is specified, a system-specific median value is
422 chosen. If the RTAUDIO_MINIMIZE_LATENCY flag bit is set, the
423 lowest allowable value is used. The actual value used is
424 returned via the structure argument. The parameter is API dependent.
426 void openStream( RtAudio::StreamParameters *outputParameters,
427 RtAudio::StreamParameters *inputParameters,
428 RtAudioFormat format, unsigned int sampleRate,
429 unsigned int *bufferFrames, RtAudioCallback callback,
430 void *userData = NULL, RtAudio::StreamOptions *options = NULL );
432 //! A function that closes a stream and frees any associated stream memory.
434 If a stream is not open, this function issues a warning and
435 returns (no exception is thrown).
437 void closeStream( void ) throw();
439 //! A function that starts a stream.
441 An RtError (type = SYSTEM_ERROR) is thrown if an error occurs
442 during processing. An RtError (type = INVALID_USE) is thrown if a
443 stream is not open. A warning is issued if the stream is already
446 void startStream( void );
448 //! Stop a stream, allowing any samples remaining in the output queue to be played.
450 An RtError (type = SYSTEM_ERROR) is thrown if an error occurs
451 during processing. An RtError (type = INVALID_USE) is thrown if a
452 stream is not open. A warning is issued if the stream is already
455 void stopStream( void );
457 //! Stop a stream, discarding any samples remaining in the input/output queue.
459 An RtError (type = SYSTEM_ERROR) is thrown if an error occurs
460 during processing. An RtError (type = INVALID_USE) is thrown if a
461 stream is not open. A warning is issued if the stream is already
464 void abortStream( void );
466 //! Returns true if a stream is open and false if not.
467 bool isStreamOpen( void ) const throw();
469 //! Returns true if the stream is running and false if it is stopped or not open.
470 bool isStreamRunning( void ) const throw();
472 //! Returns the number of elapsed seconds since the stream was started.
474 If a stream is not open, an RtError (type = INVALID_USE) will be thrown.
476 double getStreamTime( void );
478 //! Returns the internal stream latency in sample frames.
480 The stream latency refers to delay in audio input and/or output
481 caused by internal buffering by the audio system and/or hardware.
482 For duplex streams, the returned value will represent the sum of
483 the input and output latencies. If a stream is not open, an
484 RtError (type = INVALID_USE) will be thrown. If the API does not
485 report latency, the return value will be zero.
487 long getStreamLatency( void );
489 //! Returns actual sample rate in use by the stream.
491 On some systems, the sample rate used may be slightly different
492 than that specified in the stream parameters. If a stream is not
493 open, an RtError (type = INVALID_USE) will be thrown.
495 unsigned int getStreamSampleRate( void );
497 //! Specify whether warning messages should be printed to stderr.
498 void showWarnings( bool value = true ) throw();
502 void openRtApi( RtAudio::Api api );
506 // Operating system dependent thread functionality.
507 #if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__)
511 typedef unsigned long ThreadHandle;
512 typedef CRITICAL_SECTION StreamMutex;
514 #elif defined(__LINUX_ALSA__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__)
515 // Using pthread library for various flavors of unix.
518 typedef pthread_t ThreadHandle;
519 typedef pthread_mutex_t StreamMutex;
521 #else // Setup for "dummy" behavior
523 #define __RTAUDIO_DUMMY__
524 typedef int ThreadHandle;
525 typedef int StreamMutex;
529 // This global structure type is used to pass callback information
530 // between the private RtAudio stream structure and global callback
531 // handling functions.
532 struct CallbackInfo {
533 void *object; // Used as a "this" pointer.
537 void *apiInfo; // void pointer for API specific callback information
540 // Default constructor.
542 :object(0), callback(0), userData(0), apiInfo(0), isRunning(false) {}
545 // **************************************************************** //
547 // RtApi class declaration.
549 // Subclasses of RtApi contain all API- and OS-specific code necessary
550 // to fully implement the RtAudio API.
552 // Note that RtApi is an abstract base class and cannot be
553 // explicitly instantiated. The class RtAudio will create an
554 // instance of an RtApi subclass (RtApiOss, RtApiAlsa,
555 // RtApiJack, RtApiCore, RtApiAl, RtApiDs, or RtApiAsio).
557 // **************************************************************** //
559 #if defined( HAVE_GETTIMEOFDAY )
560 #include <sys/time.h>
571 virtual RtAudio::Api getCurrentApi( void ) = 0;
572 virtual unsigned int getDeviceCount( void ) = 0;
573 virtual RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) = 0;
574 virtual unsigned int getDefaultInputDevice( void );
575 virtual unsigned int getDefaultOutputDevice( void );
576 void openStream( RtAudio::StreamParameters *outputParameters,
577 RtAudio::StreamParameters *inputParameters,
578 RtAudioFormat format, unsigned int sampleRate,
579 unsigned int *bufferFrames, RtAudioCallback callback,
580 void *userData, RtAudio::StreamOptions *options );
581 virtual void closeStream( void );
582 virtual void startStream( void ) = 0;
583 virtual void stopStream( void ) = 0;
584 virtual void abortStream( void ) = 0;
585 long getStreamLatency( void );
586 unsigned int getStreamSampleRate( void );
587 virtual double getStreamTime( void );
588 bool isStreamOpen( void ) const { return stream_.state != STREAM_CLOSED; };
589 bool isStreamRunning( void ) const { return stream_.state == STREAM_RUNNING; };
590 void showWarnings( bool value ) { showWarnings_ = value; };
595 static const unsigned int MAX_SAMPLE_RATES;
596 static const unsigned int SAMPLE_RATES[];
598 enum { FAILURE, SUCCESS };
614 // A protected structure used for buffer conversion.
618 RtAudioFormat inFormat, outFormat;
619 std::vector<int> inOffset;
620 std::vector<int> outOffset;
623 // A protected structure for audio streams.
625 unsigned int device[2]; // Playback and record, respectively.
626 void *apiHandle; // void pointer for API specific stream handle information
627 StreamMode mode; // OUTPUT, INPUT, or DUPLEX.
628 StreamState state; // STOPPED, RUNNING, or CLOSED
629 char *userBuffer[2]; // Playback and record, respectively.
631 bool doConvertBuffer[2]; // Playback and record, respectively.
632 bool userInterleaved;
633 bool deviceInterleaved[2]; // Playback and record, respectively.
634 bool doByteSwap[2]; // Playback and record, respectively.
635 unsigned int sampleRate;
636 unsigned int bufferSize;
637 unsigned int nBuffers;
638 unsigned int nUserChannels[2]; // Playback and record, respectively.
639 unsigned int nDeviceChannels[2]; // Playback and record channels, respectively.
640 unsigned int channelOffset[2]; // Playback and record, respectively.
641 unsigned long latency[2]; // Playback and record, respectively.
642 RtAudioFormat userFormat;
643 RtAudioFormat deviceFormat[2]; // Playback and record, respectively.
645 CallbackInfo callbackInfo;
646 ConvertInfo convertInfo[2];
647 double streamTime; // Number of elapsed seconds since the stream started.
649 #if defined(HAVE_GETTIMEOFDAY)
650 struct timeval lastTickTimestamp;
654 :apiHandle(0), deviceBuffer(0) { device[0] = 11111; device[1] = 11111; }
657 typedef signed short Int16;
658 typedef signed int Int32;
659 typedef float Float32;
660 typedef double Float64;
662 std::ostringstream errorStream_;
663 std::string errorText_;
668 Protected, api-specific method that attempts to open a device
669 with the given parameters. This function MUST be implemented by
670 all subclasses. If an error is encountered during the probe, a
671 "warning" message is reported and FAILURE is returned. A
672 successful probe is indicated by a return value of SUCCESS.
674 virtual bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
675 unsigned int firstChannel, unsigned int sampleRate,
676 RtAudioFormat format, unsigned int *bufferSize,
677 RtAudio::StreamOptions *options );
679 //! A protected function used to increment the stream time.
680 void tickStreamTime( void );
682 //! Protected common method to clear an RtApiStream structure.
683 void clearStreamInfo();
686 Protected common method that throws an RtError (type =
687 INVALID_USE) if a stream is not open.
689 void verifyStream( void );
691 //! Protected common error method to allow global control over error handling.
692 void error( RtError::Type type );
695 Protected method used to perform format, channel number, and/or interleaving
696 conversions between the user and device buffers.
698 void convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info );
700 //! Protected common method used to perform byte-swapping on buffers.
701 void byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format );
703 //! Protected common method that returns the number of bytes for a given format.
704 unsigned int formatBytes( RtAudioFormat format );
706 //! Protected common method that sets up the parameters for buffer conversion.
707 void setConvertInfo( StreamMode mode, unsigned int firstChannel );
710 // **************************************************************** //
712 // Inline RtAudio definitions.
714 // **************************************************************** //
716 inline RtAudio::Api RtAudio :: getCurrentApi( void ) throw() { return rtapi_->getCurrentApi(); }
717 inline unsigned int RtAudio :: getDeviceCount( void ) throw() { return rtapi_->getDeviceCount(); }
718 inline RtAudio::DeviceInfo RtAudio :: getDeviceInfo( unsigned int device ) { return rtapi_->getDeviceInfo( device ); }
719 inline unsigned int RtAudio :: getDefaultInputDevice( void ) throw() { return rtapi_->getDefaultInputDevice(); }
720 inline unsigned int RtAudio :: getDefaultOutputDevice( void ) throw() { return rtapi_->getDefaultOutputDevice(); }
721 inline void RtAudio :: closeStream( void ) throw() { return rtapi_->closeStream(); }
722 inline void RtAudio :: startStream( void ) { return rtapi_->startStream(); }
723 inline void RtAudio :: stopStream( void ) { return rtapi_->stopStream(); }
724 inline void RtAudio :: abortStream( void ) { return rtapi_->abortStream(); }
725 inline bool RtAudio :: isStreamOpen( void ) const throw() { return rtapi_->isStreamOpen(); }
726 inline bool RtAudio :: isStreamRunning( void ) const throw() { return rtapi_->isStreamRunning(); }
727 inline long RtAudio :: getStreamLatency( void ) { return rtapi_->getStreamLatency(); }
728 inline unsigned int RtAudio :: getStreamSampleRate( void ) { return rtapi_->getStreamSampleRate(); };
729 inline double RtAudio :: getStreamTime( void ) { return rtapi_->getStreamTime(); }
730 inline void RtAudio :: showWarnings( bool value ) throw() { rtapi_->showWarnings( value ); }
732 // RtApi Subclass prototypes.
734 #if defined(__MACOSX_CORE__)
736 #include <CoreAudio/AudioHardware.h>
738 class RtApiCore: public RtApi
744 RtAudio::Api getCurrentApi( void ) { return RtAudio::MACOSX_CORE; };
745 unsigned int getDeviceCount( void );
746 RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
747 unsigned int getDefaultOutputDevice( void );
748 unsigned int getDefaultInputDevice( void );
749 void closeStream( void );
750 void startStream( void );
751 void stopStream( void );
752 void abortStream( void );
753 long getStreamLatency( void );
755 // This function is intended for internal use only. It must be
756 // public because it is called by the internal callback handler,
757 // which is not a member of RtAudio. External use of this function
758 // will most likely produce highly undesireable results!
759 bool callbackEvent( AudioDeviceID deviceId,
760 const AudioBufferList *inBufferList,
761 const AudioBufferList *outBufferList );
765 bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
766 unsigned int firstChannel, unsigned int sampleRate,
767 RtAudioFormat format, unsigned int *bufferSize,
768 RtAudio::StreamOptions *options );
769 static const char* getErrorCode( OSStatus code );
774 #if defined(__UNIX_JACK__)
776 class RtApiJack: public RtApi
782 RtAudio::Api getCurrentApi( void ) { return RtAudio::UNIX_JACK; };
783 unsigned int getDeviceCount( void );
784 RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
785 void closeStream( void );
786 void startStream( void );
787 void stopStream( void );
788 void abortStream( void );
789 long getStreamLatency( void );
791 // This function is intended for internal use only. It must be
792 // public because it is called by the internal callback handler,
793 // which is not a member of RtAudio. External use of this function
794 // will most likely produce highly undesireable results!
795 bool callbackEvent( unsigned long nframes );
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 );
807 #if defined(__WINDOWS_ASIO__)
809 class RtApiAsio: public RtApi
815 RtAudio::Api getCurrentApi( void ) { return RtAudio::WINDOWS_ASIO; };
816 unsigned int getDeviceCount( void );
817 RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
818 void closeStream( void );
819 void startStream( void );
820 void stopStream( void );
821 void abortStream( void );
822 long getStreamLatency( void );
824 // This function is intended for internal use only. It must be
825 // public because it is called by the internal callback handler,
826 // which is not a member of RtAudio. External use of this function
827 // will most likely produce highly undesireable results!
828 bool callbackEvent( long bufferIndex );
832 std::vector<RtAudio::DeviceInfo> devices_;
833 void saveDeviceInfo( void );
835 bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
836 unsigned int firstChannel, unsigned int sampleRate,
837 RtAudioFormat format, unsigned int *bufferSize,
838 RtAudio::StreamOptions *options );
843 #if defined(__WINDOWS_DS__)
845 class RtApiDs: public RtApi
851 RtAudio::Api getCurrentApi( void ) { return RtAudio::WINDOWS_DS; };
852 unsigned int getDeviceCount( void );
853 unsigned int getDefaultOutputDevice( void );
854 unsigned int getDefaultInputDevice( void );
855 RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
856 void closeStream( void );
857 void startStream( void );
858 void stopStream( void );
859 void abortStream( void );
860 long getStreamLatency( void );
862 // This function is intended for internal use only. It must be
863 // public because it is called by the internal callback handler,
864 // which is not a member of RtAudio. External use of this function
865 // will most likely produce highly undesireable results!
866 void callbackEvent( void );
872 long duplexPrerollBytes;
873 bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
874 unsigned int firstChannel, unsigned int sampleRate,
875 RtAudioFormat format, unsigned int *bufferSize,
876 RtAudio::StreamOptions *options );
881 #if defined(__LINUX_ALSA__)
883 class RtApiAlsa: public RtApi
889 RtAudio::Api getCurrentApi() { return RtAudio::LINUX_ALSA; };
890 unsigned int getDeviceCount( void );
891 RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
892 void closeStream( void );
893 void startStream( void );
894 void stopStream( void );
895 void abortStream( void );
897 // This function is intended for internal use only. It must be
898 // public because it is called by the internal callback handler,
899 // which is not a member of RtAudio. External use of this function
900 // will most likely produce highly undesireable results!
901 void callbackEvent( void );
905 std::vector<RtAudio::DeviceInfo> devices_;
906 void saveDeviceInfo( void );
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_OSS__)
917 class RtApiOss: public RtApi
923 RtAudio::Api getCurrentApi() { return RtAudio::LINUX_OSS; };
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 bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
940 unsigned int firstChannel, unsigned int sampleRate,
941 RtAudioFormat format, unsigned int *bufferSize,
942 RtAudio::StreamOptions *options );
947 #if defined(__RTAUDIO_DUMMY__)
949 class RtApiDummy: public RtApi
953 RtApiDummy() { errorText_ = "RtApiDummy: This class provides no functionality."; error( RtError::WARNING ); };
954 RtAudio::Api getCurrentApi( void ) { return RtAudio::RTAUDIO_DUMMY; };
955 unsigned int getDeviceCount( void ) { return 0; };
956 RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) { RtAudio::DeviceInfo info; return info; };
957 void closeStream( void ) {};
958 void startStream( void ) {};
959 void stopStream( void ) {};
960 void abortStream( 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 ) { return false; };
974 // Indentation settings for Vim and Emacs
978 // indent-tabs-mode: nil
981 // vim: et sts=2 sw=2