Browse Source

Version 3.0

tags/3.0
Gary Scavone Stephen Sinclair 11 years ago
parent
commit
8cd04dd6b7
65 changed files with 7159 additions and 5720 deletions
  1. +4573
    -3455
      RtAudio.cpp
  2. +507
    -317
      RtAudio.h
  3. +60
    -0
      RtError.h
  4. +0
    -0
      asio/asio.h
  5. +0
    -0
      asio/asiodrivers.h
  6. +0
    -0
      asio/asiolist.h
  7. +0
    -0
      asio/asiosys.h
  8. +0
    -0
      asio/ginclude.h
  9. +1
    -1
      config.guess
  10. +1
    -1
      config.sub
  11. +34
    -7
      configure.ac
  12. +1
    -1
      doc/doxygen/Doxyfile
  13. +4
    -4
      doc/doxygen/footer.html
  14. +1
    -1
      doc/doxygen/header.html
  15. +168
    -149
      doc/doxygen/tutorial.txt
  16. +0
    -0
      doc/images/ccrma.gif
  17. BIN
      doc/images/mcgill.gif
  18. +10
    -2
      doc/release.txt
  19. +6
    -4
      install
  20. +0
    -0
      install.sh
  21. +13
    -12
      readme
  22. +0
    -110
      tests/DirectSound/play_saw.dsp
  23. +0
    -137
      tests/DirectSound/rtaudio.dsw
  24. +2
    -8
      tests/Makefile.in
  25. +0
    -0
      tests/Windows/Debug/.placeholder
  26. +0
    -0
      tests/Windows/Release/.placeholder
  27. +0
    -0
      tests/Windows/asio.cpp
  28. +955
    -0
      tests/Windows/asio.h
  29. +0
    -0
      tests/Windows/asiodrivers.cpp
  30. +41
    -0
      tests/Windows/asiodrivers.h
  31. +0
    -0
      tests/Windows/asiodrvr.h
  32. +0
    -0
      tests/Windows/asiolist.cpp
  33. +46
    -0
      tests/Windows/asiolist.h
  34. +82
    -0
      tests/Windows/asiosys.h
  35. +45
    -5
      tests/Windows/call_inout.dsp
  36. +0
    -0
      tests/Windows/call_playtwo.dsp
  37. +45
    -5
      tests/Windows/call_saw.dsp
  38. +0
    -0
      tests/Windows/call_twostreams.dsp
  39. +38
    -0
      tests/Windows/ginclude.h
  40. +0
    -0
      tests/Windows/iasiodrv.h
  41. +45
    -5
      tests/Windows/in_out.dsp
  42. +45
    -5
      tests/Windows/info.dsp
  43. +45
    -5
      tests/Windows/play_raw.dsp
  44. +6
    -4
      tests/Windows/play_saw.dsp
  45. +45
    -5
      tests/Windows/record_raw.dsp
  46. +19
    -7
      tests/Windows/rtaudio.dsw
  47. +46
    -6
      tests/Windows/twostreams.dsp
  48. +0
    -0
      tests/asio/Debug/.placeholder
  49. +0
    -0
      tests/asio/Release/.placeholder
  50. +0
    -148
      tests/asio/call_inout.dsp
  51. +0
    -148
      tests/asio/call_saw.dsp
  52. +0
    -148
      tests/asio/in_out.dsp
  53. +0
    -148
      tests/asio/info.dsp
  54. +0
    -148
      tests/asio/play_raw.dsp
  55. +0
    -148
      tests/asio/record_raw.dsp
  56. +26
    -23
      tests/call_inout.cpp
  57. +0
    -178
      tests/call_playtwo.cpp
  58. +25
    -22
      tests/call_saw.cpp
  59. +0
    -162
      tests/call_twostreams.cpp
  60. +27
    -23
      tests/in_out.cpp
  61. +41
    -32
      tests/info.cpp
  62. +28
    -24
      tests/play_raw.cpp
  63. +31
    -29
      tests/play_saw.cpp
  64. +27
    -23
      tests/record_raw.cpp
  65. +70
    -60
      tests/twostreams.cpp

+ 4573
- 3455
RtAudio.cpp
File diff suppressed because it is too large
View File


+ 507
- 317
RtAudio.h View File

@@ -1,16 +1,16 @@
/************************************************************************/
/*! \class RtAudio
\brief Realtime audio i/o C++ class.
\brief Realtime audio i/o C++ classes.

RtAudio provides a common API (Application Programming Interface)
for realtime audio input/output across Linux (native ALSA and
OSS), SGI, Macintosh OS X (CoreAudio), and Windows (DirectSound
and ASIO) operating systems.
for realtime audio input/output across Linux (native ALSA, Jack,
and OSS), SGI, Macintosh OS X (CoreAudio), and Windows
(DirectSound and ASIO) operating systems.

RtAudio WWW site: http://www-ccrma.stanford.edu/~gary/rtaudio/
RtAudio WWW site: http://music.mcgill.ca/~gary/rtaudio/

RtAudio: a realtime audio i/o C++ class
Copyright (c) 2001-2002 Gary P. Scavone
Copyright (c) 2001-2004 Gary P. Scavone

Permission is hereby granted, free of charge, to any person
obtaining a copy of this software and associated documentation files
@@ -37,206 +37,302 @@
*/
/************************************************************************/

#if !defined(__RTAUDIO_H)
// RtAudio: Version 3.0, 11 March 2004

#ifndef __RTAUDIO_H
#define __RTAUDIO_H

#include <map>
#include "RtError.h"
#include <string>
#include <vector>

#if defined(__LINUX_ALSA__)
#include <alsa/asoundlib.h>
#include <pthread.h>
#include <unistd.h>
// Operating system dependent thread functionality.
#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__)
#include <windows.h>
#include <process.h>

typedef snd_pcm_t *AUDIO_HANDLE;
typedef int DEVICE_ID;
typedef pthread_t THREAD_HANDLE;
typedef pthread_mutex_t MUTEX;
typedef unsigned long ThreadHandle;
typedef CRITICAL_SECTION StreamMutex;

#elif defined(__LINUX_OSS__)
#else // Various unix flavors with pthread support.
#include <pthread.h>
#include <unistd.h>

typedef int AUDIO_HANDLE;
typedef int DEVICE_ID;
typedef pthread_t THREAD_HANDLE;
typedef pthread_mutex_t MUTEX;
typedef pthread_t ThreadHandle;
typedef pthread_mutex_t StreamMutex;

#elif defined(__WINDOWS_DS__)
#include <windows.h>
#include <process.h>
#endif

// The following struct is used to hold the extra variables
// specific to the DirectSound implementation.
typedef struct {
void * object;
void * buffer;
UINT bufferPointer;
} AUDIO_HANDLE;
// This global structure type is used to pass callback information
// between the private RtAudio stream structure and global callback
// handling functions.
struct CallbackInfo {
void *object; // Used as a "this" pointer.
ThreadHandle thread;
bool usingCallback;
void *callback;
void *userData;
void *apiInfo; // void pointer for API specific callback information

typedef LPGUID DEVICE_ID;
typedef unsigned long THREAD_HANDLE;
typedef CRITICAL_SECTION MUTEX;
// Default constructor.
CallbackInfo()
:object(0), usingCallback(false), callback(0),
userData(0), apiInfo(0) {}
};

#elif defined(__WINDOWS_ASIO__)
#include <windows.h>
#include <process.h>
// Support for signed integers and floats. Audio data fed to/from
// the tickStream() routine is assumed to ALWAYS be in host
// byte order. The internal routines will automatically take care of
// any necessary byte-swapping between the host format and the
// soundcard. Thus, endian-ness is not a concern in the following
// format definitions.
typedef unsigned long RtAudioFormat;
static const RtAudioFormat RTAUDIO_SINT8 = 0x1; /*!< 8-bit signed integer. */
static const RtAudioFormat RTAUDIO_SINT16 = 0x2; /*!< 16-bit signed integer. */
static const RtAudioFormat RTAUDIO_SINT24 = 0x4; /*!< Upper 3 bytes of 32-bit signed integer. */
static const RtAudioFormat RTAUDIO_SINT32 = 0x8; /*!< 32-bit signed integer. */
static const RtAudioFormat RTAUDIO_FLOAT32 = 0x10; /*!< Normalized between plus/minus 1.0. */
static const RtAudioFormat RTAUDIO_FLOAT64 = 0x20; /*!< Normalized between plus/minus 1.0. */

typedef int (*RtAudioCallback)(char *buffer, int bufferSize, void *userData);

//! The public device information structure for returning queried values.
struct RtAudioDeviceInfo {
std::string name; /*!< Character string device identifier. */
bool probed; /*!< true if the device capabilities were successfully probed. */
int outputChannels; /*!< Maximum output channels supported by device. */
int inputChannels; /*!< Maximum input channels supported by device. */
int duplexChannels; /*!< Maximum simultaneous input/output channels supported by device. */
bool isDefault; /*!< true if this is the default output or input device. */
std::vector<int> sampleRates; /*!< Supported sample rates (queried from list of standard rates). */
RtAudioFormat nativeFormats; /*!< Bit mask of supported data formats. */

// Default constructor.
RtAudioDeviceInfo()
:probed(false), outputChannels(0), inputChannels(0),
duplexChannels(0), isDefault(false), nativeFormats(0) {}
};

typedef int AUDIO_HANDLE;
typedef int DEVICE_ID;
typedef unsigned long THREAD_HANDLE;
typedef CRITICAL_SECTION MUTEX;
// **************************************************************** //
//
// RtApi class declaration.
//
// Note that RtApi is an abstract base class and cannot be
// explicitly instantiated. The class RtAudio will create an
// instance of an RtApi subclass (RtApiOss, RtApiAlsa,
// RtApiJack, RtApiCore, RtApiAl, RtApiDs, or RtApiAsio).
//
// **************************************************************** //

#elif defined(__IRIX_AL__)
#include <dmedia/audio.h>
#include <pthread.h>
#include <unistd.h>
class RtApi
{
public:

typedef ALport AUDIO_HANDLE;
typedef long DEVICE_ID;
typedef pthread_t THREAD_HANDLE;
typedef pthread_mutex_t MUTEX;
RtApi();
virtual ~RtApi();
void openStream( int outputDevice, int outputChannels,
int inputDevice, int inputChannels,
RtAudioFormat format, int sampleRate,
int *bufferSize, int numberOfBuffers );
virtual void setStreamCallback( RtAudioCallback callback, void *userData ) = 0;
virtual void cancelStreamCallback() = 0;
int getDeviceCount(void);
RtAudioDeviceInfo getDeviceInfo( int device );
char * const getStreamBuffer();
virtual void tickStream() = 0;
virtual void closeStream();
virtual void startStream() = 0;
virtual void stopStream() = 0;
virtual void abortStream() = 0;

#elif defined(__MACOSX_CORE__)
protected:

#include <CoreAudio/AudioHardware.h>
#include <pthread.h>
static const unsigned int MAX_SAMPLE_RATES;
static const unsigned int SAMPLE_RATES[];

typedef unsigned int AUDIO_HANDLE;
typedef AudioDeviceID DEVICE_ID;
typedef pthread_t THREAD_HANDLE;
typedef pthread_mutex_t MUTEX;
enum { FAILURE, SUCCESS };

#endif
enum StreamMode {
OUTPUT,
INPUT,
DUPLEX,
UNINITIALIZED = -75
};

enum StreamState {
STREAM_STOPPED,
STREAM_RUNNING
};

/************************************************************************/
/*! \class RtError
\brief Exception handling class for RtAudio.

The RtError class is quite simple but it does allow errors to be
"caught" by RtError::TYPE. Almost all RtAudio methods can "throw"
an RtError, most typically if an invalid stream identifier is
supplied to a method or a driver error occurs. There are a number
of cases within RtAudio where warning messages may be displayed
but an exception is not thrown. There is a private RtAudio method,
error(), which can be modified to globally control how these
messages are handled and reported.
*/
/************************************************************************/
// A protected structure for audio streams.
struct RtApiStream {
int device[2]; // Playback and record, respectively.
void *apiHandle; // void pointer for API specific stream handle information
StreamMode mode; // OUTPUT, INPUT, or DUPLEX.
StreamState state; // STOPPED or RUNNING
char *userBuffer;
char *deviceBuffer;
bool doConvertBuffer[2]; // Playback and record, respectively.
bool deInterleave[2]; // Playback and record, respectively.
bool doByteSwap[2]; // Playback and record, respectively.
int sampleRate;
int bufferSize;
int nBuffers;
int nUserChannels[2]; // Playback and record, respectively.
int nDeviceChannels[2]; // Playback and record channels, respectively.
RtAudioFormat userFormat;
RtAudioFormat deviceFormat[2]; // Playback and record, respectively.
StreamMutex mutex;
CallbackInfo callbackInfo;

RtApiStream()
:apiHandle(0), userBuffer(0), deviceBuffer(0) {}
// :apiHandle(0), mode(UNINITIALIZED), state(STREAM_STOPPED),
// userBuffer(0), deviceBuffer(0) {}
};

class RtError
{
public:
//! Defined RtError types.
enum TYPE {
WARNING,
DEBUG_WARNING,
UNSPECIFIED,
NO_DEVICES_FOUND,
INVALID_DEVICE,
INVALID_STREAM,
MEMORY_ERROR,
INVALID_PARAMETER,
DRIVER_ERROR,
SYSTEM_ERROR,
THREAD_ERROR
// A protected device structure for audio devices.
struct RtApiDevice {
std::string name; /*!< Character string device identifier. */
bool probed; /*!< true if the device capabilities were successfully probed. */
void *apiDeviceId; // void pointer for API specific device information
int maxOutputChannels; /*!< Maximum output channels supported by device. */
int maxInputChannels; /*!< Maximum input channels supported by device. */
int maxDuplexChannels; /*!< Maximum simultaneous input/output channels supported by device. */
int minOutputChannels; /*!< Minimum output channels supported by device. */
int minInputChannels; /*!< Minimum input channels supported by device. */
int minDuplexChannels; /*!< Minimum simultaneous input/output channels supported by device. */
bool hasDuplexSupport; /*!< true if device supports duplex mode. */
bool isDefault; /*!< true if this is the default output or input device. */
std::vector<int> sampleRates; /*!< Supported sample rates. */
RtAudioFormat nativeFormats; /*!< Bit mask of supported data formats. */

// Default constructor.
RtApiDevice()
:probed(false), apiDeviceId(0), maxOutputChannels(0), maxInputChannels(0),
maxDuplexChannels(0), minOutputChannels(0), minInputChannels(0),
minDuplexChannels(0), isDefault(false), nativeFormats(0) {}
};

protected:
char error_message[256];
TYPE type;
typedef signed short Int16;
typedef signed int Int32;
typedef float Float32;
typedef double Float64;

public:
//! The constructor.
RtError(const char *p, TYPE tipe = RtError::UNSPECIFIED);
char message_[256];
int nDevices_;
std::vector<RtApiDevice> devices_;
RtApiStream stream_;

//! The destructor.
virtual ~RtError(void);
/*!
Protected, api-specific method to count and identify the system
audio devices. This function MUST be implemented by all subclasses.
*/
virtual void initialize(void) = 0;

//! Prints "thrown" error message to stdout.
virtual void printMessage(void);
/*!
Protected, api-specific method which attempts to fill an
RtAudioDevice structure for a given device. This function MUST be
implemented by all subclasses. If an error is encountered during
the probe, a "warning" message is reported and the value of
"probed" remains false (no exception is thrown). A successful
probe is indicated by probed = true.
*/
virtual void probeDeviceInfo( RtApiDevice *info );

//! Returns the "thrown" error message TYPE.
virtual const TYPE& getType(void) { return type; }
/*!
Protected, api-specific method which attempts to open a device
with the given parameters. This function MUST be implemented by
all subclasses. If an error is encountered during the probe, a
"warning" message is reported and FAILURE is returned (no
exception is thrown). A successful probe is indicated by a return
value of SUCCESS.
*/
virtual bool probeDeviceOpen( int device, StreamMode mode, int channels,
int sampleRate, RtAudioFormat format,
int *bufferSize, int numberOfBuffers );

//! Returns the "thrown" error message string.
virtual const char *getMessage(void) { return error_message; }
};
/*!
Protected method which returns the index in the devices array to
the default input device.
*/
virtual int getDefaultInputDevice(void);

/*!
Protected method which returns the index in the devices array to
the default output device.
*/
virtual int getDefaultOutputDevice(void);

//! Protected common method to clear an RtApiDevice structure.
void clearDeviceInfo( RtApiDevice *info );

// This public structure type is used to pass callback information
// between the private RtAudio stream structure and global callback
// handling functions.
typedef struct {
void *object; // Used as a "this" pointer.
int streamId;
DEVICE_ID device[2];
THREAD_HANDLE thread;
void *callback;
void *buffers;
unsigned long waitTime;
bool blockTick;
bool stopStream;
bool usingCallback;
void *userData;
} CALLBACK_INFO;
//! Protected common method to clear an RtApiStream structure.
void clearStreamInfo();

//! Protected common error method to allow global control over error handling.
void error( RtError::Type type );

// *************************************************** //
/*!
Protected common method used to check whether a stream is open.
If not, an "invalid identifier" exception is thrown.
*/
void verifyStream();

/*!
Protected method used to perform format, channel number, and/or interleaving
conversions between the user and device buffers.
*/
void convertStreamBuffer( StreamMode mode );

//! Protected common method used to perform byte-swapping on buffers.
void byteSwapBuffer( char *buffer, int samples, RtAudioFormat format );

//! Protected common method which returns the number of bytes for a given format.
int formatBytes( RtAudioFormat format );
};


// **************************************************************** //
//
// RtAudio class declaration.
//
// *************************************************** //
// RtAudio is a "controller" used to select an available audio i/o
// interface. It presents a common API for the user to call but all
// functionality is implemented by the class RtAudioApi and its
// subclasses. RtAudio creates an instance of an RtAudioApi subclass
// based on the user's API choice. If no choice is made, RtAudio
// attempts to make a "logical" API selection.
//
// **************************************************************** //

class RtAudio
{
public:

// Support for signed integers and floats. Audio data fed to/from
// the tickStream() routine is assumed to ALWAYS be in host
// byte order. The internal routines will automatically take care of
// any necessary byte-swapping between the host format and the
// soundcard. Thus, endian-ness is not a concern in the following
// format definitions.
typedef unsigned long RTAUDIO_FORMAT;
static const RTAUDIO_FORMAT RTAUDIO_SINT8; /*!< 8-bit signed integer. */
static const RTAUDIO_FORMAT RTAUDIO_SINT16; /*!< 16-bit signed integer. */
static const RTAUDIO_FORMAT RTAUDIO_SINT24; /*!< Upper 3 bytes of 32-bit signed integer. */
static const RTAUDIO_FORMAT RTAUDIO_SINT32; /*!< 32-bit signed integer. */
static const RTAUDIO_FORMAT RTAUDIO_FLOAT32; /*!< Normalized between plus/minus 1.0. */
static const RTAUDIO_FORMAT RTAUDIO_FLOAT64; /*!< Normalized between plus/minus 1.0. */

//static const int MAX_SAMPLE_RATES = 14;
enum { MAX_SAMPLE_RATES = 14 };

typedef int (*RTAUDIO_CALLBACK)(char *buffer, int bufferSize, void *userData);

//! The public device information structure for passing queried values.
typedef struct {
char name[128]; /*!< Character string device identifier. */
DEVICE_ID id[2]; /* No value reported by getDeviceInfo(). */
bool probed; /*!< true if the device capabilities were successfully probed. */
int maxOutputChannels; /*!< Maximum output channels supported by device. */
int maxInputChannels; /*!< Maximum input channels supported by device. */
int maxDuplexChannels; /*!< Maximum simultaneous input/output channels supported by device. */
int minOutputChannels; /*!< Minimum output channels supported by device. */
int minInputChannels; /*!< Minimum input channels supported by device. */
int minDuplexChannels; /*!< Minimum simultaneous input/output channels supported by device. */
bool hasDuplexSupport; /*!< true if device supports duplex mode. */
bool isDefault; /*!< true if this is the default output or input device. */
int nSampleRates; /*!< Number of discrete rates or -1 if range supported. */
int sampleRates[MAX_SAMPLE_RATES]; /*!< Supported rates or (min, max) if range. */
RTAUDIO_FORMAT nativeFormats; /*!< Bit mask of supported data formats. */
} RTAUDIO_DEVICE;
//! Audio API specifier arguments.
enum RtAudioApi {
UNSPECIFIED, /*!< Search for a working compiled API. */
LINUX_ALSA, /*!< The Advanced Linux Sound Architecture API. */
LINUX_OSS, /*!< The Linux Open Sound System API. */
LINUX_JACK, /*!< The Linux Jack Low-Latency Audio Server API. */
MACOSX_CORE, /*!< Macintosh OS-X Core Audio API. */
IRIX_AL, /*!< The Irix Audio Library API. */
WINDOWS_ASIO, /*!< The Steinberg Audio Stream I/O API. */
WINDOWS_DS /*!< The Microsoft Direct Sound API. */
};

//! The default constructor.
//! The default class constructor.
/*!
Probes the system to make sure at least one audio input/output
device is available and determines the api-specific identifier for
each device found. An RtError error can be thrown if no devices
are found or if a memory allocation error occurs.

If no API argument is specified and multiple API support has been
compiled, the default order of use is JACK, ALSA, OSS (Linux
systems) and ASIO, DS (Windows systems).
*/
RtAudio();
RtAudio( RtAudioApi api=UNSPECIFIED );

//! A constructor which can be used to open a stream during instantiation.
/*!
@@ -250,23 +346,21 @@ public:
for the given parameters, if a memory allocation error occurs, or
if a driver error occurs. \sa openStream()
*/
RtAudio(int *streamId,
int outputDevice, int outputChannels,
int inputDevice, int inputChannels,
RTAUDIO_FORMAT format, int sampleRate,
int *bufferSize, int numberOfBuffers);
RtAudio( int outputDevice, int outputChannels,
int inputDevice, int inputChannels,
RtAudioFormat format, int sampleRate,
int *bufferSize, int numberOfBuffers, RtAudioApi api=UNSPECIFIED );

//! The destructor.
/*!
Stops and closes any open streams and devices and deallocates
Stops and closes an open stream and devices and deallocates
buffer and structure memory.
*/
~RtAudio();

//! A public method for opening a stream with the specified parameters.
/*!
If successful, the opened stream ID is returned. Otherwise, an
RtError is thrown.
An RtError is thrown if a stream cannot be opened.

\param outputDevice: If equal to 0, the default or first device
found meeting the given parameters is opened. Otherwise, the
@@ -280,7 +374,7 @@ public:
the getDeviceInfo() method.
\param inputChannels: The desired number of input channels. If
equal to zero, the inputDevice identifier is ignored.
\param format: An RTAUDIO_FORMAT specifying the desired sample data format.
\param format: An RtAudioFormat specifying the desired sample data format.
\param sampleRate: The desired sample rate (sample frames per second).
\param *bufferSize: A pointer value indicating the desired internal buffer
size in sample frames. The actual value used by the device is
@@ -291,47 +385,47 @@ public:
though at a cost of greater latency. A value of zero can be
specified, in which case the lowest allowable value is used.
*/
int openStream(int outputDevice, int outputChannels,
int inputDevice, int inputChannels,
RTAUDIO_FORMAT format, int sampleRate,
int *bufferSize, int numberOfBuffers);
void openStream( int outputDevice, int outputChannels,
int inputDevice, int inputChannels,
RtAudioFormat format, int sampleRate,
int *bufferSize, int numberOfBuffers );

//! A public method which sets a user-defined callback function for a given stream.
/*!
This method assigns a callback function to a specific,
previously opened stream for non-blocking stream functionality. A
separate process is initiated, though the user function is called
only when the stream is "running" (between calls to the
startStream() and stopStream() methods, respectively). The
callback process remains active for the duration of the stream and
is automatically shutdown when the stream is closed (via the
closeStream() method or by object destruction). The callback
process can also be shutdown and the user function de-referenced
through an explicit call to the cancelStreamCallback() method.
Note that a single stream can use only blocking or callback
functionality at the same time, though it is possible to alternate
modes on the same stream through the use of the
setStreamCallback() and cancelStreamCallback() methods (the
blocking tickStream() method can be used before a callback is set
and/or after a callback is cancelled). An RtError will be thrown
for an invalid device argument.
This method assigns a callback function to a previously opened
stream for non-blocking stream functionality. A separate process
is initiated, though the user function is called only when the
stream is "running" (between calls to the startStream() and
stopStream() methods, respectively). The callback process remains
active for the duration of the stream and is automatically
shutdown when the stream is closed (via the closeStream() method
or by object destruction). The callback process can also be
shutdown and the user function de-referenced through an explicit
call to the cancelStreamCallback() method. Note that the stream
can use only blocking or callback functionality at a particular
time, though it is possible to alternate modes on the same stream
through the use of the setStreamCallback() and
cancelStreamCallback() methods (the blocking tickStream() method
can be used before a callback is set and/or after a callback is
cancelled). An RtError will be thrown if called when no stream is
open or a thread errors occurs.
*/
void setStreamCallback(int streamId, RTAUDIO_CALLBACK callback, void *userData);
void setStreamCallback(RtAudioCallback callback, void *userData) { rtapi_->setStreamCallback( callback, userData ); };

//! A public method which cancels a callback process and function for a given stream.
//! A public method which cancels a callback process and function for the stream.
/*!
This method shuts down a callback process and de-references the
user function for a specific stream. Callback functionality can
user function for the stream. Callback functionality can
subsequently be restarted on the stream via the
setStreamCallback() method. An RtError will be thrown for an
invalid device argument.
setStreamCallback() method. An RtError will be thrown if called
when no stream is open.
*/
void cancelStreamCallback(int streamId);
void cancelStreamCallback() { rtapi_->cancelStreamCallback(); };

//! A public method which returns the number of audio devices found.
int getDeviceCount(void);
int getDeviceCount(void) { return rtapi_->getDeviceCount(); };

//! Fill a user-supplied RTAUDIO_DEVICE structure for a specified device number.
//! Return an RtAudioDeviceInfo structure for a specified device number.
/*!
Any device integer between 1 and getDeviceCount() is valid. If
a device is busy or otherwise unavailable, the structure member
@@ -340,185 +434,281 @@ public:
or output device, the "isDefault" member will have a value of
"true". An RtError will be thrown for an invalid device argument.
*/
void getDeviceInfo(int device, RTAUDIO_DEVICE *info);
RtAudioDeviceInfo getDeviceInfo(int device) { return rtapi_->getDeviceInfo( device ); };

//! A public method which returns a pointer to the buffer for an open stream.
/*!
The user should fill and/or read the buffer data in interleaved format
and then call the tickStream() method. An RtError will be
thrown for an invalid stream identifier.
thrown if called when no stream is open.
*/
char * const getStreamBuffer(int streamId);
char * const getStreamBuffer() { return rtapi_->getStreamBuffer(); };

//! Public method used to trigger processing of input/output data for a stream.
/*!
This method blocks until all buffer data is read/written. An
RtError will be thrown for an invalid stream identifier or if
a driver error occurs.
RtError will be thrown if a driver error occurs or if called when
no stream is open.
*/
void tickStream(int streamId);
void tickStream() { rtapi_->tickStream(); };

//! Public method which closes a stream and frees any associated buffers.
/*!
If an invalid stream identifier is specified, this method
issues a warning and returns (an RtError is not thrown).
If a stream is not open, this method issues a warning and
returns (an RtError is not thrown).
*/
void closeStream(int streamId);
void closeStream() { rtapi_->closeStream(); };

//! Public method which starts a stream.
/*!
An RtError will be thrown for an invalid stream identifier
or if a driver error occurs.
An RtError will be thrown if a driver error occurs or if called
when no stream is open.
*/
void startStream(int streamId);
void startStream() { rtapi_->startStream(); };

//! Stop a stream, allowing any samples remaining in the queue to be played out and/or read in.
/*!
An RtError will be thrown for an invalid stream identifier
or if a driver error occurs.
An RtError will be thrown if a driver error occurs or if called
when no stream is open.
*/
void stopStream(int streamId);
void stopStream() { rtapi_->stopStream(); };

//! Stop a stream, discarding any samples remaining in the input/output queue.
/*!
An RtError will be thrown for an invalid stream identifier
or if a driver error occurs.
An RtError will be thrown if a driver error occurs or if called
when no stream is open.
*/
void abortStream(int streamId);
void abortStream() { rtapi_->abortStream(); };

//! Queries a stream to determine whether a call to the tickStream() method will block.
/*!
A return value of 0 indicates that the stream will NOT block. A positive
return value indicates the number of sample frames that cannot yet be
processed without blocking.
*/
int streamWillBlock(int streamId);

#if (defined(__MACOSX_CORE__) || defined(__WINDOWS_ASIO__))
protected:

void initialize( RtAudioApi api );

RtApi *rtapi_;
};


// RtApi Subclass prototypes.

#if defined(__LINUX_ALSA__)

class RtApiAlsa: public RtApi
{
public:

RtApiAlsa();
~RtApiAlsa();
void tickStream();
void closeStream();
void startStream();
void stopStream();
void abortStream();
int streamWillBlock();
void setStreamCallback( RtAudioCallback callback, void *userData );
void cancelStreamCallback();

private:

void initialize(void);
void probeDeviceInfo( RtApiDevice *info );
bool probeDeviceOpen( int device, StreamMode mode, int channels,
int sampleRate, RtAudioFormat format,
int *bufferSize, int numberOfBuffers );
};

#endif

#if defined(__LINUX_JACK__)

class RtApiJack: public RtApi
{
public:

RtApiJack();
~RtApiJack();
void tickStream();
void closeStream();
void startStream();
void stopStream();
void abortStream();
void setStreamCallback( RtAudioCallback callback, void *userData );
void cancelStreamCallback();
// This function is intended for internal use only. It must be
// public because it is called by the internal callback handler,
// which is not a member of RtAudio. External use of this function
// will most likely produce highly undesireable results!
void callbackEvent(int streamId, DEVICE_ID deviceId, void *inData, void *outData);
void callbackEvent( unsigned long nframes );

private:

void initialize(void);
void probeDeviceInfo( RtApiDevice *info );
bool probeDeviceOpen( int device, StreamMode mode, int channels,
int sampleRate, RtAudioFormat format,
int *bufferSize, int numberOfBuffers );
};

#endif

protected:
#if defined(__LINUX_OSS__)

private:
class RtApiOss: public RtApi
{
public:

static const unsigned int SAMPLE_RATES[MAX_SAMPLE_RATES];
RtApiOss();
~RtApiOss();
void tickStream();
void closeStream();
void startStream();
void stopStream();
void abortStream();
int streamWillBlock();
void setStreamCallback( RtAudioCallback callback, void *userData );
void cancelStreamCallback();

enum { FAILURE, SUCCESS };
private:

enum STREAM_MODE {
OUTPUT,
INPUT,
DUPLEX,
UNINITIALIZED = -75
};
void initialize(void);
void probeDeviceInfo( RtApiDevice *info );
bool probeDeviceOpen( int device, StreamMode mode, int channels,
int sampleRate, RtAudioFormat format,
int *bufferSize, int numberOfBuffers );
};

enum STREAM_STATE {
STREAM_STOPPED,
STREAM_RUNNING
};
#endif

typedef struct {
int device[2]; // Playback and record, respectively.
STREAM_MODE mode; // OUTPUT, INPUT, or DUPLEX.
AUDIO_HANDLE handle[2]; // Playback and record handles, respectively.
STREAM_STATE state; // STOPPED or RUNNING
char *userBuffer;
char *deviceBuffer;
bool doConvertBuffer[2]; // Playback and record, respectively.
bool deInterleave[2]; // Playback and record, respectively.
bool doByteSwap[2]; // Playback and record, respectively.
int sampleRate;
int bufferSize;
int nBuffers;
int nUserChannels[2]; // Playback and record, respectively.
int nDeviceChannels[2]; // Playback and record channels, respectively.
RTAUDIO_FORMAT userFormat;
RTAUDIO_FORMAT deviceFormat[2]; // Playback and record, respectively.
MUTEX mutex;
CALLBACK_INFO callbackInfo;
} RTAUDIO_STREAM;
#if defined(__MACOSX_CORE__)

typedef signed short INT16;
typedef signed int INT32;
typedef float FLOAT32;
typedef double FLOAT64;
#include <CoreAudio/AudioHardware.h>

char message[256];
int nDevices;
RTAUDIO_DEVICE *devices;
class RtApiCore: public RtApi
{
public:

std::map<int, void *> streams;
RtApiCore();
~RtApiCore();
int getDefaultOutputDevice(void);
int getDefaultInputDevice(void);
void tickStream();
void closeStream();
void startStream();
void stopStream();
void abortStream();
void setStreamCallback( RtAudioCallback callback, void *userData );
void cancelStreamCallback();

// This function is intended for internal use only. It must be
// public because it is called by the internal callback handler,
// which is not a member of RtAudio. External use of this function
// will most likely produce highly undesireable results!
void callbackEvent( AudioDeviceID deviceId, void *inData, void *outData );

//! Private error method to allow global control over error handling.
void error(RtError::TYPE type);
private:

/*!
Private method to count the system audio devices, allocate the
RTAUDIO_DEVICE structures, and probe the device capabilities.
*/
void initialize(void);
void probeDeviceInfo( RtApiDevice *info );
bool probeDeviceOpen( int device, StreamMode mode, int channels,
int sampleRate, RtAudioFormat format,
int *bufferSize, int numberOfBuffers );
};

/*!
Private method which returns the index in the devices array to
the default input device.
*/
int getDefaultInputDevice(void);
#endif

/*!
Private method which returns the index in the devices array to
the default output device.
*/
#if defined(__WINDOWS_DS__)

class RtApiDs: public RtApi
{
public:

RtApiDs();
~RtApiDs();
int getDefaultOutputDevice(void);
int getDefaultInputDevice(void);
void tickStream();
void closeStream();
void startStream();
void stopStream();
void abortStream();
int streamWillBlock();
void setStreamCallback( RtAudioCallback callback, void *userData );
void cancelStreamCallback();

//! Private method to clear an RTAUDIO_DEVICE structure.
void clearDeviceInfo(RTAUDIO_DEVICE *info);
private:

/*!
Private method which attempts to fill an RTAUDIO_DEVICE
structure for a given device. If an error is encountered during
the probe, a "warning" message is reported and the value of
"probed" remains false (no exception is thrown). A successful
probe is indicated by probed = true.
*/
void probeDeviceInfo(RTAUDIO_DEVICE *info);
void initialize(void);
void probeDeviceInfo( RtApiDevice *info );
bool probeDeviceOpen( int device, StreamMode mode, int channels,
int sampleRate, RtAudioFormat format,
int *bufferSize, int numberOfBuffers );
};

/*!
Private method which attempts to open a device with the given parameters.
If an error is encountered during the probe, a "warning" message is
reported and FAILURE is returned (no exception is thrown). A
successful probe is indicated by a return value of SUCCESS.
*/
bool probeDeviceOpen(int device, RTAUDIO_STREAM *stream,
STREAM_MODE mode, int channels,
int sampleRate, RTAUDIO_FORMAT format,
int *bufferSize, int numberOfBuffers);
#endif

/*!
Private common method used to check validity of a user-passed
stream ID. When the ID is valid, this method returns a pointer to
an RTAUDIO_STREAM structure (in the form of a void pointer).
Otherwise, an "invalid identifier" exception is thrown.
*/
void *verifyStream(int streamId);
#if defined(__WINDOWS_ASIO__)

/*!
Private method used to perform format, channel number, and/or interleaving
conversions between the user and device buffers.
*/
void convertStreamBuffer(RTAUDIO_STREAM *stream, STREAM_MODE mode);
class RtApiAsio: public RtApi
{
public:

RtApiAsio();
~RtApiAsio();
void tickStream();
void closeStream();
void startStream();
void stopStream();
void abortStream();
void setStreamCallback( RtAudioCallback callback, void *userData );
void cancelStreamCallback();

// This function is intended for internal use only. It must be
// public because it is called by the internal callback handler,
// which is not a member of RtAudio. External use of this function
// will most likely produce highly undesireable results!
void callbackEvent( long bufferIndex );

private:

void initialize(void);
void probeDeviceInfo( RtApiDevice *info );
bool probeDeviceOpen( int device, StreamMode mode, int channels,
int sampleRate, RtAudioFormat format,
int *bufferSize, int numberOfBuffers );
};

#endif

#if defined(__IRIX_AL__)

class RtApiAl: public RtApi
{
public:

RtApiAl();
~RtApiAl();
int getDefaultOutputDevice(void);
int getDefaultInputDevice(void);
void tickStream();
void closeStream();
void startStream();
void stopStream();
void abortStream();
int streamWillBlock();
void setStreamCallback( RtAudioCallback callback, void *userData );
void cancelStreamCallback();

//! Private method used to perform byte-swapping on buffers.
void byteSwapBuffer(char *buffer, int samples, RTAUDIO_FORMAT format);
private:

//! Private method which returns the number of bytes for a given format.
int formatBytes(RTAUDIO_FORMAT format);
void initialize(void);
void probeDeviceInfo( RtApiDevice *info );
bool probeDeviceOpen( int device, StreamMode mode, int channels,
int sampleRate, RtAudioFormat format,
int *bufferSize, int numberOfBuffers );
};

#endif

// Define the following flag to have extra information spewed to stderr.
//#define __RTAUDIO_DEBUG__



+ 60
- 0
RtError.h View File

@@ -0,0 +1,60 @@
/************************************************************************/
/*! \class RtError
\brief Exception handling class for RtAudio & RtMidi.

The RtError class is quite simple but it does allow errors to be
"caught" by RtError::Type. See the RtAudio and RtMidi
documentation to know which methods can throw an RtError.

*/
/************************************************************************/

#ifndef RTERROR_H
#define RTERROR_H

#include <iostream>
#include <string>

class RtError
{
public:
//! Defined RtError types.
enum Type {
WARNING, /*!< A non-critical error. */
DEBUG_WARNING, /*!< A non-critical error which might be useful for debugging. */
UNSPECIFIED, /*!< The default, unspecified error type. */
NO_DEVICES_FOUND, /*!< No devices found on system. */
INVALID_DEVICE, /*!< An invalid device ID was specified. */
INVALID_STREAM, /*!< An invalid stream ID was specified. */
MEMORY_ERROR, /*!< An error occured during memory allocation. */
INVALID_PARAMETER, /*!< An invalid parameter was specified to a function. */
DRIVER_ERROR, /*!< A system driver error occured. */
SYSTEM_ERROR, /*!< A system error occured. */
THREAD_ERROR /*!< A thread error occured. */
};

protected:
std::string message_;
Type type_;

public:
//! The constructor.
RtError(const std::string& message, Type type = RtError::UNSPECIFIED) : message_(message), type_(type){}

//! The destructor.
virtual ~RtError(void) {};

//! Prints thrown error message to stderr.
virtual void printMessage(void) { std::cerr << '\n' << message_ << "\n\n"; }

//! Returns the thrown error message type.
virtual const Type& getType(void) { return type_; }

//! Returns the thrown error message string.
virtual const std::string& getMessage(void) { return message_; }

//! Returns the thrown error message as a C string.
virtual const char *getMessageString(void) { return message_.c_str(); }
};

#endif

tests/asio/asio.h → asio/asio.h View File


tests/asio/asiodrivers.h → asio/asiodrivers.h View File


tests/asio/asiolist.h → asio/asiolist.h View File


tests/asio/asiosys.h → asio/asiosys.h View File


tests/asio/ginclude.h → asio/ginclude.h View File


+ 1
- 1
config.guess View File

@@ -3,7 +3,7 @@
# Copyright (C) 1992, 1993, 1994, 1995, 1996, 1997, 1998, 1999, 2000, 2001
# Free Software Foundation, Inc.

timestamp='2001-04-20'
timestamp='2004-02-26'

# This file is free software; you can redistribute it and/or modify it
# under the terms of the GNU General Public License as published by


+ 1
- 1
config.sub View File

@@ -3,7 +3,7 @@
# Copyright (C) 1992, 1993, 1994, 1995, 1996, 1997, 1998, 1999, 2000, 2001
# Free Software Foundation, Inc.

timestamp='2001-04-20'
timestamp='2004-02-26'

# This file is (in principle) common to ALL GNU software.
# The presence of a machine in this file suggests that SOME GNU software


+ 34
- 7
configure.ac View File

@@ -1,11 +1,14 @@
# Process this file with autoconf to produce a configure script.
AC_INIT(RtAudio, 2.1, gary@ccrma.stanford.edu, rtaudio)
AC_INIT(RtAudio, 3.0, gary@ccrma.stanford.edu, rtaudio)
AC_CONFIG_SRCDIR(RtAudio.cpp)
AC_CONFIG_FILES(tests/Makefile)

# Fill GXX with something before test.
AC_SUBST( GXX, ["no"] )

# Checks for programs.
AC_PROG_CC
AC_PROG_CXX(CC g++ c++ cxx)
AC_PROG_CXX(g++ CC c++ cxx)

# Checks for libraries.
AC_CHECK_LIB(pthread, pthread_create, , AC_MSG_ERROR(RtAudio requires the pthread library!))
@@ -25,7 +28,7 @@ AC_ARG_ENABLE(debug,
[AC_SUBST( debug, [] ) AC_SUBST( cflags, [-O2] ) AC_SUBST( object_path, [Release] ) AC_MSG_RESULT(no)])

# Check compiler and use -Wall if gnu.
if test $GXX = "yes" ; then
if [test $GXX = "yes" ;] then
AC_SUBST( warn, [-Wall] )
fi

@@ -34,15 +37,39 @@ AC_CANONICAL_HOST
AC_MSG_CHECKING(for audio API)
case $host in
*-*-linux*)
AC_ARG_WITH(alsa, [ --with-alsa = choose native ALSA API support (linux only)], [AC_SUBST( sound_api, [-D__LINUX_ALSA__] ) AC_MSG_RESULT(using ALSA) ], [AC_SUBST( sound_api, [-D__LINUX_OSS__] ) AC_MSG_RESULT(using OSS)])
AC_SUBST( sound_api, [_NO_API_] )
AC_ARG_WITH(jack, [ --with-jack = choose JACK server support (linux only)], [AC_SUBST( sound_api, [-D__LINUX_JACK__] ) AC_MSG_RESULT(using JACK)], )
if [test $sound_api = -D__LINUX_JACK__;] then
TEMP_LIBS=$LIBS
AC_CHECK_LIB(jack, jack_client_new, , AC_MSG_ERROR(JACK support requires the jack library!))
AC_CHECK_LIB(asound, snd_pcm_open, , AC_MSG_ERROR(Jack support also requires the asound library!))
LIBS="`pkg-config --cflags --libs jack` $TEMP_LIBS -lasound"
audio_apis="-D__LINUX_JACK__"
fi

if test $sound_api = -D__LINUX_ALSA__; then
# Look for Alsa flag
AC_ARG_WITH(alsa, [ --with-alsa = choose native ALSA API support (linux only)], [AC_SUBST( sound_api, [-D__LINUX_ALSA__] ) AC_MSG_RESULT(using ALSA)], )
if [test $sound_api = -D__LINUX_ALSA__;] then
AC_CHECK_LIB(asound, snd_pcm_open, , AC_MSG_ERROR(ALSA support requires the asound library!))
audio_apis="-D__LINUX_ALSA__ $audio_apis"
fi

# Look for OSS flag
AC_ARG_WITH(oss, [ --with-oss = choose OSS API support (linux only)], [AC_SUBST( sound_api, [-D__LINUX_OSS__] ) AC_MSG_RESULT(using OSS)], )
if test $sound_api = -D__LINUX_OSS__; then
audio_apis="-D__LINUX_OSS__ $audio_apis"
fi

# If no audio api flags specified, use OSS
if [test $sound_api = _NO_API_;] then
AC_SUBST( sound_api, [-D__LINUX_OSS__] )
AC_MSG_RESULT(using OSS)
AC_SUBST( audio_apis, [-D__LINUX_OSS__] )
fi
;;

*-sgi*)
AC_SUBST( sound_api, [-D__IRIX_AL__] )
AC_SUBST( audio_apis, ["-D__IRIX_AL__ -LANG:std -w"] )
AC_MSG_RESULT(using IRIX AL)
AC_CHECK_LIB(audio, alOpenPort, , AC_MSG_ERROR(IRIX audio support requires the audio library!) )
;;
@@ -50,7 +77,7 @@ case $host in
*-apple*)
# Check for CoreAudio framework
AC_CHECK_HEADER(CoreAudio/CoreAudio.h,
[AC_SUBST( sound_api, [-D__MACOSX_CORE__] )],
[AC_SUBST( audio_apis, [-D__MACOSX_CORE__] )],
[AC_MSG_ERROR(CoreAudio header files not found!)] )
AC_SUBST( frameworks, ["-framework CoreAudio"] )
AC_CHECK_LIB(stdc++, printf, , AC_MSG_ERROR(RtAudio requires the C++ library!) )


+ 1
- 1
doc/doxygen/Doxyfile View File

@@ -51,7 +51,7 @@ WARN_LOGFILE =
#---------------------------------------------------------------------------
# configuration options related to the input files
#---------------------------------------------------------------------------
INPUT = tutorial.txt ../../RtAudio.h
INPUT = tutorial.txt ../../RtAudio.h ../../RtError.h
FILE_PATTERNS =
RECURSIVE = NO
EXCLUDE =


+ 4
- 4
doc/doxygen/footer.html View File

@@ -1,9 +1,9 @@
<HR>

<table><tr><td><img src="../images/ccrma.gif">
<td>&copy;2001-2002 Gary P. Scavone, CCRMA, Stanford University. All Rights Reserved.<br>
Maintained by Gary P. Scavone, <a href="mailto:gary@ccrma.stanford.edu">gary@ccrma.stanford.edu</a><P>
<table><tr><td><img src="../images/mcgill.gif" width=165></td>
<td>&copy;2001-2004 Gary P. Scavone, McGill University. All Rights Reserved.<br>
Maintained by Gary P. Scavone, <a href="mailto:gary@music.mcgill.ca">gary@music.mcgill.ca</a></td></tr>
</table>

</BODY>
</HTML>
</HTML>

+ 1
- 1
doc/doxygen/header.html View File

@@ -6,4 +6,4 @@
<BODY BGCOLOR="#FFFFFF">
<CENTER>
<a class="qindex" href="index.html">Tutorial</a> &nbsp; <a class="qindex" href="annotated.html">Class/Enum List</a> &nbsp; <a class="qindex" href="files.html">File List</a> &nbsp; <a class="qindex" href="functions.html">Compound Members</a> &nbsp; </CENTER>
<HR>
<HR>

+ 168
- 149
doc/doxygen/tutorial.txt View File

@@ -2,51 +2,49 @@

<BODY BGCOLOR="white">

- \ref intro
- \ref download
- \ref start
- \ref error
- \ref probing
- \ref settings
- \ref playbackb
- \ref playbackc
- \ref recording
- \ref duplex
- \ref methods
- \ref compiling
- \ref debug
- \ref osnotes
- \ref acknowledge
- \ref license
<CENTER>\ref intro &nbsp;&nbsp; \ref changes &nbsp;&nbsp;\ref download &nbsp;&nbsp; \ref start &nbsp;&nbsp; \ref error &nbsp;&nbsp; \ref probing &nbsp;&nbsp; \ref settings &nbsp;&nbsp; \ref playbackb &nbsp;&nbsp; \ref playbackc &nbsp;&nbsp; \ref recording &nbsp;&nbsp; \ref duplex &nbsp;&nbsp; \ref multi &nbsp;&nbsp; \ref methods &nbsp;&nbsp; \ref compiling &nbsp;&nbsp; \ref debug &nbsp;&nbsp; \ref apinotes &nbsp;&nbsp; \ref acknowledge &nbsp;&nbsp; \ref license</CENTER>

\section intro Introduction

RtAudio is a C++ class which provides a common API (Application Programming Interface) for realtime audio input/output across Linux (native ALSA and OSS), Macintosh OS X, SGI, and Windows (DirectSound and ASIO) operating systems. RtAudio significantly simplifies the process of interacting with computer audio hardware. It was designed with the following goals:
RtAudio is a set of C++ classes which provide a common API (Application Programming Interface) for realtime audio input/output across Linux (native ALSA, JACK, and OSS), Macintosh OS X, SGI, and Windows (DirectSound and ASIO) operating systems. RtAudio significantly simplifies the process of interacting with computer audio hardware. It was designed with the following goals:

<UL>
<LI>object oriented C++ design</LI>
<LI>simple, common API across all supported platforms</LI>
<LI>single independent header and source file for easy inclusion in programming projects</LI>
<LI>only two header files and one source file for easy inclusion in programming projects</LI>
<LI>allow simultaneous multi-api support</LI>
<LI>blocking functionality</LI>
<LI>callback functionality</LI>
<LI>extensive audio device parameter control</LI>
<LI>audio device capability probing</LI>
<LI>automatic internal conversion for data format, channel number compensation, de-interleaving, and byte-swapping</LI>
<LI>control over multiple audio streams and devices with a single instance</LI>
</UL>

RtAudio incorporates the concept of audio streams, which represent audio output (playback) and/or input (recording). Available audio devices and their capabilities can be enumerated and then specified when opening a stream. When allowed by the underlying audio API, multiple streams can run at the same time and a single device can serve multiple streams. See the \ref osnotes section for information specific to each of the supported audio APIs.
RtAudio incorporates the concept of audio streams, which represent audio output (playback) and/or input (recording). Available audio devices and their capabilities can be enumerated and then specified when opening a stream. Where applicable, multiple API support can be compiled and a particular API specified when creating an RtAudio instance. See the \ref apinotes section for information specific to each of the supported audio APIs.

The RtAudio API provides both blocking (synchronous) and callback (asyncronous) functionality. Callbacks are typically used in conjunction with graphical user interfaces (GUI). Blocking functionality is often necessary for explicit control of multiple input/output stream synchronization or when audio must be synchronized with other system events.
The RtAudio API provides both blocking (synchronous) and callback (asynchronous) functionality. Callbacks are typically used in conjunction with graphical user interfaces (GUI). Blocking functionality is often necessary for explicit control of multiple input/output stream synchronization or when audio must be synchronized with other system events.

\section download Download
\section changes What's New (Version 3.0)

RtAudio now allows simultaneous multi-api support. For example, you can compile RtAudio to provide both DirectSound and ASIO support on Windows platforms or ALSA, JACK, and OSS support on Linux platforms. This was accomplished by creating an abstract base class, RtApi, with subclasses for each supported API (RtApiAlsa, RtApiJack, RtApiOss, RtApiDs, RtApiAsio, RtApiCore, and RtApiAl). The class RtAudio is now a "controller" which creates an instance of an RtApi subclass based on the user's API choice via an optional RtAudio::RtAudioApi instantiation argument. If no API is specified, RtAudio attempts to make a "logical" API selection.

Support for the JACK low-latency audio server has been added with this version of RtAudio. It is necessary to have the JACK server running before creating an instance of RtAudio.

Several API changes have been made in version 3.0 of RtAudio in an effort to provide more consistent behavior across all supported audio APIs. The most significant of these changes is that multiple stream support from a single RtAudio instance has been discontinued. As a result, stream identifier input arguments are no longer required. Also, the RtAudio::streamWillBlock() function was poorly supported by most APIs and has been deprecated (though the function still exists in those subclasses of RtApi that do allow it to be implemented).

The RtAudio::getDeviceInfo() function was modified to return a globally defined RtAudioDeviceInfo structure. This structure is a simplified version of the previous RTAUDIO_DEVICE structure. In addition, the RTAUDIO_FORMAT structure was renamed RtAudioFormat and defined globally within RtAudio.h. These changes were made for clarity and to better conform with standard C++ programming practices.

The RtError class declaration and definition have been extracted to a separate file (RtError.h). This was done in preparation for a new release of the RtMidi class (planned for Summer 2004).

Latest Release (24 October 2002): <A href="http://www-ccrma.stanford.edu/~gary/rtaudio/release/rtaudio-2.1.1.tar.gz">Version 2.1.1 (165 kB tar/gzipped)</A>
\section download Download

Latest Release (11 March 2004): <A href="http://music.mcgill.ca/~gary/rtaudio/release/rtaudio-3.0.tar.gz">Version 3.0 (200 kB tar/gzipped)</A>

\section start Getting Started

The first thing that must be done when using RtAudio is to create an instance of the class. The default constructor RtAudio::RtAudio() scans the underlying audio system to verify that at least one device is available. RtAudio often uses C++ exceptions to report errors, necessitating try/catch blocks around most member functions. The following code example demonstrates default object construction and destruction:
With version 3.0, it is now possible to compile multiple API support on a given platform and to specify an API choice during class instantiation. In the examples that follow, no API will be specified (in which case, RtAudio attempts to select the most "logical" available API).

The first thing that must be done when using RtAudio is to create an instance of the class. The default constructor scans the underlying audio system to verify that at least one device is available. RtAudio often uses C++ exceptions to report errors, necessitating try/catch blocks around most member functions. The following code example demonstrates default object construction and destruction:

\code

@@ -62,6 +60,7 @@ int main()
}
catch (RtError &error) {
// Handle the exception here
error.printMessage();
}

// Clean up
@@ -74,7 +73,7 @@ Obviously, this example doesn't demonstrate any of the real functionality of RtA

\section error Error Handling

RtAudio uses a C++ exception handler called RtError, which is declared and defined within the RtAudio class files. The RtError class is quite simple but it does allow errors to be "caught" by RtError::TYPE. Almost all RtAudio methods can "throw" an RtError, most typically if an invalid stream identifier is supplied to a method or a driver error occurs. There are a number of cases within RtAudio where warning messages may be displayed but an exception is not thrown. There is a private RtAudio method, error(), which can be modified to globally control how these messages are handled and reported.
RtAudio uses a C++ exception handler called RtError, which is declared and defined in RtError.h. The RtError class is quite simple but it does allow errors to be "caught" by RtError::Type. Almost all RtAudio methods can "throw" an RtError, most typically if a driver error occurs or a stream function is called when no stream is open. There are a number of cases within RtAudio where warning messages may be displayed but an exception is not thrown. There is a protected RtAudio method, error(), which can be modified to globally control how these messages are handled and reported. By default, error messages are not automatically displayed in RtAudio unless the preprocessor definition __RTAUDIO_DEBUG__ is defined. Messages associated with caught exceptions can be displayed with, for example, the RtError::printMessage() function.


\section probing Probing Device Capabilities
@@ -105,11 +104,11 @@ int main()
int devices = audio->getDeviceCount();

// Scan through devices for various capabilities
RtAudio::RTAUDIO_DEVICE info;
RtAudioDeviceInfo info;
for (int i=1; i<=devices; i++) {

try {
audio->getDeviceInfo(i, &info);
info = audio->getDeviceInfo(i);
}
catch (RtError &error) {
error.printMessage();
@@ -117,8 +116,8 @@ int main()
}

// Print, for example, the maximum number of output channels for each device
cout << "device = " << i;
cout << ": maximum output channels = " << info.maxOutputChannels << "\n";
std::cout << "device = " << i;
std::cout << ": maximum output channels = " << info.outputChannels << "\n";
}

// Clean up
@@ -128,43 +127,38 @@ int main()
}
\endcode

The RTAUDIO_DEVICE structure is defined in RtAudio.h and provides a variety of information useful in assessing the capabilities of a device:
The RtAudioDeviceInfo structure is defined in RtAudio.h and provides a variety of information useful in assessing the capabilities of a device:

\code
typedef struct {
char name[128];
bool probed; // true if the device probe was successful.
int maxOutputChannels;
int maxInputChannels;
int maxDuplexChannels;
int minOutputChannels;
int minInputChannels;
int minDuplexChannels;
bool hasDuplexSupport; // true if duplex mode is supported.
bool isDefault; // true if this is the default output or input device.
int nSampleRates; // Number of discrete rates, or -1 if range supported.
double sampleRates[MAX_SAMPLE_RATES]; // Supported sample rates, or {min, max} if range.
RTAUDIO_FORMAT nativeFormats;
} RTAUDIO_DEVICE;
typedef struct RtAudioDeviceInfo{
std::string name; // Character string device identifier.
bool probed; // true if the device capabilities were successfully probed.
int outputChannels; // Maximum output channels supported by device.
int inputChannels; // Maximum input channels supported by device.
int duplexChannels; // Maximum simultaneous input/output channels supported by device.
bool isDefault; // true if this is the default output or input device.
std::vector<int> sampleRates; // Supported sample rates.
RtAudioFormat nativeFormats; // Bit mask of supported data formats.
};
\endcode

The following data formats are defined and fully supported by RtAudio:

\code
typedef unsigned long RTAUDIO_FORMAT;
static const RTAUDIO_FORMAT RTAUDIO_SINT8; // Signed 8-bit integer
static const RTAUDIO_FORMAT RTAUDIO_SINT16; // Signed 16-bit integer
static const RTAUDIO_FORMAT RTAUDIO_SINT24; // Signed 24-bit integer (upper 3 bytes of 32-bit signed integer.)
static const RTAUDIO_FORMAT RTAUDIO_SINT32; // Signed 32-bit integer
static const RTAUDIO_FORMAT RTAUDIO_FLOAT32; // 32-bit float normalized between +/- 1.0
static const RTAUDIO_FORMAT RTAUDIO_FLOAT64; // 64-bit double normalized between +/- 1.0
typedef unsigned long RtAudioFormat;
static const RtAudioFormat RTAUDIO_SINT8; // Signed 8-bit integer
static const RtAudioFormat RTAUDIO_SINT16; // Signed 16-bit integer
static const RtAudioFormat RTAUDIO_SINT24; // Signed 24-bit integer (upper 3 bytes of 32-bit signed integer.)
static const RtAudioFormat RTAUDIO_SINT32; // Signed 32-bit integer
static const RtAudioFormat RTAUDIO_FLOAT32; // 32-bit float normalized between +/- 1.0
static const RtAudioFormat RTAUDIO_FLOAT64; // 64-bit double normalized between +/- 1.0
\endcode

The <I>nativeFormats</I> member of the RtAudio::RTAUDIO_DEVICE structure is a bit mask of the above formats which are natively supported by the device. However, RtAudio will automatically provide format conversion if a particular format is not natively supported. When the <I>probed</I> member of the RTAUDIO_DEVICE structure is false, the remaining structure members are undefined and the device is probably unuseable.
The <I>nativeFormats</I> member of the RtAudioDeviceInfo structure is a bit mask of the above formats which are natively supported by the device. However, RtAudio will automatically provide format conversion if a particular format is not natively supported. When the <I>probed</I> member of the RtAudioDeviceInfo structure is false, the remaining structure members are undefined and the device is probably unuseable.

In general, the user need not be concerned with the minimum channel values reported in the RTAUDIO_DEVICE structure. While some audio devices may require a minimum channel value > 1, RtAudio will provide automatic channel number compensation when the number of channels set by the user is less than that required by the device. Channel compensation is <I>NOT</I> possible when the number of channels set by the user is greater than that supported by the device.
While some audio devices may require a minimum channel value greater than one, RtAudio will provide automatic channel number compensation when the number of channels set by the user is less than that required by the device. Channel compensation is <I>NOT</I> possible when the number of channels set by the user is greater than that supported by the device.

It should be noted that the capabilities reported by a device driver or underlying audio API are not always accurate and/or may be dependent on a combination of device settings. For this reason, RtAudio does not rely on the reported values when attempting to open a stream.
It should be noted that the capabilities reported by a device driver or underlying audio API are not always accurate and/or may be dependent on a combination of device settings. For this reason, RtAudio does not typically rely on the queried values when attempting to open a stream.


\section settings Device Settings
@@ -178,24 +172,30 @@ The next step in using RtAudio is to open a stream with particular device and pa
int main()
{
int channels = 2;
int sample_rate = 44100;
int buffer_size = 256; // 256 sample frames
int n_buffers = 4; // number of internal buffers used by device
int device = 0; // 0 indicates the default or first available device
int stream; // our stream identifier
int sampleRate = 44100;
int bufferSize = 256; // 256 sample frames
int nBuffers = 4; // number of internal buffers used by device
int device = 0; // 0 indicates the default or first available device
RtAudio *audio;

// Instantiate RtAudio and open a stream within a try/catch block
try {
audio = new RtAudio();
stream = audio->openStream(device, channels, 0, 0, RtAudio::RTAUDIO_FLOAT32,
sample_rate, &buffer_size, n_buffers);
}
catch (RtError &error) {
error.printMessage();
exit(EXIT_FAILURE);
}

try {
audio->openStream(device, channels, 0, 0, RTAUDIO_FLOAT32,
sampleRate, &bufferSize, nBuffers);
}
catch (RtError &error) {
error.printMessage();
// Perhaps try other parameters?
}

// Clean up
delete audio;

@@ -203,11 +203,11 @@ int main()
}
\endcode

The RtAudio::openStream() method attempts to open a stream with a specified set of parameter values. When successful, a stream identifier is returned. In this case, we attempt to open a two channel playback stream with the default output device, 32-bit floating point data, a sample rate of 44100 Hz, a frame rate of 256 sample frames per read/write, and 4 internal device buffers. When device = 0, RtAudio first attempts to open the default audio device with the given parameters. If that attempt fails, RtAudio searches through the remaining available devices in an effort to find a device which will meet the given parameters. If all attempts are unsuccessful, an RtError is thrown. When a non-zero device value is specified, an attempt is made to open that device only (device = 1 specifies the first identified device, as reported by RtAudio::getDeviceInfo()).
The RtAudio::openStream() method attempts to open a stream with a specified set of parameter values. In this case, we attempt to open a two channel playback stream with the default output device, 32-bit floating point data, a sample rate of 44100 Hz, a frame rate of 256 sample frames per read/write, and 4 internal device buffers. When device = 0, RtAudio first attempts to open the default audio device with the given parameters. If that attempt fails, RtAudio searches through the remaining available devices in an effort to find a device which will meet the given parameters. If all attempts are unsuccessful, an RtError is thrown. When a non-zero device value is specified, an attempt is made to open that device \e ONLY (device = 1 specifies the first identified device, as reported by RtAudio::getDeviceInfo()).

RtAudio provides four signed integer and two floating point data formats which can be specified using the RtAudio::RTAUDIO_FORMAT parameter values mentioned earlier. If the opened device does not natively support the given format, RtAudio will automatically perform the necessary data format conversion.
RtAudio provides four signed integer and two floating point data formats which can be specified using the RtAudioFormat parameter values mentioned earlier. If the opened device does not natively support the given format, RtAudio will automatically perform the necessary data format conversion.

The <I>bufferSize</I> parameter specifies the desired number of sample frames which will be written to and/or read from a device per write/read operation. The <I>nBuffers</I> parameter is used in setting the underlying device buffer parameters. Both the <I>bufferSize</I> and <I>nBuffers</I> parameters can be used to control stream latency though there is no guarantee that the passed values will be those used by a device (the <I>nBuffers</I> parameter is ignored when using the OS X CoreAudio and the Windows ASIO APIs). In general, lower values for both parameters will produce less latency but perhaps less robust performance. Both parameters can be specified with values of zero, in which case the smallest allowable values will be used. The <I>bufferSize</I> parameter is passed as a pointer and the actual value used by the stream is set during the device setup procedure. <I>bufferSize</I> values should be a power of two. Optimal and allowable buffer values tend to vary between systems and devices. Check the \ref osnotes section for general guidelines.
The <I>bufferSize</I> parameter specifies the desired number of sample frames which will be written to and/or read from a device per write/read operation. The <I>nBuffers</I> parameter is used in setting the underlying device buffer parameters. Both the <I>bufferSize</I> and <I>nBuffers</I> parameters can be used to control stream latency though there is no guarantee that the passed values will be those used by a device (the <I>nBuffers</I> parameter is ignored when using the OS X CoreAudio, Linux Jack, and the Windows ASIO APIs). In general, lower values for both parameters will produce less latency but perhaps less robust performance. Both parameters can be specified with values of zero, in which case the smallest allowable values will be used. The <I>bufferSize</I> parameter is passed as a pointer and the actual value used by the stream is set during the device setup procedure. <I>bufferSize</I> values should be a power of two. Optimal and allowable buffer values tend to vary between systems and devices. Check the \ref apinotes section for general guidelines.

As noted earlier, the device capabilities reported by a driver or underlying audio API are not always accurate and/or may be dependent on a combination of device settings. Because of this, RtAudio does not attempt to query a device's capabilities or use previously reported values when opening a device. Instead, RtAudio simply attempts to set the given parameters on a specified device and then checks whether the setup is successful or not.

@@ -225,18 +225,17 @@ int main()
{
int count;
int channels = 2;
int sample_rate = 44100;
int buffer_size = 256; // 256 sample frames
int n_buffers = 4; // number of internal buffers used by device
int sampleRate = 44100;
int bufferSize = 256; // 256 sample frames
int nBuffers = 4; // number of internal buffers used by device
float *buffer;
int device = 0; // 0 indicates the default or first available device
int stream; // our stream identifier
int device = 0; // 0 indicates the default or first available device
RtAudio *audio;

// Open a stream during RtAudio instantiation
try {
audio = new RtAudio(&stream, device, channels, 0, 0, RtAudio::RTAUDIO_FLOAT32,
sample_rate, &buffer_size, n_buffers);
audio = new RtAudio(device, channels, 0, 0, RTAUDIO_FLOAT32,
sampleRate, &bufferSize, nBuffers);
}
catch (RtError &error) {
error.printMessage();
@@ -245,38 +244,38 @@ int main()

try {
// Get a pointer to the stream buffer
buffer = (float *) audio->getStreamBuffer(stream);
buffer = (float *) audio->getStreamBuffer();

// Start the stream
audio->startStream(stream);
audio->startStream();
}
catch (RtError &error) {
error.printMessage();
goto cleanup;
}

// An example loop which runs for about 40000 sample frames
// An example loop which runs for 40000 sample frames
count = 0;
while (count < 40000) {
// Generate your samples and fill the buffer with buffer_size sample frames of data
// Generate your samples and fill the buffer with bufferSize sample frames of data
...

// Trigger the output of the data buffer
try {
audio->tickStream(stream);
audio->tickStream();
}
catch (RtError &error) {
error.printMessage();
goto cleanup;
}

count += buffer_size;
count += bufferSize;
}

try {
// Stop and close the stream
audio->stopStream(stream);
audio->closeStream(stream);
audio->stopStream();
audio->closeStream();
}
catch (RtError &error) {
error.printMessage();
@@ -289,9 +288,9 @@ int main()
}
\endcode

The first thing to notice in this example is that we attempt to open a stream during class instantiation with an overloaded constructor. This constructor simply combines the functionality of the default constructor, used earlier, and the RtAudio::openStream() method. Again, we have specified a device value of 0, indicating that the default or first available device meeting the given parameters should be used. The integer identifier of the opened stream is returned via the <I>stream</I> pointer value. An attempt is made to open the stream with the specified <I>bufferSize</I> value. However, it is possible that the device will not accept this value, in which case the closest allowable size is used and returned via the pointer value. The constructor can fail if no available devices are found, or a memory allocation or device driver error occurs. Note that you should not call the RtAudio destructor if an exception is thrown during instantiation.
The first thing to notice in this example is that we attempt to open a stream during class instantiation with an overloaded constructor. This constructor simply combines the functionality of the default constructor, used earlier, and the RtAudio::openStream() method. Again, we have specified a device value of 0, indicating that the default or first available device meeting the given parameters should be used. An attempt is made to open the stream with the specified <I>bufferSize</I> value. However, it is possible that the device will not accept this value, in which case the closest allowable size is used and returned via the pointer value. The constructor can fail if no available devices are found, or a memory allocation or device driver error occurs. Note that you should not call the RtAudio destructor if an exception is thrown during instantiation.

Because RtAudio can typically be used to simultaneously control more than a single stream, it is necessary that the stream identifier be provided to nearly all public methods. Assuming the constructor is successful, it is necessary to get a pointer to the buffer, provided by RtAudio, for use in feeding data to/from the opened stream. Note that the user should <I>NOT</I> attempt to deallocate the stream buffer memory ... memory management for the stream buffer will be automatically controlled by RtAudio. After starting the stream with RtAudio::startStream(), one simply fills that buffer, which is of length equal to the returned <I>bufferSize</I> value, with interleaved audio data (in the specified format) for playback. Finally, a call to the RtAudio::tickStream() routine triggers a blocking write call for the stream.
Assuming the constructor is successful, it is necessary to get a pointer to the buffer, provided by RtAudio, for use in feeding data to/from the opened stream. Note that the user should <I>NOT</I> attempt to deallocate the stream buffer memory ... memory management for the stream buffer will be automatically controlled by RtAudio. After starting the stream with RtAudio::startStream(), one simply fills that buffer, which is of length equal to the returned <I>bufferSize</I> value, with interleaved audio data (in the specified format) for playback. Finally, a call to the RtAudio::tickStream() routine triggers a blocking write call for the stream.

In general, one should call the RtAudio::stopStream() and RtAudio::closeStream() methods after finishing with a stream. However, both methods will implicitly be called during object destruction if necessary.

@@ -306,14 +305,14 @@ The primary difference in using RtAudio with callback functionality involves the
#include "RtAudio.h"

// Two-channel sawtooth wave generator.
int sawtooth(char *buffer, int buffer_size, void *data)
int sawtooth(char *buffer, int bufferSize, void *data)
{
int i, j;
double *my_buffer = (double *) buffer;
double *my_data = (double *) data;

// Write interleaved audio data.
for (i=0; i<buffer_size; i++) {
for (i=0; i<bufferSize; i++) {
for (j=0; j<2; j++) {
*my_buffer++ = my_data[j];

@@ -328,19 +327,18 @@ int sawtooth(char *buffer, int buffer_size, void *data)
int main()
{
int channels = 2;
int sample_rate = 44100;
int buffer_size = 256; // 256 sample frames
int n_buffers = 4; // number of internal buffers used by device
int device = 0; // 0 indicates the default or first available device
int stream; // our stream identifier
int sampleRate = 44100;
int bufferSize = 256; // 256 sample frames
int nBuffers = 4; // number of internal buffers used by device
int device = 0; // 0 indicates the default or first available device
double data[2];
char input;
RtAudio *audio;

// Open a stream during RtAudio instantiation
try {
audio = new RtAudio(&stream, device, channels, 0, 0, RtAudio::RTAUDIO_FLOAT64,
sample_rate, &buffer_size, n_buffers);
audio = new RtAudio(device, channels, 0, 0, RTAUDIO_FLOAT64,
sampleRate, &bufferSize, nBuffers);
}
catch (RtError &error) {
error.printMessage();
@@ -349,23 +347,23 @@ int main()

try {
// Set the stream callback function
audio->setStreamCallback(stream, &sawtooth, (void *)data);
audio->setStreamCallback(&sawtooth, (void *)data);

// Start the stream
audio->startStream(stream);
audio->startStream();
}
catch (RtError &error) {
error.printMessage();
goto cleanup;
}

cout << "\nPlaying ... press <enter> to quit.\n";
cin.get(input);
std::cout << "\nPlaying ... press <enter> to quit.\n";
std::cin.get(input);

try {
// Stop and close the stream
audio->stopStream(stream);
audio->closeStream(stream);
audio->stopStream();
audio->closeStream();
}
catch (RtError &error) {
error.printMessage();
@@ -378,7 +376,7 @@ int main()
}
\endcode

After opening the device in exactly the same way as the previous example (except with a data format change), we must set our callback function for the stream using RtAudio::setStreamCallback(). When the underlying audio API uses blocking calls (OSS, ALSA, SGI, and Windows DirectSound), this method will spawn a new process (or thread) which automatically calls the callback function when more data is needed. Callback-based audio APIs (OS X CoreAudio and ASIO) implement their own event notification schemes. Note that the callback function is called only when the stream is "running" (between calls to the RtAudio::startStream() and RtAudio::stopStream() methods). The last argument to RtAudio::setStreamCallback() is a pointer to arbitrary data that you wish to access from within your callback function.
After opening the device in exactly the same way as the previous example (except with a data format change), we must set our callback function for the stream using RtAudio::setStreamCallback(). When the underlying audio API uses blocking calls (OSS, ALSA, SGI, and Windows DirectSound), this method will spawn a new process (or thread) which automatically calls the callback function when more data is needed. Callback-based audio APIs (OS X CoreAudio Linux Jack, and ASIO) implement their own event notification schemes. Note that the callback function is called only when the stream is "running" (between calls to the RtAudio::startStream() and RtAudio::stopStream() methods). The last argument to RtAudio::setStreamCallback() is a pointer to arbitrary data that you wish to access from within your callback function.

In this example, we stop the stream with an explicit call to RtAudio::stopStream(). When using callback functionality, it is also possible to stop a stream by returning a non-zero value from the callback function.

@@ -398,18 +396,17 @@ int main()
{
int count;
int channels = 2;
int sample_rate = 44100;
int buffer_size = 256; // 256 sample frames
int n_buffers = 4; // number of internal buffers used by device
int sampleRate = 44100;
int bufferSize = 256; // 256 sample frames
int nBuffers = 4; // number of internal buffers used by device
float *buffer;
int device = 0; // 0 indicates the default or first available device
int stream; // our stream identifier
int device = 0; // 0 indicates the default or first available device
RtAudio *audio;

// Instantiate RtAudio and open a stream.
try {
audio = new RtAudio(&stream, 0, 0, device, channels,
RtAudio::RTAUDIO_FLOAT32, sample_rate, &buffer_size, n_buffers);
RTAUDIO_FLOAT32, sampleRate, &bufferSize, nBuffers);
}
catch (RtError &error) {
error.printMessage();
@@ -418,10 +415,10 @@ int main()

try {
// Get a pointer to the stream buffer
buffer = (float *) audio->getStreamBuffer(stream);
buffer = (float *) audio->getStreamBuffer();

// Start the stream
audio->startStream(stream);
audio->startStream();
}
catch (RtError &error) {
error.printMessage();
@@ -434,22 +431,22 @@ int main()

// Read a buffer of data
try {
audio->tickStream(stream);
audio->tickStream();
}
catch (RtError &error) {
error.printMessage();
goto cleanup;
}

// Process the input samples (buffer_size sample frames) that were read
// Process the input samples (bufferSize sample frames) that were read
...

count += buffer_size;
count += bufferSize;
}

try {
// Stop the stream
audio->stopStream(stream);
audio->stopStream();
}
catch (RtError &error) {
error.printMessage();
@@ -476,13 +473,13 @@ Finally, it is easy to use RtAudio for simultaneous audio input/output, or duple
#include "RtAudio.h"

// Pass-through function.
int scale(char *buffer, int buffer_size, void *)
int scale(char *buffer, int bufferSize, void *)
{
// Note: do nothing here for pass through.
double *my_buffer = (double *) buffer;

// Scale input data for output.
for (int i=0; i<buffer_size; i++) {
for (int i=0; i<bufferSize; i++) {
// Do for two channels.
*my_buffer++ *= 0.5;
*my_buffer++ *= 0.5;
@@ -494,18 +491,17 @@ int scale(char *buffer, int buffer_size, void *)
int main()
{
int channels = 2;
int sample_rate = 44100;
int buffer_size = 256; // 256 sample frames
int n_buffers = 4; // number of internal buffers used by device
int device = 0; // 0 indicates the default or first available device
int stream; // our stream identifier
int sampleRate = 44100;
int bufferSize = 256; // 256 sample frames
int nBuffers = 4; // number of internal buffers used by device
int device = 0; // 0 indicates the default or first available device
char input;
RtAudio *audio;

// Open a stream during RtAudio instantiation
try {
audio = new RtAudio(&stream, device, channels, device, channels, RtAudio::RTAUDIO_FLOAT64,
sample_rate, &buffer_size, n_buffers);
audio = new RtAudio(device, channels, device, channels, RTAUDIO_FLOAT64,
sampleRate, &bufferSize, nBuffers);
}
catch (RtError &error) {
error.printMessage();
@@ -514,23 +510,23 @@ int main()

try {
// Set the stream callback function
audio->setStreamCallback(stream, &scale, NULL);
audio->setStreamCallback(&scale, NULL);

// Start the stream
audio->startStream(stream);
audio->startStream();
}
catch (RtError &error) {
error.printMessage();
goto cleanup;
}

cout << "\nRunning duplex ... press <enter> to quit.\n";
cin.get(input);
std::cout << "\nRunning duplex ... press <enter> to quit.\n";
std::cin.get(input);

try {
// Stop and close the stream
audio->stopStream(stream);
audio->closeStream(stream);
audio->stopStream();
audio->closeStream();
}
catch (RtError &error) {
error.printMessage();
@@ -547,26 +543,30 @@ When an RtAudio stream is running in duplex mode (nonzero input <I>AND</I> outpu

As we see with this example, the write-read sequence of operations does not preclude the use of RtAudio in situations where input data is first processed and then output through a duplex stream. When the stream buffer is first allocated, it is initialized with zeros, which produces no audible result when output to the device. In this example, anything recorded by the audio stream input will be scaled and played out during the next round of audio processing.

Note that duplex operation can also be achieved by opening one output stream and one input stream using the same or different devices. However, there may be timing problems when attempting to use two different devices, due to possible device clock variations, unless a common external "sync" is provided. This becomes even more difficult to achieve using two separate callback streams because it is not possible to <I>explicitly</I> control the calling order of the callback functions.
Note that duplex operation can also be achieved by opening one output stream instance and one input stream instance using the same or different devices. However, there may be timing problems when attempting to use two different devices, due to possible device clock variations, unless a common external "sync" is provided. This becomes even more difficult to achieve using two separate callback streams because it is not possible to <I>explicitly</I> control the calling order of the callback functions.


\section multi Using Simultaneous Multiple APIs

Because support for each audio API is encapsulated in a specific RtApi subclass, it is possible to compile and instantiate multiple API-specific subclasses on a given operating system. For example, one can compile both the RtApiDs and RtApiAsio classes on Windows operating systems by providing the appropriate preprocessor definitions, include files, and libraries for each. In a run-time situation, one might first attempt to determine whether any ASIO device drivers exist. This can be done by specifying the api argument RtAudio::WINDOWS_ASIO when attempting to create an instance of RtAudio. If an RtError is thrown (indicating no available drivers), then an instance of RtAudio with the api argument RtAudio::WINDOWS_DS can be created. Alternately, if no api argument is specified, RtAudio will first look for ASIO drivers and then DirectSound drivers (on Linux systems, the default API search order is Jack, Alsa, and finally OSS). In theory, it should also be possible to have separate instances of RtAudio open at the same time with different underlying audio API support, though this has not been tested. It is difficult to know how well different audio APIs can simultaneously coexist on a given operating system. In particular, it is most unlikely that the same device could be simultaneously controlled with two different audio APIs.


\section methods Summary of Methods

The following is short summary of public methods (not including constructors and the destructor) provided by RtAudio:
The following is a short summary of public methods (not including constructors and the destructor) provided by RtAudio:

<UL>
<LI>RtAudio::openStream(): opens a stream with the specified parameters.</LI>
<LI>RtAudio::setStreamCallback(): sets a user-defined callback function for a given stream.</LI>
<LI>RtAudio::cancelStreamCallback(): cancels a callback process and function for a given stream.</LI>
<LI>RtAudio::setStreamCallback(): sets a user-defined callback function for the stream.</LI>
<LI>RtAudio::cancelStreamCallback(): cancels a callback process and function for the stream.</LI>
<LI>RtAudio::getDeviceCount(): returns the number of audio devices available.</LI>
<LI>RtAudio::getDeviceInfo(): fills a user-supplied RTAUDIO_DEVICE structure for a specified device.</LI>
<LI>RtAudio::getDeviceInfo(): returns an RtAudioDeviceInfo structure for a specified device.</LI>
<LI>RtAudio::getStreamBuffer(): returns a pointer to the stream buffer.</LI>
<LI>RtAudio::tickStream(): triggers processing of input/output data for a stream (blocking).</LI>
<LI>RtAudio::closeStream(): closes the specified stream (implicitly called during object destruction). Once a stream is closed, the stream identifier is invalid and should not be used in calling any other RtAudio methods.</LI>
<LI>RtAudio::startStream(): (re)starts the specified stream, typically after it has been stopped with either stopStream() or abortStream() or after first opening the stream.</LI>
<LI>RtAudio::stopStream(): stops the specified stream, allowing any remaining samples in the queue to be played out and/or read in. This does not implicitly call RtAudio::closeStream().</LI>
<LI>RtAudio::abortStream(): stops the specified stream, discarding any remaining samples in the queue. This does not implicitly call closeStream().</LI>
<LI>RtAudio::streamWillBlock(): queries a stream to determine whether a call to the <I>tickStream()</I> method will block. A return value of 0 indicates that the stream will NOT block. A positive return value indicates the number of sample frames that cannot yet be processed without blocking.</LI>
<LI>RtAudio::tickStream(): triggers processing of input/output data for the stream (blocking).</LI>
<LI>RtAudio::closeStream(): closes the stream (implicitly called during object destruction).</LI>
<LI>RtAudio::startStream(): (re)starts the stream, typically after it has been stopped with either stopStream() or abortStream() or after first opening the stream.</LI>
<LI>RtAudio::stopStream(): stops the stream, allowing any remaining samples in the queue to be played out and/or read in. This does not implicitly call RtAudio::closeStream().</LI>
<LI>RtAudio::abortStream(): stops the stream, discarding any remaining samples in the queue. This does not implicitly call closeStream().</LI>
</UL>


@@ -579,6 +579,7 @@ In order to compile RtAudio for a specific OS and audio API, it is necessary to
<TR BGCOLOR="beige">
<TD WIDTH="5%"><B>OS:</B></TD>
<TD WIDTH="5%"><B>Audio API:</B></TD>
<TD WIDTH="5%"><B>C++ Class:</B></TD>
<TD WIDTH="5%"><B>Preprocessor Definition:</B></TD>
<TD WIDTH="5%"><B>Library or Framework:</B></TD>
<TD><B>Example Compiler Statement:</B></TD>
@@ -586,13 +587,23 @@ In order to compile RtAudio for a specific OS and audio API, it is necessary to
<TR>
<TD>Linux</TD>
<TD>ALSA</TD>
<TD>RtApiAlsa</TD>
<TD>__LINUX_ALSA__</TD>
<TD><TT>asound, pthread</TT></TD>
<TD><TT>g++ -Wall -D__LINUX_ALSA__ -o probe probe.cpp RtAudio.cpp -lasound -lpthread</TT></TD>
</TR>
<TR>
<TD>Linux</TD>
<TD>Jack Audio Server</TD>
<TD>RtApiJack</TD>
<TD>__LINUX_JACK__</TD>
<TD><TT>jack, pthread</TT></TD>
<TD><TT>g++ -Wall -D__LINUX_JACK__ -o probe probe.cpp RtAudio.cpp `pkg-config --cflags --libs jack` -lpthread</TT></TD>
</TR>
<TR>
<TD>Linux</TD>
<TD>OSS</TD>
<TD>RtApiOss</TD>
<TD>__LINUX_OSS__</TD>
<TD><TT>pthread</TT></TD>
<TD><TT>g++ -Wall -D__LINUX_OSS__ -o probe probe.cpp RtAudio.cpp -lpthread</TT></TD>
@@ -600,13 +611,15 @@ In order to compile RtAudio for a specific OS and audio API, it is necessary to
<TR>
<TD>Macintosh OS X</TD>
<TD>CoreAudio</TD>
<TD>RtApiCore</TD>
<TD>__MACOSX_CORE__</TD>
<TD><TT>pthread, stdc++, CoreAudio</TT></TD>
<TD><TT>CC -Wall -D__MACOSX_CORE__ -o probe probe.cpp RtAudio.cpp -framework CoreAudio -lstdc++ -lpthread</TT></TD>
<TD><TT>g++ -Wall -D__MACOSX_CORE__ -o probe probe.cpp RtAudio.cpp -framework CoreAudio -lpthread</TT></TD>
</TR>
<TR>
<TD>Irix</TD>
<TD>AL</TD>
<TD>RtApiAl</TD>
<TD>__IRIX_AL__</TD>
<TD><TT>audio, pthread</TT></TD>
<TD><TT>CC -Wall -D__IRIX_AL__ -o probe probe.cpp RtAudio.cpp -laudio -lpthread</TT></TD>
@@ -614,6 +627,7 @@ In order to compile RtAudio for a specific OS and audio API, it is necessary to
<TR>
<TD>Windows</TD>
<TD>Direct Sound</TD>
<TD>RtApiDs</TD>
<TD>__WINDOWS_DS__</TD>
<TD><TT>dsound.lib (ver. 5.0 or higher), multithreaded</TT></TD>
<TD><I>compiler specific</I></TD>
@@ -621,6 +635,7 @@ In order to compile RtAudio for a specific OS and audio API, it is necessary to
<TR>
<TD>Windows</TD>
<TD>ASIO</TD>
<TD>RtApiAsio</TD>
<TD>__WINDOWS_ASIO__</TD>
<TD><I>various ASIO header and source files</I></TD>
<TD><I>compiler specific</I></TD>
@@ -628,27 +643,31 @@ In order to compile RtAudio for a specific OS and audio API, it is necessary to
</TABLE>
<P>

The example compiler statements above could be used to compile the <TT>probe.cpp</TT> example file, assuming that <TT>probe.cpp</TT>, <TT>RtAudio.h</TT>, and <TT>RtAudio.cpp</TT> all exist in the same directory.
The example compiler statements above could be used to compile the <TT>probe.cpp</TT> example file, assuming that <TT>probe.cpp</TT>, <TT>RtAudio.h</TT>, <tt>RtError.h</tt>, and <TT>RtAudio.cpp</TT> all exist in the same directory.

\section debug Debugging

If you are having problems getting RtAudio to run on your system, try passing the preprocessor definition <TT>__RTAUDIO_DEBUG__</TT> to the compiler (or uncomment the definition at the bottom of RtAudio.h). A variety of warning messages will be displayed which may help in determining the problem.
If you are having problems getting RtAudio to run on your system, try passing the preprocessor definition <TT>__RTAUDIO_DEBUG__</TT> to the compiler (or uncomment the definition at the bottom of RtAudio.h). A variety of warning messages will be displayed which may help in determining the problem. Also try using the programs included in the <tt>test</tt> directory. The program <tt>info</tt> displays the queried capabilities of all hardware devices found.

\section osnotes OS Notes
\section apinotes API Notes

RtAudio is designed to provide a common API across the various supported operating systems and audio libraries. Despite that, some issues need to be mentioned with regard to each.
RtAudio is designed to provide a common API across the various supported operating systems and audio libraries. Despite that, some issues should be mentioned with regard to each.

\subsection linux Linux:

RtAudio for Linux was developed under Redhat distributions 7.0 - 7.2. Two different audio APIs are supported on Linux platforms: OSS and <A href="http://www.alsa-project.org/">ALSA</A>. The OSS API has existed for at least 6 years and the Linux kernel is distributed with free versions of OSS audio drivers. Therefore, a generic Linux system is most likely to have OSS support. The ALSA API, although relatively new, is now part of the Linux development kernel and offers significantly better functionality than the OSS API. RtAudio provides support for the 0.9 and higher versions of ALSA. Input/output latency on the order of 15 milliseconds can typically be achieved under both OSS or ALSA by fine-tuning the RtAudio buffer parameters (without kernel modifications). Latencies on the order of 5 milliseconds or less can be achieved using a low-latency kernel patch and increasing FIFO scheduling priority. The pthread library, which is used for callback functionality, is a standard component of all Linux distributions.
RtAudio for Linux was developed under Redhat distributions 7.0 - Fedora. Three different audio APIs are supported on Linux platforms: OSS, <A href="http://www.alsa-project.org/">ALSA</A>, and <A href="http://jackit.sourceforge.net/">Jack</A>. The OSS API has existed for at least 6 years and the Linux kernel is distributed with free versions of OSS audio drivers. Therefore, a generic Linux system is most likely to have OSS support (though the availability and quality of OSS drivers for new hardware is decreasing). The ALSA API, although relatively new, is now part of the Linux development kernel and offers significantly better functionality than the OSS API. RtAudio provides support for the 1.0 and higher versions of ALSA. Jack, which is still in development, is a low-latency audio server, written primarily for the GNU/Linux operating system. It can connect a number of different applications to an audio device, as well as allow them to share audio between themselves. Input/output latency on the order of 15 milliseconds can typically be achieved using any of the Linux APIs by fine-tuning the RtAudio buffer parameters (without kernel modifications). Latencies on the order of 5 milliseconds or less can be achieved using a low-latency kernel patch and increasing FIFO scheduling priority. The pthread library, which is used for callback functionality, is a standard component of all Linux distributions.

The ALSA library includes OSS emulation support. That means that you can run programs compiled for the OSS API even when using the ALSA drivers and library. It should be noted however that OSS emulation under ALSA is not perfect. Specifically, channel number queries seem to consistently produce invalid results. While OSS emulation is successful for the majority of RtAudio tests, it is recommended that the native ALSA implementation of RtAudio be used on systems which have ALSA drivers installed.

The ALSA implementation of RtAudio makes no use of the ALSA "plug" interface. All necessary data format conversions, channel compensation, de-interleaving, and byte-swapping is handled by internal RtAudio routines.

The Jack API is based on a callback scheme. RtAudio provides blocking functionality, in addition to callback functionality, within the context of that behavior. It should be noted, however, that the best performance is achieved when using RtAudio's callback functionality with the Jack API. At the moment, only one RtAudio instance can be connected to the Jack server. Because RtAudio does not provide a mechanism for allowing the user to specify particular channels (or ports) of a device, it simply opens the first <I>N</I> enumerated Jack ports for input/output.

\subsection macosx Macintosh OS X (CoreAudio):

The Apple CoreAudio API is based on a callback scheme. RtAudio provides blocking functionality, in addition to callback functionality, within the context of that behavior. CoreAudio is designed to use a separate callback procedure for each of its audio devices. A single RtAudio duplex stream using two different devices is supported, though it cannot be guaranteed to always behave correctly because we cannot synchronize these two callbacks. This same functionality can be achieved with better synchrony by opening two separate streams for the devices and using RtAudio blocking calls (i.e. RtAudio::tickStream()). The <I>numberOfBuffers</I> parameter to the RtAudio::openStream() function has no affect in this implementation. It is not currently possible to have multiple simultaneous RtAudio streams accessing the same device.
The Apple CoreAudio API is based on a callback scheme. RtAudio provides blocking functionality, in addition to callback functionality, within the context of that behavior. CoreAudio is designed to use a separate callback procedure for each of its audio devices. A single RtAudio duplex stream using two different devices is supported, though it cannot be guaranteed to always behave correctly because we cannot synchronize these two callbacks. This same functionality might be achieved with better synchrony by creating separate instances of RtAudio for each device and making use of RtAudio blocking calls (i.e. RtAudio::tickStream()). The <I>numberOfBuffers</I> parameter to the RtAudio::openStream() function has no affect in this implementation.

It is not possible to have multiple instances of RtAudio accessing the same CoreAudio device.

\subsection irix Irix (SGI):

@@ -656,25 +675,25 @@ The Irix version of RtAudio was written and tested on an SGI Indy running Irix v

\subsection windowsds Windows (DirectSound):

In order to compile RtAudio under Windows for the DirectSound API, you must have the header and source files for DirectSound version 5.0 or higher. As far as I know, there is no DirectSoundCapture support for Windows NT. Audio output latency with DirectSound can be reasonably good (on the order of 20 milliseconds). On the other hand, input audio latency tends to be terrible (100 milliseconds or more). Further, DirectSound drivers tend to crash easily when experimenting with buffer parameters. On my system, I found it necessary to use values around nBuffers = 8 and bufferSize = 512 to avoid crashes. RtAudio was developed with Visual C++ version 6.0. I was forced in several instances to modify code in order to get it to compile under the non-standard version of C++ that Microsoft so unprofessionally implemented. Unfortunately, it appears they are continuing to undermine the C++ standard with more recent compiler releases.
In order to compile RtAudio under Windows for the DirectSound API, you must have the header and source files for DirectSound version 5.0 or higher. As far as I know, there is no DirectSoundCapture support for Windows NT. Audio output latency with DirectSound can be reasonably good (on the order of 20 milliseconds). On the other hand, input audio latency tends to be terrible (100 milliseconds or more). Further, DirectSound drivers tend to crash easily when experimenting with buffer parameters. On my system, I found it necessary to use values around nBuffers = 8 and bufferSize = 512 to avoid crashes. RtAudio was originally developed with Visual C++ version 6.0.

\subsection windowsasio Windows (ASIO):

The Steinberg ASIO audio API is based on a callback scheme. In addition, the API allows only a single device driver to be loaded and accessed at a time. Therefore, it is not possible to have multiple simultaneous RtAudio streams running concurrently with this API. ASIO device drivers must be supplied by audio hardware manufacturers, though ASIO emulation is possible on top of systems with DirectSound drivers. The <I>numberOfBuffers</I> parameter to the RtAudio::openStream() function has no affect in this implementation.
The Steinberg ASIO audio API is based on a callback scheme. In addition, the API allows only a single device driver to be loaded and accessed at a time. ASIO device drivers must be supplied by audio hardware manufacturers, though ASIO emulation is possible on top of systems with DirectSound drivers. The <I>numberOfBuffers</I> parameter to the RtAudio::openStream() function has no affect in this implementation.

A number of ASIO source and header files are required for use with RtAudio. Specifically, an RtAudio project must include the following files: <TT>asio.h,cpp; asiodrivers.h,cpp; asiolist.h,cpp; asiodrvr.h; asiosys.h; ginclude.h; iasiodrv.h</TT>. See the <TT>/tests/asio/</TT> directory for example Visual C++ 6.0 projects.
A number of ASIO source and header files are required for use with RtAudio. Specifically, an RtAudio project must include the following files: <TT>asio.h,cpp; asiodrivers.h,cpp; asiolist.h,cpp; asiodrvr.h; asiosys.h; ginclude.h; iasiodrv.h</TT>. The Visual C++ projects found in <TT>/tests/Windows/</TT> compile both ASIO and DirectSound support.


\section acknowledge Acknowledgements

The RtAudio API incorporates many of the concepts developed in the <A href="http://www.portaudio.com/">PortAudio</A> project by Phil Burk and Ross Bencina. Early development also incorporated ideas from Bill Schottstaedt's <A href="http://www-ccrma.stanford.edu/software/snd/sndlib/">sndlib</A>. The CCRMA <A href="http://www-ccrma.stanford.edu/groups/soundwire/">SoundWire group</A> provided valuable feedback during the API proposal stages.

RtAudio, version 2.0, was slowly developed over the course of many months while in residence at the <A href="http://www.iua.upf.es/">Institut Universitari de L'Audiovisual (IUA)</A> in Barcelona, Spain, the <A href="http://www.acoustics.hut.fi/">Laboratory of Acoustics and Audio Signal Processing</A> at the Helsinki University of Technology, Finland, and the <A href="http://www-ccrma.stanford.edu/">Center for Co