RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/
RtAudio: realtime audio i/o C++ classes
- Copyright (c) 2001-2008 Gary P. Scavone
+ Copyright (c) 2001-2013 Gary P. Scavone
Permission is hereby granted, free of charge, to any person
obtaining a copy of this software and associated documentation files
\file RtAudio.h
*/
-// RtAudio: Version 4.0.4
-
#ifndef __RTAUDIO_H
#define __RTAUDIO_H
#include <vector>
#include "RtError.h"
+// RtAudio version
+static const std::string VERSION( "4.0.12" );
+
/*! \typedef typedef unsigned long RtAudioFormat;
\brief RtAudio data format type.
- \e RTAUDIO_SINT8: 8-bit signed integer.
- \e RTAUDIO_SINT16: 16-bit signed integer.
- - \e RTAUDIO_SINT24: Upper 3 bytes of 32-bit signed integer.
+ - \e RTAUDIO_SINT24: 24-bit signed integer.
- \e RTAUDIO_SINT32: 32-bit signed integer.
- \e RTAUDIO_FLOAT32: Normalized between plus/minus 1.0.
- \e RTAUDIO_FLOAT64: Normalized between plus/minus 1.0.
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; // Lower 3 bytes of 32-bit signed integer.
+static const RtAudioFormat RTAUDIO_SINT24 = 0x4; // 24-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.
- \e RTAUDIO_NONINTERLEAVED: Use non-interleaved buffers (default = interleaved).
- \e RTAUDIO_MINIMIZE_LATENCY: Attempt to set stream parameters for lowest possible latency.
- \e RTAUDIO_HOG_DEVICE: Attempt grab device for exclusive use.
+ - \e RTAUDIO_ALSA_USE_DEFAULT: Use the "default" PCM device (ALSA only).
By default, RtAudio streams pass and receive audio data from the
client in an interleaved format. By passing the
If the RTAUDIO_SCHEDULE_REALTIME flag is set, RtAudio will attempt
to select realtime scheduling (round-robin) for the callback thread.
+
+ If the RTAUDIO_ALSA_USE_DEFAULT flag is set, RtAudio will attempt to
+ open the "default" PCM device when using the ALSA API. Note that this
+ will override any specified input or output device id.
*/
typedef unsigned int RtAudioStreamFlags;
static const RtAudioStreamFlags RTAUDIO_NONINTERLEAVED = 0x1; // Use non-interleaved buffers (default = interleaved).
static const RtAudioStreamFlags RTAUDIO_MINIMIZE_LATENCY = 0x2; // Attempt to set stream parameters for lowest possible latency.
static const RtAudioStreamFlags RTAUDIO_HOG_DEVICE = 0x4; // Attempt grab device and prevent use by others.
static const RtAudioStreamFlags RTAUDIO_SCHEDULE_REALTIME = 0x8; // Try to select realtime scheduling for callback thread.
+static const RtAudioStreamFlags RTAUDIO_ALSA_USE_DEFAULT = 0x10; // Use the "default" PCM device (ALSA only).
/*! \typedef typedef unsigned long RtAudioStreamStatus;
\brief RtAudio stream status (over- or underflow) flags.
RtAudioStreamStatus status,
void *userData );
+//! RtAudio error callback function prototype.
+/*!
+ \param type Type of error.
+ \param errorText Error description.
+ */
+typedef void (*RtAudioErrorCallback)( RtError::Type type, const std::string &errorText );
// **************************************************************** //
//
enum Api {
UNSPECIFIED, /*!< Search for a working compiled API. */
LINUX_ALSA, /*!< The Advanced Linux Sound Architecture API. */
+ LINUX_PULSE, /*!< The Linux PulseAudio API. */
LINUX_OSS, /*!< The Linux Open Sound System API. */
UNIX_JACK, /*!< The Jack Low-Latency Audio Server API. */
MACOSX_CORE, /*!< Macintosh OS-X Core Audio API. */
- \e RTAUDIO_MINIMIZE_LATENCY: Attempt to set stream parameters for lowest possible latency.
- \e RTAUDIO_HOG_DEVICE: Attempt grab device for exclusive use.
- \e RTAUDIO_SCHEDULE_REALTIME: Attempt to select realtime scheduling for callback thread.
+ - \e RTAUDIO_ALSA_USE_DEFAULT: Use the "default" PCM device (ALSA only).
By default, RtAudio streams pass and receive audio data from the
client in an interleaved format. By passing the
If the RTAUDIO_SCHEDULE_REALTIME flag is set, RtAudio will attempt
to select realtime scheduling (round-robin) for the callback thread.
The \c priority parameter will only be used if the RTAUDIO_SCHEDULE_REALTIME
- flag is set. It defines the thread's realtime priority.
+ flag is set. It defines the thread's realtime priority.
+
+ If the RTAUDIO_ALSA_USE_DEFAULT flag is set, RtAudio will attempt to
+ open the "default" PCM device when using the ALSA API. Note that this
+ will override any specified input or output device id.
The \c numberOfBuffers parameter can be used to control stream
latency in the Windows DirectSound, Linux OSS, and Linux Alsa APIs
RtAudio with Jack, each instance must have a unique client name.
*/
struct StreamOptions {
- RtAudioStreamFlags flags; /*!< A bit-mask of stream flags (RTAUDIO_NONINTERLEAVED, RTAUDIO_MINIMIZE_LATENCY, RTAUDIO_HOG_DEVICE). */
+ RtAudioStreamFlags flags; /*!< A bit-mask of stream flags (RTAUDIO_NONINTERLEAVED, RTAUDIO_MINIMIZE_LATENCY, RTAUDIO_HOG_DEVICE, RTAUDIO_ALSA_USE_DEFAULT). */
unsigned int numberOfBuffers; /*!< Number of stream buffers. */
std::string streamName; /*!< A stream name (currently used only in Jack). */
int priority; /*!< Scheduling priority of callback thread (only used with flag RTAUDIO_SCHEDULE_REALTIME). */
: flags(0), numberOfBuffers(0), priority(0) {}
};
+ //! A static function to determine the current RtAudio version.
+ static std::string getVersion( void ) { return VERSION; }
+
//! A static function to determine the available compiled audio APIs.
/*!
The values returned in the std::vector can be compared against
chosen. If the RTAUDIO_MINIMIZE_LATENCY flag bit is set, the
lowest allowable value is used. The actual value used is
returned via the structure argument. The parameter is API dependent.
+ \param errorCallback A client-defined function that will be invoked
+ when an error has occured.
*/
void openStream( RtAudio::StreamParameters *outputParameters,
RtAudio::StreamParameters *inputParameters,
RtAudioFormat format, unsigned int sampleRate,
unsigned int *bufferFrames, RtAudioCallback callback,
- void *userData = NULL, RtAudio::StreamOptions *options = NULL );
+ void *userData = NULL, RtAudio::StreamOptions *options = NULL, RtAudioErrorCallback errorCallback = NULL );
//! A function that closes a stream and frees any associated stream memory.
/*!
typedef unsigned long ThreadHandle;
typedef CRITICAL_SECTION StreamMutex;
-#elif defined(__LINUX_ALSA__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__)
+#elif defined(__LINUX_ALSA__) || defined(__LINUX_PULSE__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__)
// Using pthread library for various flavors of unix.
#include <pthread.h>
ThreadHandle thread;
void *callback;
void *userData;
+ void *errorCallback;
void *apiInfo; // void pointer for API specific callback information
bool isRunning;
+ bool doRealtime;
+ int priority;
// Default constructor.
CallbackInfo()
- :object(0), callback(0), userData(0), apiInfo(0), isRunning(false) {}
+ :object(0), callback(0), userData(0), errorCallback(0), apiInfo(0), isRunning(false), doRealtime(false) {}
};
// **************************************************************** //
// 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).
+// RtApiJack, RtApiCore, RtApiDs, or RtApiAsio).
//
// **************************************************************** //
+#pragma pack(push, 1)
+class S24 {
+
+ protected:
+ unsigned char c3[3];
+
+ public:
+ S24() {}
+
+ S24& operator = ( const int& i ) {
+ c3[0] = (i & 0x000000ff);
+ c3[1] = (i & 0x0000ff00) >> 8;
+ c3[2] = (i & 0x00ff0000) >> 16;
+ return *this;
+ }
+
+ S24( const S24& v ) { *this = v; }
+ S24( const double& d ) { *this = (int) d; }
+ S24( const float& f ) { *this = (int) f; }
+ S24( const signed short& s ) { *this = (int) s; }
+ S24( const char& c ) { *this = (int) c; }
+
+ int asInt() {
+ int i = c3[0] | (c3[1] << 8) | (c3[2] << 16);
+ if (i & 0x800000) i |= ~0xffffff;
+ return i;
+ }
+};
+#pragma pack(pop)
+
#if defined( HAVE_GETTIMEOFDAY )
#include <sys/time.h>
#endif
RtAudio::StreamParameters *inputParameters,
RtAudioFormat format, unsigned int sampleRate,
unsigned int *bufferFrames, RtAudioCallback callback,
- void *userData, RtAudio::StreamOptions *options );
+ void *userData, RtAudio::StreamOptions *options,
+ RtAudioErrorCallback errorCallback );
virtual void closeStream( void );
virtual void startStream( void ) = 0;
virtual void stopStream( void ) = 0;
long getStreamLatency( void );
unsigned int getStreamSampleRate( void );
virtual double getStreamTime( void );
- bool isStreamOpen( void ) const { return stream_.state != STREAM_CLOSED; };
- bool isStreamRunning( void ) const { return stream_.state == STREAM_RUNNING; };
- void showWarnings( bool value ) { showWarnings_ = value; };
+ bool isStreamOpen( void ) const { return stream_.state != STREAM_CLOSED; }
+ bool isStreamRunning( void ) const { return stream_.state == STREAM_RUNNING; }
+ void showWarnings( bool value ) { showWarnings_ = value; }
protected:
enum StreamState {
STREAM_STOPPED,
+ STREAM_STOPPING,
STREAM_RUNNING,
STREAM_CLOSED = -50
};
:apiHandle(0), deviceBuffer(0) { device[0] = 11111; device[1] = 11111; }
};
+ typedef S24 Int24;
typedef signed short Int16;
typedef signed int Int32;
typedef float Float32;
inline bool RtAudio :: isStreamOpen( void ) const throw() { return rtapi_->isStreamOpen(); }
inline bool RtAudio :: isStreamRunning( void ) const throw() { return rtapi_->isStreamRunning(); }
inline long RtAudio :: getStreamLatency( void ) { return rtapi_->getStreamLatency(); }
-inline unsigned int RtAudio :: getStreamSampleRate( void ) { return rtapi_->getStreamSampleRate(); };
+inline unsigned int RtAudio :: getStreamSampleRate( void ) { return rtapi_->getStreamSampleRate(); }
inline double RtAudio :: getStreamTime( void ) { return rtapi_->getStreamTime(); }
inline void RtAudio :: showWarnings( bool value ) throw() { rtapi_->showWarnings( value ); }
RtApiCore();
~RtApiCore();
- RtAudio::Api getCurrentApi( void ) { return RtAudio::MACOSX_CORE; };
+ RtAudio::Api getCurrentApi( void ) { return RtAudio::MACOSX_CORE; }
unsigned int getDeviceCount( void );
RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
unsigned int getDefaultOutputDevice( void );
RtApiJack();
~RtApiJack();
- RtAudio::Api getCurrentApi( void ) { return RtAudio::UNIX_JACK; };
+ RtAudio::Api getCurrentApi( void ) { return RtAudio::UNIX_JACK; }
unsigned int getDeviceCount( void );
RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
void closeStream( void );
RtApiAsio();
~RtApiAsio();
- RtAudio::Api getCurrentApi( void ) { return RtAudio::WINDOWS_ASIO; };
+ RtAudio::Api getCurrentApi( void ) { return RtAudio::WINDOWS_ASIO; }
unsigned int getDeviceCount( void );
RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
void closeStream( void );
RtApiDs();
~RtApiDs();
- RtAudio::Api getCurrentApi( void ) { return RtAudio::WINDOWS_DS; };
+ RtAudio::Api getCurrentApi( void ) { return RtAudio::WINDOWS_DS; }
unsigned int getDeviceCount( void );
unsigned int getDefaultOutputDevice( void );
unsigned int getDefaultInputDevice( void );
bool coInitialized_;
bool buffersRolling;
long duplexPrerollBytes;
+ std::vector<struct DsDevice> dsDevices;
bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
unsigned int firstChannel, unsigned int sampleRate,
RtAudioFormat format, unsigned int *bufferSize,
RtApiAlsa();
~RtApiAlsa();
- RtAudio::Api getCurrentApi() { return RtAudio::LINUX_ALSA; };
+ RtAudio::Api getCurrentApi() { return RtAudio::LINUX_ALSA; }
unsigned int getDeviceCount( void );
RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
void closeStream( void );
#endif
+#if defined(__LINUX_PULSE__)
+
+class RtApiPulse: public RtApi
+{
+public:
+ ~RtApiPulse();
+ RtAudio::Api getCurrentApi() { return RtAudio::LINUX_PULSE; }
+ unsigned int getDeviceCount( void );
+ RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
+ void closeStream( void );
+ void startStream( void );
+ void stopStream( void );
+ void abortStream( void );
+
+ // 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( void );
+
+ private:
+
+ std::vector<RtAudio::DeviceInfo> devices_;
+ void saveDeviceInfo( void );
+ bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int *bufferSize,
+ RtAudio::StreamOptions *options );
+};
+
+#endif
+
#if defined(__LINUX_OSS__)
class RtApiOss: public RtApi
RtApiOss();
~RtApiOss();
- RtAudio::Api getCurrentApi() { return RtAudio::LINUX_OSS; };
+ RtAudio::Api getCurrentApi() { return RtAudio::LINUX_OSS; }
unsigned int getDeviceCount( void );
RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
void closeStream( void );
{
public:
- RtApiDummy() { errorText_ = "RtApiDummy: This class provides no functionality."; error( RtError::WARNING ); };
- RtAudio::Api getCurrentApi( void ) { return RtAudio::RTAUDIO_DUMMY; };
- unsigned int getDeviceCount( void ) { return 0; };
- RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) { RtAudio::DeviceInfo info; return info; };
- void closeStream( void ) {};
- void startStream( void ) {};
- void stopStream( void ) {};
- void abortStream( void ) {};
+ RtApiDummy() { errorText_ = "RtApiDummy: This class provides no functionality."; error( RtError::WARNING ); }
+ RtAudio::Api getCurrentApi( void ) { return RtAudio::RTAUDIO_DUMMY; }
+ unsigned int getDeviceCount( void ) { return 0; }
+ RtAudio::DeviceInfo getDeviceInfo( unsigned int /*device*/ ) { RtAudio::DeviceInfo info; return info; }
+ void closeStream( void ) {}
+ void startStream( void ) {}
+ void stopStream( void ) {}
+ void abortStream( void ) {}
private:
- bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
- unsigned int firstChannel, unsigned int sampleRate,
- RtAudioFormat format, unsigned int *bufferSize,
- RtAudio::StreamOptions *options ) { return false; };
+ bool probeDeviceOpen( unsigned int /*device*/, StreamMode /*mode*/, unsigned int /*channels*/,
+ unsigned int /*firstChannel*/, unsigned int /*sampleRate*/,
+ RtAudioFormat /*format*/, unsigned int * /*bufferSize*/,
+ RtAudio::StreamOptions * /*options*/ ) { return false; }
};
#endif
projects / rtaudio-cdist.git / blobdiff