X-Git-Url: https://main.carlh.net/gitweb/?a=blobdiff_plain;f=RtAudio.h;h=91124cf104e85c3a5e8b9e885d233f30f2210fcd;hb=ee6fd4b4609a1c05a3f4ded4c8022b5ba84d7afd;hp=d9fac56fe60c392f90a9c2878bba737da63cadf4;hpb=8eb71e693530726068addf6b8088aea0fd340f2a;p=rtaudio-cdist.git diff --git a/RtAudio.h b/RtAudio.h index d9fac56..91124cf 100644 --- a/RtAudio.h +++ b/RtAudio.h @@ -1,434 +1,1192 @@ -/******************************************/ -/* - RtAudio - realtime sound I/O C++ class - by Gary P. Scavone, 2001-2002. -*/ -/******************************************/ - -#if !defined(__RTAUDIO_H) -#define __RTAUDIO_H - -#include - -#if defined(__LINUX_ALSA__) - #include - #include - #include - - #define THREAD_TYPE - typedef snd_pcm_t *AUDIO_HANDLE; - typedef int DEVICE_ID; - typedef pthread_t THREAD_HANDLE; - typedef pthread_mutex_t MUTEX; - -#elif defined(__LINUX_OSS__) - #include - #include - - #define THREAD_TYPE - typedef int AUDIO_HANDLE; - typedef int DEVICE_ID; - typedef pthread_t THREAD_HANDLE; - typedef pthread_mutex_t MUTEX; - -#elif defined(__WINDOWS_DS__) - #include - #include - - // 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; - - #define THREAD_TYPE __stdcall - typedef LPGUID DEVICE_ID; - typedef unsigned long THREAD_HANDLE; - typedef CRITICAL_SECTION MUTEX; - -#elif defined(__IRIX_AL__) - #include - #include - #include - - #define THREAD_TYPE - typedef ALport AUDIO_HANDLE; - typedef int DEVICE_ID; - typedef pthread_t THREAD_HANDLE; - typedef pthread_mutex_t MUTEX; - -#endif - - -// *************************************************** // -// -// RtError class declaration. -// -// *************************************************** // - -class RtError -{ -public: - enum TYPE { - WARNING, - DEBUG_WARNING, - UNSPECIFIED, - NO_DEVICES_FOUND, - INVALID_DEVICE, - INVALID_STREAM, - MEMORY_ERROR, - INVALID_PARAMETER, - DRIVER_ERROR, - SYSTEM_ERROR, - THREAD_ERROR - }; - -protected: - char error_message[256]; - TYPE type; - -public: - //! The constructor. - RtError(const char *p, TYPE tipe = RtError::UNSPECIFIED); - - //! The destructor. - virtual ~RtError(void); - - //! Prints "thrown" error message to stdout. - virtual void printMessage(void); - - //! Returns the "thrown" error message TYPE. - virtual const TYPE& getType(void) { return type; } - - //! Returns the "thrown" error message string. - virtual const char *getMessage(void) { return error_message; } -}; - - -// *************************************************** // -// -// RtAudio class declaration. -// -// *************************************************** // - -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; - static const RTAUDIO_FORMAT RTAUDIO_SINT16; - static const RTAUDIO_FORMAT RTAUDIO_SINT24; /*!< Upper 3 bytes of 32-bit integer. */ - static const RTAUDIO_FORMAT RTAUDIO_SINT32; - 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); - - typedef struct { - char name[128]; - DEVICE_ID id[2]; /*!< No value reported by getDeviceInfo(). */ - bool probed; /*!< true if the device capabilities were successfully probed. */ - int maxOutputChannels; - int maxInputChannels; - int maxDuplexChannels; - int minOutputChannels; - int minInputChannels; - int minDuplexChannels; - bool hasDuplexSupport; /*!< true if device supports duplex mode. */ - 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; - - //! The default 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. - */ - RtAudio(); - - //! A constructor which can be used to open a stream during instantiation. - /*! - The specified output and/or input device identifiers correspond - to those enumerated via the getDeviceInfo() method. If device = - 0, the default or first available devices meeting the given - parameters is selected. If an output or input channel value is - zero, the corresponding device value is ignored. When a stream is - successfully opened, its identifier is returned via the "streamId" - pointer. An RtError can be thrown if no devices are found - 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); - - //! The destructor. - /*! - Stops and closes any open streams 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. - - \param outputDevice: If equal to 0, the default or first device - found meeting the given parameters is opened. Otherwise, the - device number should correspond to one of those enumerated via - the getDeviceInfo() method. - \param outputChannels: The desired number of output channels. If - equal to zero, the outputDevice identifier is ignored. - \param inputDevice: If equal to 0, the default or first device - found meeting the given parameters is opened. Otherwise, the - device number should correspond to one of those enumerated via - 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 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 - returned via the same pointer. A value of zero can be specified, - in which case the lowest allowable value is determined. - \param numberOfBuffers: A value which can be used to help control device - latency. More buffers typically result in more robust performance, - 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); - - //! 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. - */ - void setStreamCallback(int streamId, RTAUDIO_CALLBACK callback, void *userData); - - //! A public method which cancels a callback process and function for a given stream. - /*! - This method shuts down a callback process and de-references the - user function for a specific stream. Callback functionality can - subsequently be restarted on the stream via the - setStreamCallback() method. An RtError will be thrown for an - invalid device argument. - */ - void cancelStreamCallback(int streamId); - - //! A public method which returns the number of audio devices found. - int getDeviceCount(void); - - //! Fill a user-supplied RTAUDIO_DEVICE structure for a specified device. - /*! - Any device between 0 and getDeviceCount()-1 is valid. If a - device is busy or otherwise unavailable, the structure member - "probed" has a value of "false". The system default input and - output devices are referenced by device identifier = 0. On - systems which allow dynamic default device settings, the default - devices are not identified by name (specific device enumerations - are assigned device identifiers > 0). An RtError will be - thrown for an invalid device argument. - */ - void getDeviceInfo(int device, RTAUDIO_DEVICE *info); - - //! 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. - */ - char * const getStreamBuffer(int streamId); - - //! 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. - */ - void tickStream(int streamId); - - //! 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). - */ - void closeStream(int streamId); - - //! Public method which starts a stream. - /*! - An RtError will be thrown for an invalid stream identifier - or if a driver error occurs. - */ - void startStream(int streamId); - - //! 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. - */ - void stopStream(int streamId); - - //! 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. - */ - void abortStream(int streamId); - - //! 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); - -protected: - -private: - - static const unsigned int SAMPLE_RATES[MAX_SAMPLE_RATES]; - - enum { FAILURE, SUCCESS }; - - enum STREAM_MODE { - PLAYBACK, - RECORD, - DUPLEX, - UNINITIALIZED = -75 - }; - - enum STREAM_STATE { - STREAM_STOPPED, - STREAM_RUNNING - }; - - typedef struct { - int device[2]; // Playback and record, respectively. - STREAM_MODE mode; // PLAYBACK, RECORD, 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. - bool usingCallback; - THREAD_HANDLE thread; - MUTEX mutex; - RTAUDIO_CALLBACK callback; - void *userData; - } RTAUDIO_STREAM; - - typedef signed short INT16; - typedef signed int INT32; - typedef float FLOAT32; - typedef double FLOAT64; - - char message[256]; - int nDevices; - RTAUDIO_DEVICE *devices; - - std::map streams; - - //! Private error method to allow global control over error handling. - void error(RtError::TYPE type); - - /*! - Private method to count the system audio devices, allocate the - RTAUDIO_DEVICE structures, and probe the device capabilities. - */ - void initialize(void); - - //! Private method to clear an RTAUDIO_DEVICE structure. - void clearDeviceInfo(RTAUDIO_DEVICE *info); - - /*! - 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); - - /*! - 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); - - /*! - 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); - - /*! - 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); - - //! Private method used to perform byte-swapping on buffers. - void byteSwapBuffer(char *buffer, int samples, RTAUDIO_FORMAT format); - - //! Private method which returns the number of bytes for a given format. - int formatBytes(RTAUDIO_FORMAT format); -}; - -// Uncomment the following definition to have extra information spewed to stderr. -//#define RTAUDIO_DEBUG - -#endif +/************************************************************************/ +/*! \class RtAudio + \brief Realtime audio i/o C++ classes. + + RtAudio provides a common API (Application Programming Interface) + for realtime audio input/output across Linux (native ALSA, Jack, + and OSS), Macintosh OS X (CoreAudio and Jack), and Windows + (DirectSound, ASIO and WASAPI) operating systems. + + RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/ + + RtAudio: realtime audio i/o C++ classes + Copyright (c) 2001-2017 Gary P. Scavone + + Permission is hereby granted, free of charge, to any person + obtaining a copy of this software and associated documentation files + (the "Software"), to deal in the Software without restriction, + including without limitation the rights to use, copy, modify, merge, + publish, distribute, sublicense, and/or sell copies of the Software, + and to permit persons to whom the Software is furnished to do so, + subject to the following conditions: + + The above copyright notice and this permission notice shall be + included in all copies or substantial portions of the Software. + + Any person wishing to distribute modifications to the Software is + asked to send the modifications to the original developer so that + they can be incorporated into the canonical version. This is, + however, not a binding provision of this license. + + THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, + EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF + MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. + IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR + ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF + CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION + WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. +*/ +/************************************************************************/ + +/*! + \file RtAudio.h + */ + +#ifndef __RTAUDIO_H +#define __RTAUDIO_H + +#define RTAUDIO_VERSION "5.0.0" + +#if defined _WIN32 || defined __CYGWIN__ + #define RTAUDIO_DLL_PUBLIC +#else + #if __GNUC__ >= 4 + #define RTAUDIO_DLL_PUBLIC __attribute__( (visibility( "default" )) ) + #else + #define RTAUDIO_DLL_PUBLIC + #endif +#endif + +#include +#include +#include +#include + +/*! \typedef typedef unsigned long RtAudioFormat; + \brief RtAudio data format type. + + Support for signed integers and floats. Audio data fed to/from an + RtAudio stream 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. + + - \e RTAUDIO_SINT8: 8-bit signed integer. + - \e RTAUDIO_SINT16: 16-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; // 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. + +/*! \typedef typedef unsigned long RtAudioStreamFlags; + \brief RtAudio stream option flags. + + The following flags can be OR'ed together to allow a client to + make changes to the default stream behavior: + + - \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). + - \e RTAUDIO_JACK_DONT_CONNECT: Do not automatically connect ports (JACK only). + + By default, RtAudio streams pass and receive audio data from the + client in an interleaved format. By passing the + RTAUDIO_NONINTERLEAVED flag to the openStream() function, audio + data will instead be presented in non-interleaved buffers. In + this case, each buffer argument in the RtAudioCallback function + will point to a single array of data, with \c nFrames samples for + each channel concatenated back-to-back. For example, the first + sample of data for the second channel would be located at index \c + nFrames (assuming the \c buffer pointer was recast to the correct + data type for the stream). + + Certain audio APIs offer a number of parameters that influence the + I/O latency of a stream. By default, RtAudio will attempt to set + these parameters internally for robust (glitch-free) performance + (though some APIs, like Windows Direct Sound, make this difficult). + By passing the RTAUDIO_MINIMIZE_LATENCY flag to the openStream() + function, internal stream settings will be influenced in an attempt + to minimize stream latency, though possibly at the expense of stream + performance. + + If the RTAUDIO_HOG_DEVICE flag is set, RtAudio will attempt to + open the input and/or output stream device(s) for exclusive use. + Note that this is not possible with all supported audio APIs. + + 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. + + If the RTAUDIO_JACK_DONT_CONNECT flag is set, RtAudio will not attempt + to automatically connect the ports of the client to the audio device. +*/ +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). +static const RtAudioStreamFlags RTAUDIO_JACK_DONT_CONNECT = 0x20; // Do not automatically connect ports (JACK only). + +/*! \typedef typedef unsigned long RtAudioStreamStatus; + \brief RtAudio stream status (over- or underflow) flags. + + Notification of a stream over- or underflow is indicated by a + non-zero stream \c status argument in the RtAudioCallback function. + The stream status can be one of the following two options, + depending on whether the stream is open for output and/or input: + + - \e RTAUDIO_INPUT_OVERFLOW: Input data was discarded because of an overflow condition at the driver. + - \e RTAUDIO_OUTPUT_UNDERFLOW: The output buffer ran low, likely producing a break in the output sound. +*/ +typedef unsigned int RtAudioStreamStatus; +static const RtAudioStreamStatus RTAUDIO_INPUT_OVERFLOW = 0x1; // Input data was discarded because of an overflow condition at the driver. +static const RtAudioStreamStatus RTAUDIO_OUTPUT_UNDERFLOW = 0x2; // The output buffer ran low, likely causing a gap in the output sound. + +//! RtAudio callback function prototype. +/*! + All RtAudio clients must create a function of type RtAudioCallback + to read and/or write data from/to the audio stream. When the + underlying audio system is ready for new input or output data, this + function will be invoked. + + \param outputBuffer For output (or duplex) streams, the client + should write \c nFrames of audio sample frames into this + buffer. This argument should be recast to the datatype + specified when the stream was opened. For input-only + streams, this argument will be NULL. + + \param inputBuffer For input (or duplex) streams, this buffer will + hold \c nFrames of input audio sample frames. This + argument should be recast to the datatype specified when the + stream was opened. For output-only streams, this argument + will be NULL. + + \param nFrames The number of sample frames of input or output + data in the buffers. The actual buffer size in bytes is + dependent on the data type and number of channels in use. + + \param streamTime The number of seconds that have elapsed since the + stream was started. + + \param status If non-zero, this argument indicates a data overflow + or underflow condition for the stream. The particular + condition can be determined by comparison with the + RtAudioStreamStatus flags. + + \param userData A pointer to optional data provided by the client + when opening the stream (default = NULL). + + To continue normal stream operation, the RtAudioCallback function + should return a value of zero. To stop the stream and drain the + output buffer, the function should return a value of one. To abort + the stream immediately, the client should return a value of two. + */ +typedef int (*RtAudioCallback)( void *outputBuffer, void *inputBuffer, + unsigned int nFrames, + double streamTime, + RtAudioStreamStatus status, + void *userData ); + +/************************************************************************/ +/*! \class RtAudioError + \brief Exception handling class for RtAudio. + + The RtAudioError class is quite simple but it does allow errors to be + "caught" by RtAudioError::Type. See the RtAudio documentation to know + which methods can throw an RtAudioError. +*/ +/************************************************************************/ + +class RTAUDIO_DLL_PUBLIC RtAudioError : public std::runtime_error +{ + public: + //! Defined RtAudioError 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. */ + MEMORY_ERROR, /*!< An error occured during memory allocation. */ + INVALID_PARAMETER, /*!< An invalid parameter was specified to a function. */ + INVALID_USE, /*!< The function was called incorrectly. */ + DRIVER_ERROR, /*!< A system driver error occured. */ + SYSTEM_ERROR, /*!< A system error occured. */ + THREAD_ERROR /*!< A thread error occured. */ + }; + + //! The constructor. + RtAudioError( const std::string& message, + Type type = RtAudioError::UNSPECIFIED ) + : std::runtime_error(message), type_(type) {} + + //! Prints thrown error message to stderr. + virtual void printMessage( void ) const + { std::cerr << '\n' << what() << "\n\n"; } + + //! Returns the thrown error message type. + virtual const Type& getType(void) const { return type_; } + + //! Returns the thrown error message string. + virtual const std::string getMessage(void) const + { return std::string(what()); } + + protected: + Type type_; +}; + +//! RtAudio error callback function prototype. +/*! + \param type Type of error. + \param errorText Error description. + */ +typedef void (*RtAudioErrorCallback)( RtAudioError::Type type, const std::string &errorText ); + +// **************************************************************** // +// +// 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 RtApi and its +// subclasses. RtAudio creates an instance of an RtApi subclass +// based on the user's API choice. If no choice is made, RtAudio +// attempts to make a "logical" API selection. +// +// **************************************************************** // + +class RtApi; + +class RTAUDIO_DLL_PUBLIC RtAudio +{ + public: + + //! Audio API specifier arguments. + 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. */ + WINDOWS_WASAPI, /*!< The Microsoft WASAPI API. */ + WINDOWS_ASIO, /*!< The Steinberg Audio Stream I/O API. */ + WINDOWS_DS, /*!< The Microsoft Direct Sound API. */ + RTAUDIO_DUMMY /*!< A compilable but non-functional API. */ + }; + + //! The public device information structure for returning queried values. + struct DeviceInfo { + bool probed; /*!< true if the device capabilities were successfully probed. */ + std::string name; /*!< Character string device identifier. */ + unsigned int outputChannels; /*!< Maximum output channels supported by device. */ + unsigned int inputChannels; /*!< Maximum input channels supported by device. */ + unsigned int duplexChannels; /*!< Maximum simultaneous input/output channels supported by device. */ + bool isDefaultOutput; /*!< true if this is the default output device. */ + bool isDefaultInput; /*!< true if this is the default input device. */ + std::vector sampleRates; /*!< Supported sample rates (queried from list of standard rates). */ + unsigned int preferredSampleRate; /*!< Preferred sample rate, eg. for WASAPI the system sample rate. */ + RtAudioFormat nativeFormats; /*!< Bit mask of supported data formats. */ + + // Default constructor. + DeviceInfo() + :probed(false), outputChannels(0), inputChannels(0), duplexChannels(0), + isDefaultOutput(false), isDefaultInput(false), preferredSampleRate(0), nativeFormats(0) {} + }; + + //! The structure for specifying input or ouput stream parameters. + struct StreamParameters { + unsigned int deviceId; /*!< Device index (0 to getDeviceCount() - 1). */ + unsigned int nChannels; /*!< Number of channels. */ + unsigned int firstChannel; /*!< First channel index on device (default = 0). */ + + // Default constructor. + StreamParameters() + : deviceId(0), nChannels(0), firstChannel(0) {} + }; + + //! The structure for specifying stream options. + /*! + The following flags can be OR'ed together to allow a client to + make changes to the default stream behavior: + + - \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_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 + RTAUDIO_NONINTERLEAVED flag to the openStream() function, audio + data will instead be presented in non-interleaved buffers. In + this case, each buffer argument in the RtAudioCallback function + will point to a single array of data, with \c nFrames samples for + each channel concatenated back-to-back. For example, the first + sample of data for the second channel would be located at index \c + nFrames (assuming the \c buffer pointer was recast to the correct + data type for the stream). + + Certain audio APIs offer a number of parameters that influence the + I/O latency of a stream. By default, RtAudio will attempt to set + these parameters internally for robust (glitch-free) performance + (though some APIs, like Windows Direct Sound, make this difficult). + By passing the RTAUDIO_MINIMIZE_LATENCY flag to the openStream() + function, internal stream settings will be influenced in an attempt + to minimize stream latency, though possibly at the expense of stream + performance. + + If the RTAUDIO_HOG_DEVICE flag is set, RtAudio will attempt to + open the input and/or output stream device(s) for exclusive use. + Note that this is not possible with all supported audio APIs. + + 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. + + 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 + only. A value of two is usually the smallest allowed. Larger + numbers can potentially result in more robust stream performance, + though likely at the cost of stream latency. The value set by the + user is replaced during execution of the RtAudio::openStream() + function by the value actually used by the system. + + The \c streamName parameter can be used to set the client name + when using the Jack API. By default, the client name is set to + RtApiJack. However, if you wish to create multiple instances of + 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, 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). */ + + // Default constructor. + StreamOptions() + : flags(0), numberOfBuffers(0), priority(0) {} + }; + + //! A static function to determine the current RtAudio version. + static std::string getVersion( void ); + + //! A static function to determine the available compiled audio APIs. + /*! + The values returned in the std::vector can be compared against + the enumerated list values. Note that there can be more than one + API compiled for certain operating systems. + */ + static void getCompiledApi( std::vector &apis ); + + //! Return the name of a specified compiled audio API. + /*! + If the API is unknown or not compiled, this function will return + the empty string. + */ + static const std::string &getCompiledApiName( RtAudio::Api api ); + + //! Return the compiled audio API having the given name. + /*! + A case insensitive comparison will check the specified name + against the list of compiled APIs, and return the one which + matches. On failure, the function returns UNSPECIFIED. + */ + static RtAudio::Api getCompiledApiByName( const std::string &name ); + + //! The class constructor. + /*! + The constructor performs minor initialization tasks. An exception + can be thrown if no API support is compiled. + + 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::Api api=UNSPECIFIED ); + + //! The destructor. + /*! + If a stream is running or open, it will be stopped and closed + automatically. + */ + ~RtAudio(); + + //! Returns the audio API specifier for the current instance of RtAudio. + RtAudio::Api getCurrentApi( void ); + + //! A public function that queries for the number of audio devices available. + /*! + This function performs a system query of available devices each time it + is called, thus supporting devices connected \e after instantiation. If + a system error occurs during processing, a warning will be issued. + */ + unsigned int getDeviceCount( void ); + + //! Return an RtAudio::DeviceInfo structure for a specified device number. + /*! + + Any device integer between 0 and getDeviceCount() - 1 is valid. + If an invalid argument is provided, an RtAudioError (type = INVALID_USE) + will be thrown. If a device is busy or otherwise unavailable, the + structure member "probed" will have a value of "false" and all + other members are undefined. If the specified device is the + current default input or output device, the corresponding + "isDefault" member will have a value of "true". + */ + RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); + + //! A function that returns the index of the default output device. + /*! + If the underlying audio API does not provide a "default + device", or if no devices are available, the return value will be + 0. Note that this is a valid device identifier and it is the + client's responsibility to verify that a device is available + before attempting to open a stream. + */ + unsigned int getDefaultOutputDevice( void ); + + //! A function that returns the index of the default input device. + /*! + If the underlying audio API does not provide a "default + device", or if no devices are available, the return value will be + 0. Note that this is a valid device identifier and it is the + client's responsibility to verify that a device is available + before attempting to open a stream. + */ + unsigned int getDefaultInputDevice( void ); + + //! A public function for opening a stream with the specified parameters. + /*! + An RtAudioError (type = SYSTEM_ERROR) is thrown if a stream cannot be + opened with the specified parameters or an error occurs during + processing. An RtAudioError (type = INVALID_USE) is thrown if any + invalid device ID or channel number parameters are specified. + + \param outputParameters Specifies output stream parameters to use + when opening a stream, including a device ID, number of channels, + and starting channel number. For input-only streams, this + argument should be NULL. The device ID is an index value between + 0 and getDeviceCount() - 1. + \param inputParameters Specifies input stream parameters to use + when opening a stream, including a device ID, number of channels, + and starting channel number. For output-only streams, this + argument should be NULL. The device ID is an index value between + 0 and getDeviceCount() - 1. + \param format An RtAudioFormat specifying the desired sample data format. + \param sampleRate The desired sample rate (sample frames per second). + \param *bufferFrames A pointer to a value indicating the desired + internal buffer size in sample frames. The actual value + used by the device is returned via the same pointer. A + value of zero can be specified, in which case the lowest + allowable value is determined. + \param callback A client-defined function that will be invoked + when input data is available and/or output data is needed. + \param userData An optional pointer to data that can be accessed + from within the callback function. + \param options An optional pointer to a structure containing various + global stream options, including a list of OR'ed RtAudioStreamFlags + and a suggested number of stream buffers that can be used to + control stream latency. More buffers typically result in more + robust performance, though at a cost of greater latency. If a + value of zero is specified, a system-specific median value is + 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, RtAudioErrorCallback errorCallback = NULL ); + + //! A function that closes a stream and frees any associated stream memory. + /*! + If a stream is not open, this function issues a warning and + returns (no exception is thrown). + */ + void closeStream( void ); + + //! A function that starts a stream. + /*! + An RtAudioError (type = SYSTEM_ERROR) is thrown if an error occurs + during processing. An RtAudioError (type = INVALID_USE) is thrown if a + stream is not open. A warning is issued if the stream is already + running. + */ + void startStream( void ); + + //! Stop a stream, allowing any samples remaining in the output queue to be played. + /*! + An RtAudioError (type = SYSTEM_ERROR) is thrown if an error occurs + during processing. An RtAudioError (type = INVALID_USE) is thrown if a + stream is not open. A warning is issued if the stream is already + stopped. + */ + void stopStream( void ); + + //! Stop a stream, discarding any samples remaining in the input/output queue. + /*! + An RtAudioError (type = SYSTEM_ERROR) is thrown if an error occurs + during processing. An RtAudioError (type = INVALID_USE) is thrown if a + stream is not open. A warning is issued if the stream is already + stopped. + */ + void abortStream( void ); + + //! Returns true if a stream is open and false if not. + bool isStreamOpen( void ) const; + + //! Returns true if the stream is running and false if it is stopped or not open. + bool isStreamRunning( void ) const; + + //! Returns the number of elapsed seconds since the stream was started. + /*! + If a stream is not open, an RtAudioError (type = INVALID_USE) will be thrown. + */ + double getStreamTime( void ); + + //! Set the stream time to a time in seconds greater than or equal to 0.0. + /*! + If a stream is not open, an RtAudioError (type = INVALID_USE) will be thrown. + */ + void setStreamTime( double time ); + + //! Returns the internal stream latency in sample frames. + /*! + The stream latency refers to delay in audio input and/or output + caused by internal buffering by the audio system and/or hardware. + For duplex streams, the returned value will represent the sum of + the input and output latencies. If a stream is not open, an + RtAudioError (type = INVALID_USE) will be thrown. If the API does not + report latency, the return value will be zero. + */ + long getStreamLatency( void ); + + //! Returns actual sample rate in use by the stream. + /*! + On some systems, the sample rate used may be slightly different + than that specified in the stream parameters. If a stream is not + open, an RtAudioError (type = INVALID_USE) will be thrown. + */ + unsigned int getStreamSampleRate( void ); + + //! Specify whether warning messages should be printed to stderr. + void showWarnings( bool value = true ); + + protected: + + void openRtApi( RtAudio::Api api ); + RtApi *rtapi_; +}; + +// Operating system dependent thread functionality. +#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) || defined(__WINDOWS_WASAPI__) + + #ifndef NOMINMAX + #define NOMINMAX + #endif + #include + #include + + typedef uintptr_t ThreadHandle; + typedef CRITICAL_SECTION StreamMutex; + +#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 + + typedef pthread_t ThreadHandle; + typedef pthread_mutex_t StreamMutex; + +#else // Setup for "dummy" behavior + + #define __RTAUDIO_DUMMY__ + typedef int ThreadHandle; + typedef int StreamMutex; + +#endif + +// 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; + 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), errorCallback(0), apiInfo(0), isRunning(false), doRealtime(false), priority(0) {} +}; + +// **************************************************************** // +// +// RtApi class declaration. +// +// Subclasses of RtApi contain all API- and OS-specific code necessary +// to fully implement the RtAudio API. +// +// 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, 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 +#endif + +#include + +class RTAUDIO_DLL_PUBLIC RtApi +{ +public: + + RtApi(); + virtual ~RtApi(); + virtual RtAudio::Api getCurrentApi( void ) = 0; + virtual unsigned int getDeviceCount( void ) = 0; + virtual RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) = 0; + virtual unsigned int getDefaultInputDevice( void ); + virtual unsigned int getDefaultOutputDevice( void ); + void openStream( RtAudio::StreamParameters *outputParameters, + RtAudio::StreamParameters *inputParameters, + RtAudioFormat format, unsigned int sampleRate, + unsigned int *bufferFrames, RtAudioCallback callback, + void *userData, RtAudio::StreamOptions *options, + RtAudioErrorCallback errorCallback ); + virtual void closeStream( void ); + virtual void startStream( void ) = 0; + virtual void stopStream( void ) = 0; + virtual void abortStream( void ) = 0; + long getStreamLatency( void ); + unsigned int getStreamSampleRate( void ); + virtual double getStreamTime( void ); + virtual void setStreamTime( double time ); + 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: + + static const unsigned int MAX_SAMPLE_RATES; + static const unsigned int SAMPLE_RATES[]; + + enum { FAILURE, SUCCESS }; + + enum StreamState { + STREAM_STOPPED, + STREAM_STOPPING, + STREAM_RUNNING, + STREAM_CLOSED = -50 + }; + + enum StreamMode { + OUTPUT, + INPUT, + DUPLEX, + UNINITIALIZED = -75 + }; + + // A protected structure used for buffer conversion. + struct ConvertInfo { + int channels; + int inJump, outJump; + RtAudioFormat inFormat, outFormat; + std::vector inOffset; + std::vector outOffset; + }; + + // A protected structure for audio streams. + struct RtApiStream { + unsigned 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, RUNNING, or CLOSED + char *userBuffer[2]; // Playback and record, respectively. + char *deviceBuffer; + bool doConvertBuffer[2]; // Playback and record, respectively. + bool userInterleaved; + bool deviceInterleaved[2]; // Playback and record, respectively. + bool doByteSwap[2]; // Playback and record, respectively. + unsigned int sampleRate; + unsigned int bufferSize; + unsigned int nBuffers; + unsigned int nUserChannels[2]; // Playback and record, respectively. + unsigned int nDeviceChannels[2]; // Playback and record channels, respectively. + unsigned int channelOffset[2]; // Playback and record, respectively. + unsigned long latency[2]; // Playback and record, respectively. + RtAudioFormat userFormat; + RtAudioFormat deviceFormat[2]; // Playback and record, respectively. + StreamMutex mutex; + CallbackInfo callbackInfo; + ConvertInfo convertInfo[2]; + double streamTime; // Number of elapsed seconds since the stream started. + +#if defined(HAVE_GETTIMEOFDAY) + struct timeval lastTickTimestamp; +#endif + + RtApiStream() + :apiHandle(0), deviceBuffer(0) { device[0] = 11111; device[1] = 11111; } + }; + + typedef S24 Int24; + typedef signed short Int16; + typedef signed int Int32; + typedef float Float32; + typedef double Float64; + + std::ostringstream errorStream_; + std::string errorText_; + bool showWarnings_; + RtApiStream stream_; + bool firstErrorOccurred_; + + /*! + Protected, api-specific method that 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. A + successful probe is indicated by a return value of SUCCESS. + */ + virtual bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ); + + //! A protected function used to increment the stream time. + void tickStreamTime( void ); + + //! Protected common method to clear an RtApiStream structure. + void clearStreamInfo(); + + /*! + Protected common method that throws an RtAudioError (type = + INVALID_USE) if a stream is not open. + */ + void verifyStream( void ); + + //! Protected common error method to allow global control over error handling. + void error( RtAudioError::Type type ); + + /*! + Protected method used to perform format, channel number, and/or interleaving + conversions between the user and device buffers. + */ + void convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info ); + + //! Protected common method used to perform byte-swapping on buffers. + void byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format ); + + //! Protected common method that returns the number of bytes for a given format. + unsigned int formatBytes( RtAudioFormat format ); + + //! Protected common method that sets up the parameters for buffer conversion. + void setConvertInfo( StreamMode mode, unsigned int firstChannel ); +}; + +// **************************************************************** // +// +// Inline RtAudio definitions. +// +// **************************************************************** // + +inline RtAudio::Api RtAudio :: getCurrentApi( void ) { return rtapi_->getCurrentApi(); } +inline unsigned int RtAudio :: getDeviceCount( void ) { return rtapi_->getDeviceCount(); } +inline RtAudio::DeviceInfo RtAudio :: getDeviceInfo( unsigned int device ) { return rtapi_->getDeviceInfo( device ); } +inline unsigned int RtAudio :: getDefaultInputDevice( void ) { return rtapi_->getDefaultInputDevice(); } +inline unsigned int RtAudio :: getDefaultOutputDevice( void ) { return rtapi_->getDefaultOutputDevice(); } +inline void RtAudio :: closeStream( void ) { return rtapi_->closeStream(); } +inline void RtAudio :: startStream( void ) { return rtapi_->startStream(); } +inline void RtAudio :: stopStream( void ) { return rtapi_->stopStream(); } +inline void RtAudio :: abortStream( void ) { return rtapi_->abortStream(); } +inline bool RtAudio :: isStreamOpen( void ) const { return rtapi_->isStreamOpen(); } +inline bool RtAudio :: isStreamRunning( void ) const { return rtapi_->isStreamRunning(); } +inline long RtAudio :: getStreamLatency( void ) { return rtapi_->getStreamLatency(); } +inline unsigned int RtAudio :: getStreamSampleRate( void ) { return rtapi_->getStreamSampleRate(); } +inline double RtAudio :: getStreamTime( void ) { return rtapi_->getStreamTime(); } +inline void RtAudio :: setStreamTime( double time ) { return rtapi_->setStreamTime( time ); } +inline void RtAudio :: showWarnings( bool value ) { rtapi_->showWarnings( value ); } + +// RtApi Subclass prototypes. + +#if defined(__MACOSX_CORE__) + +#include + +class RtApiCore: public RtApi +{ +public: + + RtApiCore(); + ~RtApiCore(); + RtAudio::Api getCurrentApi( void ) { return RtAudio::MACOSX_CORE; } + unsigned int getDeviceCount( void ); + RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); + unsigned int getDefaultOutputDevice( void ); + unsigned int getDefaultInputDevice( void ); + void closeStream( void ); + void startStream( void ); + void stopStream( void ); + void abortStream( void ); + long getStreamLatency( 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! + bool callbackEvent( AudioDeviceID deviceId, + const AudioBufferList *inBufferList, + const AudioBufferList *outBufferList ); + + 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 ); + static const char* getErrorCode( OSStatus code ); +}; + +#endif + +#if defined(__UNIX_JACK__) + +class RtApiJack: public RtApi +{ +public: + + RtApiJack(); + ~RtApiJack(); + RtAudio::Api getCurrentApi( void ) { return RtAudio::UNIX_JACK; } + unsigned int getDeviceCount( void ); + RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); + void closeStream( void ); + void startStream( void ); + void stopStream( void ); + void abortStream( void ); + long getStreamLatency( 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! + bool callbackEvent( unsigned long nframes ); + + 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 ); + + bool shouldAutoconnect_; +}; + +#endif + +#if defined(__WINDOWS_ASIO__) + +class RtApiAsio: public RtApi +{ +public: + + RtApiAsio(); + ~RtApiAsio(); + RtAudio::Api getCurrentApi( void ) { return RtAudio::WINDOWS_ASIO; } + unsigned int getDeviceCount( void ); + RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); + void closeStream( void ); + void startStream( void ); + void stopStream( void ); + void abortStream( void ); + long getStreamLatency( 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! + bool callbackEvent( long bufferIndex ); + + private: + + std::vector devices_; + void saveDeviceInfo( void ); + bool coInitialized_; + 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(__WINDOWS_DS__) + +class RtApiDs: public RtApi +{ +public: + + RtApiDs(); + ~RtApiDs(); + RtAudio::Api getCurrentApi( void ) { return RtAudio::WINDOWS_DS; } + unsigned int getDeviceCount( void ); + unsigned int getDefaultOutputDevice( void ); + unsigned int getDefaultInputDevice( void ); + RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); + void closeStream( void ); + void startStream( void ); + void stopStream( void ); + void abortStream( void ); + long getStreamLatency( 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: + + bool coInitialized_; + bool buffersRolling; + long duplexPrerollBytes; + std::vector dsDevices; + 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(__WINDOWS_WASAPI__) + +struct IMMDeviceEnumerator; + +class RtApiWasapi : public RtApi +{ +public: + RtApiWasapi(); + ~RtApiWasapi(); + + RtAudio::Api getCurrentApi( void ) { return RtAudio::WINDOWS_WASAPI; } + unsigned int getDeviceCount( void ); + RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); + unsigned int getDefaultOutputDevice( void ); + unsigned int getDefaultInputDevice( void ); + void closeStream( void ); + void startStream( void ); + void stopStream( void ); + void abortStream( void ); + +private: + bool coInitialized_; + IMMDeviceEnumerator* deviceEnumerator_; + + bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int* bufferSize, + RtAudio::StreamOptions* options ); + + static DWORD WINAPI runWasapiThread( void* wasapiPtr ); + static DWORD WINAPI stopWasapiThread( void* wasapiPtr ); + static DWORD WINAPI abortWasapiThread( void* wasapiPtr ); + void wasapiThread(); +}; + +#endif + +#if defined(__LINUX_ALSA__) + +class RtApiAlsa: public RtApi +{ +public: + + RtApiAlsa(); + ~RtApiAlsa(); + RtAudio::Api getCurrentApi() { return RtAudio::LINUX_ALSA; } + 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 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_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 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 +{ +public: + + RtApiOss(); + ~RtApiOss(); + RtAudio::Api getCurrentApi() { return RtAudio::LINUX_OSS; } + 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: + + 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(__RTAUDIO_DUMMY__) + +class RtApiDummy: public RtApi +{ +public: + + RtApiDummy() { errorText_ = "RtApiDummy: This class provides no functionality."; error( RtAudioError::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; } +}; + +#endif + +#endif + +// Indentation settings for Vim and Emacs +// +// Local Variables: +// c-basic-offset: 2 +// indent-tabs-mode: nil +// End: +// +// vim: et sts=2 sw=2