|
- /************************************************************************/
- /*! \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 and ASIO) operating systems.
-
- RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/
-
- RtAudio: realtime audio i/o C++ classes
- 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
- (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
-
- #include <string>
- #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.
-
- 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).
-
- 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.
- */
- 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.
-
- 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 );
-
- //! RtAudio error callback function prototype.
- /*!
- \param type Type of error.
- \param errorText Error description.
- */
- typedef void (*RtAudioErrorCallback)( RtError::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
- {
- 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_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<unsigned int> sampleRates; /*!< Supported sample rates (queried from list of standard rates). */
- RtAudioFormat nativeFormats; /*!< Bit mask of supported data formats. */
-
- // Default constructor.
- DeviceInfo()
- :probed(false), outputChannels(0), inputChannels(0), duplexChannels(0),
- isDefaultOutput(false), isDefaultInput(false), 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 ) { return VERSION; }
-
- //! 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<RtAudio::Api> &apis ) throw();
-
- //! The class constructor.
- /*!
- The constructor performs minor initialization tasks. No exceptions
- can be thrown.
-
- 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 ) throw();
-
- //! The destructor.
- /*!
- If a stream is running or open, it will be stopped and closed
- automatically.
- */
- ~RtAudio() throw();
-
- //! Returns the audio API specifier for the current instance of RtAudio.
- RtAudio::Api getCurrentApi( void ) throw();
-
- //! 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 ) throw();
-
- //! 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 RtError (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 ) throw();
-
- //! 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 ) throw();
-
- //! A public function for opening a stream with the specified parameters.
- /*!
- An RtError (type = SYSTEM_ERROR) is thrown if a stream cannot be
- opened with the specified parameters or an error occurs during
- processing. An RtError (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 ) throw();
-
- //! A function that starts a stream.
- /*!
- An RtError (type = SYSTEM_ERROR) is thrown if an error occurs
- during processing. An RtError (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 RtError (type = SYSTEM_ERROR) is thrown if an error occurs
- during processing. An RtError (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 RtError (type = SYSTEM_ERROR) is thrown if an error occurs
- during processing. An RtError (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 throw();
-
- //! Returns true if the stream is running and false if it is stopped or not open.
- bool isStreamRunning( void ) const throw();
-
- //! Returns the number of elapsed seconds since the stream was started.
- /*!
- If a stream is not open, an RtError (type = INVALID_USE) will be thrown.
- */
- double getStreamTime( void );
-
- //! 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
- RtError (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 RtError (type = INVALID_USE) will be thrown.
- */
- unsigned int getStreamSampleRate( void );
-
- //! Specify whether warning messages should be printed to stderr.
- void showWarnings( bool value = true ) throw();
-
- protected:
-
- void openRtApi( RtAudio::Api api );
- RtApi *rtapi_;
- };
-
- // Operating system dependent thread functionality.
- #if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__)
- #include <winsock2.h>
- #include <windows.h>
- #include <process.h>
-
- typedef unsigned long 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 <pthread.h>
-
- 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) {}
- };
-
- // **************************************************************** //
- //
- // 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 <sys/time.h>
- #endif
-
- #include <sstream>
-
- class 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 );
- 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<int> inOffset;
- std::vector<int> 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_;
-
- /*!
- 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 RtError (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( RtError::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 ) throw() { return rtapi_->getCurrentApi(); }
- inline unsigned int RtAudio :: getDeviceCount( void ) throw() { return rtapi_->getDeviceCount(); }
- inline RtAudio::DeviceInfo RtAudio :: getDeviceInfo( unsigned int device ) { return rtapi_->getDeviceInfo( device ); }
- inline unsigned int RtAudio :: getDefaultInputDevice( void ) throw() { return rtapi_->getDefaultInputDevice(); }
- inline unsigned int RtAudio :: getDefaultOutputDevice( void ) throw() { return rtapi_->getDefaultOutputDevice(); }
- inline void RtAudio :: closeStream( void ) throw() { 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 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 double RtAudio :: getStreamTime( void ) { return rtapi_->getStreamTime(); }
- inline void RtAudio :: showWarnings( bool value ) throw() { rtapi_->showWarnings( value ); }
-
- // RtApi Subclass prototypes.
-
- #if defined(__MACOSX_CORE__)
-
- #include <CoreAudio/AudioHardware.h>
-
- 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 );
- };
-
- #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<RtAudio::DeviceInfo> 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<struct DsDevice> 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(__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<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_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
- {
- 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( 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; }
- };
-
- #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
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