X-Git-Url: https://git.carlh.net/gitweb/?a=blobdiff_plain;f=RtAudio.h;h=5c5fd6f9a4f34af46d23e8bd8d593bd10bc27e38;hb=f44483cc73ce2a36229c3f904c4230243a181e51;hp=5e0666d7ae143cd395cd36146bec814fbe23ddd2;hpb=637f7dc5e805bad55dba7fb7693a45f5992b2ae7;p=rtaudio.git diff --git a/RtAudio.h b/RtAudio.h index 5e0666d..5c5fd6f 100644 --- a/RtAudio.h +++ b/RtAudio.h @@ -4,13 +4,13 @@ RtAudio provides a common API (Application Programming Interface) for realtime audio input/output across Linux (native ALSA, Jack, - and OSS), SGI, Macintosh OS X (CoreAudio and Jack), and Windows + 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) 2001-2007 Gary P. Scavone + Copyright (c) 2001-2009 Gary P. Scavone Permission is hereby granted, free of charge, to any person obtaining a copy of this software and associated documentation files @@ -42,7 +42,7 @@ \file RtAudio.h */ -// RtAudio: Version 4.0.4pre +// RtAudio: Version 4.0.5 #ifndef __RTAUDIO_H #define __RTAUDIO_H @@ -108,11 +108,15 @@ static const RtAudioFormat RTAUDIO_FLOAT64 = 0x20; // Normalized between plus/mi 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. */ 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. /*! \typedef typedef unsigned long RtAudioStreamStatus; \brief RtAudio stream status (over- or underflow) flags. @@ -240,9 +244,10 @@ class RtAudio 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_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. By default, RtAudio streams pass and receive audio data from the client in an interleaved format. By passing the @@ -268,6 +273,11 @@ class RtAudio 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. + 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 @@ -285,10 +295,11 @@ class RtAudio RtAudioStreamFlags flags; /*!< A bit-mask of stream flags (RTAUDIO_NONINTERLEAVED, RTAUDIO_MINIMIZE_LATENCY, RTAUDIO_HOG_DEVICE). */ 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) {} + : flags(0), numberOfBuffers(0), priority(0) {} }; //! A static function to determine the available compiled audio APIs. @@ -440,10 +451,10 @@ class RtAudio void abortStream( void ); //! Returns true if a stream is open and false if not. - bool isStreamOpen( void ) throw(); + bool isStreamOpen( void ) const throw(); //! Returns true if the stream is running and false if it is stopped or not open. - bool isStreamRunning( void ) throw(); + bool isStreamRunning( void ) const throw(); //! Returns the number of elapsed seconds since the stream was started. /*! @@ -462,6 +473,14 @@ class RtAudio */ 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(); @@ -551,9 +570,10 @@ public: virtual void stopStream( void ) = 0; virtual void abortStream( void ) = 0; long getStreamLatency( void ); + unsigned int getStreamSampleRate( void ); virtual double getStreamTime( void ); - bool isStreamOpen( void ) { return stream_.state != STREAM_CLOSED; }; - bool isStreamRunning( void ) { return stream_.state == STREAM_RUNNING; }; + bool isStreamOpen( void ) const { return stream_.state != STREAM_CLOSED; }; + bool isStreamRunning( void ) const { return stream_.state == STREAM_RUNNING; }; void showWarnings( bool value ) { showWarnings_ = value; }; @@ -617,7 +637,7 @@ protected: #endif RtApiStream() - :apiHandle(0), deviceBuffer(0) {} + :apiHandle(0), deviceBuffer(0) { device[0] = 11111; device[1] = 11111; } }; typedef signed short Int16; @@ -688,9 +708,10 @@ 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 ) throw() { return rtapi_->isStreamOpen(); } -inline bool RtAudio :: isStreamRunning( void ) throw() { return rtapi_->isStreamRunning(); } +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 ); } @@ -867,6 +888,8 @@ public: 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,