88 lines
3.9 KiB
C
88 lines
3.9 KiB
C
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/**
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* @author Gary P. SCAVONE
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*
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* @copyright 2001-2013 Gary P. Scavone, all right reserved
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*
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* @license like MIT (see license file)
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*/
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#ifndef __AIRTAUDIO_STREAM_OPTION_H__
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#define __AIRTAUDIO_STREAM_OPTION_H__
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namespace airtaudio {
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/**
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* @brief The structure for specifying stream options.
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*
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* The following flags can be OR'ed together to allow a client to
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* make changes to the default stream behavior:
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*
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* - \e RTAUDIO_NONINTERLEAVED: Use non-interleaved buffers (default = interleaved).
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* - \e RTAUDIO_MINIMIZE_LATENCY: Attempt to set stream parameters for lowest possible latency.
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* - \e RTAUDIO_HOG_DEVICE: Attempt grab device for exclusive use.
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* - \e RTAUDIO_SCHEDULE_REALTIME: Attempt to select realtime scheduling for callback thread.
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* - \e RTAUDIO_ALSA_USE_DEFAULT: Use the "default" PCM device (ALSA only).
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*
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* By default, RtAudio streams pass and receive audio data from the
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* client in an interleaved format. By passing the
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* RTAUDIO_NONINTERLEAVED flag to the openStream() function, audio
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* data will instead be presented in non-interleaved buffers. In
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* this case, each buffer argument in the RtAudioCallback function
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* will point to a single array of data, with \c nFrames samples for
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* each channel concatenated back-to-back. For example, the first
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* sample of data for the second channel would be located at index \c
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* nFrames (assuming the \c buffer pointer was recast to the correct
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* data type for the stream).
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*
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* Certain audio APIs offer a number of parameters that influence the
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* I/O latency of a stream. By default, RtAudio will attempt to set
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* these parameters internally for robust (glitch-free) performance
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* (though some APIs, like Windows Direct Sound, make this difficult).
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* By passing the RTAUDIO_MINIMIZE_LATENCY flag to the openStream()
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* function, internal stream settings will be influenced in an attempt
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* to minimize stream latency, though possibly at the expense of stream
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* performance.
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*
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* If the RTAUDIO_HOG_DEVICE flag is set, RtAudio will attempt to
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* open the input and/or output stream device(s) for exclusive use.
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* Note that this is not possible with all supported audio APIs.
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*
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* If the RTAUDIO_SCHEDULE_REALTIME flag is set, RtAudio will attempt
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* to select realtime scheduling (round-robin) for the callback thread.
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* The \c priority parameter will only be used if the RTAUDIO_SCHEDULE_REALTIME
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* flag is set. It defines the thread's realtime priority.
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*
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* If the RTAUDIO_ALSA_USE_DEFAULT flag is set, RtAudio will attempt to
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* open the "default" PCM device when using the ALSA API. Note that this
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* will override any specified input or output device id.
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*
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* The \c numberOfBuffers parameter can be used to control stream
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* latency in the Windows DirectSound, Linux OSS, and Linux Alsa APIs
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* only. A value of two is usually the smallest allowed. Larger
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* numbers can potentially result in more robust stream performance,
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* though likely at the cost of stream latency. The value set by the
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* user is replaced during execution of the RtAudio::openStream()
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* function by the value actually used by the system.
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*
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* The \c streamName parameter can be used to set the client name
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* when using the Jack API. By default, the client name is set to
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* RtApiJack. However, if you wish to create multiple instances of
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* RtAudio with Jack, each instance must have a unique client name.
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*/
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class StreamOptions {
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public:
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airtaudio::streamFlags flags; //!< A bit-mask of stream flags (RTAUDIO_NONINTERLEAVED, RTAUDIO_MINIMIZE_LATENCY, RTAUDIO_HOG_DEVICE, RTAUDIO_ALSA_USE_DEFAULT).
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uint32_t numberOfBuffers; //!< Number of stream buffers.
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std::string streamName; //!< A stream name (currently used only in Jack).
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int32_t priority; //!< Scheduling priority of callback thread (only used with flag RTAUDIO_SCHEDULE_REALTIME).
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// Default constructor.
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StreamOptions(void) :
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flags(0),
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numberOfBuffers(0),
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priority(0) {}
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};
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};
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#endif
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