|Exception handling class for RtAudio. More...|
|Realtime audio i/o C++ classes. More...|
|The public device information structure for returning queried values. More...|
|The structure for specifying input or ouput stream parameters. More...|
|The structure for specifying stream options. More...|
|typedef unsigned long||RtAudioFormat|
|RtAudio data format type. More...|
|typedef unsigned int||RtAudioStreamFlags|
|RtAudio stream option flags. More...|
|typedef unsigned int||RtAudioStreamStatus|
|RtAudio stream status (over- or underflow) flags. More...|
|typedef int(*||RtAudioCallback )(void *outputBuffer, void *inputBuffer, unsigned int nFrames, double streamTime, RtAudioStreamStatus status, void *userData)|
|RtAudio callback function prototype. More...|
|typedef void(*||RtAudioErrorCallback )(RtAudioError::Type type, const std::string &errorText)|
|RtAudio error callback function prototype. More...|
|typedef unsigned long RtAudioFormat|
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.
|typedef unsigned long RtAudioStreamFlags|
RtAudio stream option flags.
The following flags can be OR'ed together to allow a client to make changes to the default stream behavior:
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
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
nFrames (assuming the
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 DirectSound, 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 long RtAudioStreamStatus|
RtAudio stream status (over- or underflow) flags.
Notification of a stream over- or underflow is indicated by a non-zero stream
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:
|typedef int(* RtAudioCallback)(void *outputBuffer, void *inputBuffer, unsigned int nFrames, double streamTime, RtAudioStreamStatus status, void *userData)|
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.
|outputBuffer||For output (or duplex) streams, the client should write |
|inputBuffer||For input (or duplex) streams, this buffer will hold |
|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.|
|streamTime||The number of seconds that have elapsed since the stream was started.|
|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.|
|userData||A pointer to optional data provided by the client when opening the stream (default = NULL).|
|©2001-2019 Gary P. Scavone, McGill University. All Rights Reserved.|
Maintained by Gary P. Scavone.