From 0876bbd0bad22677b23342c828bc36c3babbed4e Mon Sep 17 00:00:00 2001 From: Gary Scavone Date: Fri, 11 Oct 2013 01:46:50 +0200 Subject: Release 4.0.0 tarball --- doc/html/structRtAudio_1_1StreamOptions.html | 121 +++++++++++++++++++++++++++ 1 file changed, 121 insertions(+) create mode 100644 doc/html/structRtAudio_1_1StreamOptions.html (limited to 'doc/html/structRtAudio_1_1StreamOptions.html') diff --git a/doc/html/structRtAudio_1_1StreamOptions.html b/doc/html/structRtAudio_1_1StreamOptions.html new file mode 100644 index 0000000..244cc66 --- /dev/null +++ b/doc/html/structRtAudio_1_1StreamOptions.html @@ -0,0 +1,121 @@ + + +The RtAudio Home Page + + + + +
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+RtAudio::StreamOptions
+

RtAudio::StreamOptions Struct Reference

The structure for specifying stream options. +More... +

+#include <RtAudio.h> +

+List of all members. + + + + + + + + +

Public Attributes

RtAudioStreamFlags flags
unsigned int numberOfBuffers
std::string streamName
+

Detailed Description

+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:

+

+

+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 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.

+The 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 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. +

+

Member Data Documentation

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RtAudioStreamFlags RtAudio::StreamOptions::flags
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+   + + +

+A bit-mask of stream flags (RTAUDIO_NONINTERLEAVED, RTAUDIO_MINIMIZE_LATENCY, RTAUDIO_HOG_DEVICE).

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unsigned int RtAudio::StreamOptions::numberOfBuffers
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+Number of stream buffers.

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std::string RtAudio::StreamOptions::streamName
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+   + + +

+A stream name (currently used only in Jack).

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The documentation for this struct was generated from the following file: +
+ + + +
©2001-2007 Gary P. Scavone, McGill University. All Rights Reserved.
Maintained by Gary P. Scavone.
+ + + -- cgit v1.2.3