mirror of
https://github.com/thestk/stk
synced 2026-04-20 14:36:55 +00:00
Release 4.4.3 tarball
This commit is contained in:
committed by
Stephen Sinclair
parent
cfdfe7736a
commit
f13d5bb3cd
@@ -9,7 +9,7 @@
|
||||
<a class="qindex" href="index.html">Home</a> <a class="qindex" href="information.html">Information</a> <a class="qindex" href="classes.html">Classes</a> <a class="qindex" href="download.html">Download</a> <a class="qindex" href="usage.html">Usage</a> <a class="qindex" href="maillist.html">Mail List</a> <a class="qindex" href="system.html">Requirements</a> <a class="qindex" href="links.html">Links</a> <a class="qindex" href="faq.html">FAQ</a> <a class="qindex" href="tutorial.html">Tutorial</a></CENTER>
|
||||
<HR>
|
||||
<!-- Generated by Doxygen 1.6.2 -->
|
||||
<div class="navpath"><a class="el" href="dir_ca1e4533604ab7cb0cdaaff730a9c38f.html">include</a>
|
||||
<div class="navpath"><a class="el" href="dir_f14fd23bc74c76f288031ad23b3f3505.html">include</a>
|
||||
</div>
|
||||
<div class="contents">
|
||||
<h1>RtAudio.h File Reference</h1>
|
||||
@@ -48,11 +48,11 @@
|
||||
<div class="memdoc">
|
||||
|
||||
<p><a class="el" href="classRtAudio.html" title="Realtime audio i/o C++ classes.">RtAudio</a> data format type. </p>
|
||||
<p>Support for signed integers and floats. Audio data fed to/from an <a class="el" href="classRtAudio.html" title="Realtime audio i/o C++ classes.">RtAudio</a> 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.</p>
|
||||
<p>Support for signed integers and floats. Audio data fed to/from an <a class="el" href="classRtAudio.html" title="Realtime audio i/o C++ classes.">RtAudio</a> 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. Note that 24-bit data is expected to be encapsulated in a 32-bit format.</p>
|
||||
<ul>
|
||||
<li><em>RTAUDIO_SINT8:</em> 8-bit signed integer.</li>
|
||||
<li><em>RTAUDIO_SINT16:</em> 16-bit signed integer.</li>
|
||||
<li><em>RTAUDIO_SINT24:</em> Upper 3 bytes of 32-bit signed integer.</li>
|
||||
<li><em>RTAUDIO_SINT24:</em> Lower 3 bytes of 32-bit signed integer.</li>
|
||||
<li><em>RTAUDIO_SINT32:</em> 32-bit signed integer.</li>
|
||||
<li><em>RTAUDIO_FLOAT32:</em> Normalized between plus/minus 1.0.</li>
|
||||
<li><em>RTAUDIO_FLOAT64:</em> Normalized between plus/minus 1.0. </li>
|
||||
@@ -77,11 +77,13 @@
|
||||
<li><em>RTAUDIO_NONINTERLEAVED:</em> Use non-interleaved buffers (default = interleaved).</li>
|
||||
<li><em>RTAUDIO_MINIMIZE_LATENCY:</em> Attempt to set stream parameters for lowest possible latency.</li>
|
||||
<li><em>RTAUDIO_HOG_DEVICE:</em> Attempt grab device for exclusive use.</li>
|
||||
<li><em>RTAUDIO_ALSA_USE_DEFAULT:</em> Use the "default" PCM device (ALSA only).</li>
|
||||
</ul>
|
||||
<p>By default, <a class="el" href="classRtAudio.html" title="Realtime audio i/o C++ classes.">RtAudio</a> 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 <code>nFrames</code> samples for each channel concatenated back-to-back. For example, the first sample of data for the second channel would be located at index <code>nFrames</code> (assuming the <code>buffer</code> pointer was recast to the correct data type for the stream).</p>
|
||||
<p>Certain audio APIs offer a number of parameters that influence the I/O latency of a stream. By default, <a class="el" href="classRtAudio.html" title="Realtime audio i/o C++ classes.">RtAudio</a> 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.</p>
|
||||
<p>If the RTAUDIO_HOG_DEVICE flag is set, <a class="el" href="classRtAudio.html" title="Realtime audio i/o C++ classes.">RtAudio</a> 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.</p>
|
||||
<p>If the RTAUDIO_SCHEDULE_REALTIME flag is set, <a class="el" href="classRtAudio.html" title="Realtime audio i/o C++ classes.">RtAudio</a> will attempt to select realtime scheduling (round-robin) for the callback thread. </p>
|
||||
<p>If the RTAUDIO_SCHEDULE_REALTIME flag is set, <a class="el" href="classRtAudio.html" title="Realtime audio i/o C++ classes.">RtAudio</a> will attempt to select realtime scheduling (round-robin) for the callback thread.</p>
|
||||
<p>If the RTAUDIO_ALSA_USE_DEFAULT flag is set, <a class="el" href="classRtAudio.html" title="Realtime audio i/o C++ classes.">RtAudio</a> 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. </p>
|
||||
|
||||
</div>
|
||||
</div>
|
||||
@@ -138,7 +140,7 @@
|
||||
|
||||
<table>
|
||||
<tr><td><A HREF="http://ccrma.stanford.edu/software/stk/"><I>The Synthesis ToolKit in C++ (STK)</I></A></td></tr>
|
||||
<tr><td>©1995-2010 Perry R. Cook and Gary P. Scavone. All Rights Reserved.</td></tr>
|
||||
<tr><td>©1995-2011 Perry R. Cook and Gary P. Scavone. All Rights Reserved.</td></tr>
|
||||
</table>
|
||||
|
||||
</BODY>
|
||||
|
||||
Reference in New Issue
Block a user