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<div class="refentry" title="srconv">
<a id="srconv"></a>
<div class="titlepage"></div>
<div class="refnamediv">
<h2>
<span class="refentrytitle">srconv</span>
</h2>
<p>srconv —
Converts the sample rate of an audio file.
<a id="IndexSrconv" class="indexterm"></a>
</p>
</div>
<div class="refsect1" title="Description">
<a id="idp81732768"></a>
<h2>Description</h2>
<p>
Converts the sample rate of an audio file at sample rate Rin to
a sample rate of Rout. Optionally the ratio (Rin / Rout) may be
linearly time-varying according to a set of (time, ratio) pairs in
an auxiliary file.
</p>
</div>
<div class="refsect1" title="Syntax">
<a id="idp81733608"></a>
<h2>Syntax</h2>
<pre class="synopsis"><span class="command"><strong>srconv</strong></span> [flags] infile</pre>
</div>
<div class="refsect1" title="Initialization">
<a id="idp81743840"></a>
<h2>Initialization</h2>
<p>
Flags:
</p>
<div class="itemizedlist">
<ul class="itemizedlist" type="disc">
<li class="listitem">
<p>-<span class="emphasis"><em>P num</em></span> = pitch
transposition ratio (srate / r) [don't specify both P and
r]</p>
</li>
<li class="listitem">
<p>-<span class="emphasis"><em>P num</em></span> = pitch
transposition ratio (srate / r) [don't specify both P and
r]</p>
</li>
<li class="listitem">
<p>-<span class="emphasis"><em>Q num</em></span> =quality factor
(1, 2, 3, or 4: default = 2)</p>
</li>
<li class="listitem">
<p>-<span class="emphasis"><em>i filnam</em></span> = auxiliary
breakpoints file (no breakpoint by default. i.e. No ratio change)
</p>
</li>
<li class="listitem">
<p>-<span class="emphasis"><em>r num</em></span> = output sample
rate (must be specified)</p>
</li>
<li class="listitem">
<p>-<span class="emphasis"><em>o fnam</em></span> = sound output
filename</p>
</li>
<li class="listitem">
<p>-<span class="emphasis"><em>A</em></span> = create an AIFF
format output soundfile</p>
</li>
<li class="listitem">
<p>-<span class="emphasis"><em>J</em></span> = create an IRCAM
format output soundfile</p>
</li>
<li class="listitem">
<p>-<span class="emphasis"><em>W</em></span> = create a WAV format
output soundfile</p>
</li>
<li class="listitem">
<p>-<span class="emphasis"><em>h</em></span> = no header on output
soundfile</p>
</li>
<li class="listitem">
<p>-<span class="emphasis"><em>c</em></span> = 8-bit signed_char
sound samples</p>
</li>
<li class="listitem">
<p>-<span class="emphasis"><em>a</em></span> = alaw sound
samples</p>
</li>
<li class="listitem">
<p>-<span class="emphasis"><em>8</em></span> = 8-bit unsigned_char
sound samples</p>
</li>
<li class="listitem">
<p>-<span class="emphasis"><em>u</em></span> = ulaw sound
samples</p>
</li>
<li class="listitem">
<p>-<span class="emphasis"><em>s</em></span> = short_int sound
samples</p>
</li>
<li class="listitem">
<p>-<span class="emphasis"><em>l</em></span> = long_int sound
samples</p>
</li>
<li class="listitem">
<p>-<span class="emphasis"><em>f</em></span> = float sound
samples</p>
</li>
<li class="listitem">
<p>-<span class="emphasis"><em>r N</em></span> = orchestra srate
override</p>
</li>
<li class="listitem">
<p>-<span class="emphasis"><em>K</em></span> = Do not generate PEAK
chunks</p>
</li>
<li class="listitem">
<p>-<span class="emphasis"><em>R</em></span> = continually rewrite
header while writing soundfile (WAV/AIFF)</p>
</li>
<li class="listitem">
<p>-<span class="emphasis"><em>H#</em></span> = print a heartbeat
style 1, 2 or 3 at each soundfile write</p>
</li>
<li class="listitem">
<p>-<span class="emphasis"><em>N</em></span> = notify (ring the
bell) when score or miditrack is done</p>
</li>
<li class="listitem">
<p>-<span class="emphasis"><em>- fnam</em></span> = log output to
file</p>
</li>
</ul>
</div>
<p>
</p>
<p>
This program performs arbitrary sample-rate conversion with high
fidelity. The method is to step through the input at the
desired sampling increment, and to compute the output points as
appropriately weighted averages of the surrounding input points.
There are two cases to consider:
</p>
<div class="orderedlist">
<ol class="orderedlist" type="1">
<li class="listitem">
<p>sample rates are in a small-integer ratio -
weights are obtained from table.</p>
</li>
<li class="listitem">
<p>sample rates are in a large-integer ratio -
weights are linearly interpolated from
table.</p>
</li>
</ol>
</div>
<p>
</p>
<p>
Calculate increment: if decimating, then window is impulse
response of low-pass filter with cutoff frequency at half of
output sample rate; if interpolating, then window is impulse
response of lowpass filter with cutoff frequency at half of input
sample rate.
</p>
</div>
<div class="refsect1" title="Credits">
<a id="idp81833904"></a>
<h2>Credits</h2>
<p>Author: Mark Dolson</p>
<p>August 26, 1989</p>
<p>Author: John ffitch</p>
<p>December 30, 2000</p>
</div>
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distrib
>
Mageia
>
5
>
x86_64
>
media
>
core-release
>
by-pkgid
>
b707d9a4ee443103660a75ccb6e51334
>
files
>
2639
