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<a id="src_conv"></a>
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<div class="refnamediv">
<h2>
<span class="refentrytitle">src_conv</span>
</h2>
<p>src_conv —
Converts the sample rate of an audio file.
<a id="IndexSrc_Conv" class="indexterm"></a>
</p>
</div>
<div class="refsect1">
<a id="idm281472810524792"></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">
<a id="idm281472810481528"></a>
<h2>Syntax</h2>
<pre class="synopsis"><span class="command"><strong>src_conv</strong></span> [flags] infile</pre>
</div>
<div class="refsect1">
<a id="idm281472810479480"></a>
<h2>Initialization</h2>
<p>
Flags:
</p>
<div class="itemizedlist">
<ul class="itemizedlist" style="list-style-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>Q num</em></span> =quality factor
(1, 2, 3, 4 or 5: default = 3)</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 if no P)</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 done</p>
</li>
</ul>
</div>
<p>
</p>
<p>
This program performs arbitrary sample-rate conversion with high
fidelity using the libsamplerate library.
</p>
<p>The five levels of accuracy are:
</p>
<div class="itemizedlist">
<ul class="itemizedlist" style="list-style-type: disc; ">
<li class="listitem"> 1: A linear converter. The quality is poor, but the
conversion speed is blindingly fast. </li>
<li class="listitem"> 2: A Zero Order Hold converter (interpolated value is
equal to the last value). The quality is poor but the conversion
speed is blindingly fast.</li>
<li class="listitem"> 3: This is the fastest bandlimited interpolator and
has an SNR of 97dB and a bandwidth of 80%.</li>
<li class="listitem"> 4: This is another bandlimited interpolator much like
the previous one. It has an SNR of 97dB and a bandwidth of
90%. The speed of the conversion is much faster than the next
one. Blindingly fast.</li>
<li class="listitem"> 5: This is a bandlimited interpolator derived from
the mathematical sinc function and this is the highest quality
sinc based converter, providing a worst case Signal-to-Noise
Ratio (SNR) of 97dB at a bandwidth of 97%.</li>
</ul>
</div>
<p>
</p>
</div>
<div class="refsect1">
<a id="idm281472810452920"></a>
<h2>Credits</h2>
<p>Author: John ffitch</p>
<p>May 2015</p>
</div>
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distrib
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Mageia
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7
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armv7hl
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media
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core-release
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by-pkgid
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5fcfcb7517038d1f44ab4e478e6e61fa
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files
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