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Secret Rabbit Code (aka libsamplerate)
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Author :<BR>Erik de Castro Lopo
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<H1><B>Frequently Asked Questions</B></H1>
<P>
<A HREF="#Q001">Q1 : Is it normal for the output of libsamplerate to be louder
than its input?</A><BR><BR>
<A HREF="#Q002">Q2 : On Unix/Linux/MacOSX, what is the best way of detecting
the presence and location of libsamplerate and its header file using
autoconf?</A><BR><BR>
<A HREF="#Q003">Q3 : If I upsample and downsample to the original rate, for
example 44.1->96->44.1, do I get an identical signal as the one before the
up/down resampling?</A><BR><BR>
<A HREF="#Q004">Q4 : If I ran src_simple (libsamplerate) on small chunks (160
frames) would that sound bad?</A><BR><BR>
</P>
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<A NAME="Q001"></A>
<H2><BR><B>Q1 : Is it normal for the output of libsamplerate to be louder
than its input?</B></H2>
<P>
The output of libsamplerate will be roughly the same volume as the input.
However, even if the input is strictly in the range (-1.0, 1.0), it is still
possible for the output to contain peak values outside this range.
</P>
<P>
Consider four consecutive samples of [0.5 0.999 0.999 0.5].
If we are up sampling by a factor of two we need to insert samples between
each of the existing samples.
Its pretty obvious then, that the sample between the two 0.999 values should
and will be bigger than 0.999.
</P>
<P>
This means that anyone using libsamplerate should normalise its output before
doing things like saving the audio to a 16 bit WAV file.
</P>
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<a NAME="Q002"></a>
<h2><br><b>Q2 : On Unix/Linux/MacOSX, what is the best way of detecting
the presence and location of libsamplerate and its header file using
autoconf?</b></h2>
<p>
libsamplrate uses the pkg-config (man pkg-config) method of registering itself
with the host system.
The best way of detecting its presence is using something like this in configure.ac
(or configure.in):
</p>
<pre>
PKG_CHECK_MODULES(SAMPLERATE, samplerate >= 0.0.15,
ac_cv_samplerate=1, ac_cv_samplerate=0)
AC_DEFINE_UNQUOTED([HAVE_SAMPLERATE],${ac_cv_samplerate},
[Set to 1 if you have libsamplerate.])
AC_SUBST(SAMPLERATE_CFLAGS)
AC_SUBST(SAMPLERATE_LIBS)
</pre>
<p>
This will automatically set the <b>SAMPLERATE_CFLAGS</b> and <b>SAMPLERATE_LIBS</b>
variables which can be used in Makefile.am or Makefile.in like this:
</p>
<pre>
SAMPLERATE_CFLAGS = @SAMPLERATE_CFLAGS@
SAMPLERATE_LIBS = @SAMPLERATE_LIBS@
</pre>
<p>
If you install libsamplerate from source, you will probably need to set the
<b>PKG_CONFIG_PATH</b> environment variable's suggested at the end of the
libsamplerate configure process. For instance on my system I get this:
</p>
<pre>
-=-=-=-=-=-=-=-=-=-= Configuration Complete =-=-=-=-=-=-=-=-=-=-=-
Configuration summary :
Version : ..................... 0.1.0
Enable debugging : ............ no
Tools :
Compiler is GCC : ............. yes
GCC major version : ........... 3
Extra tools required for testing and examples :
Have FFTW : ................... yes
Have libsndfile : ............. yes
Have libefence : .............. no
Installation directories :
Library directory : ........... /usr/local/lib
Program directory : ........... /usr/local/bin
Pkgconfig directory : ......... /usr/local/lib/pkgconfig
</pre>
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<A NAME="Q003"></A>
<H2><BR><B>Q3 : If I upsample and downsample to the original rate, for
example 44.1->96->44.1, do I get an identical signal as the one before the
up/down resampling?</B></H2>
<P>
The short answer is that for the general case, no, you don't.
The long answer is that for some signals, with some converters, you will
get very, very close.
</P>
<P>
In order to resample correctly (ie using the <B>SRC_SINC_*</B> converters),
filtering needs to be applied, regardless of whether its upsampling or
downsampling.
This filter needs to attenuate all frequencis above 0.5 times the minimum of
the source and desination sample rate (call this fshmin).
Since the filter needed to achieve full attemuation at this point, it has to
start rolling off a some frequency below this point.
It is this rolloff of the very highest frequencies which causes some of the
loss.
</P>
<P>
The other factor is that the filter itself can introduce transient artifacts
which causes the output to be different to the input.
</P>
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<A NAME="Q004"></A>
<H2><BR><B>Q4 : If I ran src_simple on small chunks (say 160 frames) would that
sound bad?</B></H2>
<P>
Well if you are after odd sound effects, it might sound OK.
If you are after high quality sample rate conversion you will be disappointed.
</P>
<P>
The src_simple() was designed to provide a simple to use interface for people
who wanted to do sample rate conversion on say, a whole file all at once.
</P>
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