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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>
<A HREF="#Q005">Q5 : I'm using libsamplerate but the high quality settings
sound worse than the SRC_LINEAR converter. Why?</A><BR><BR>
</P>
<HR>
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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 normalize 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>
libsamplerate 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 frequencies above 0.5 times the minimum of
the source and destination sample rate (call this fshmin).
Since the filter needed to achieve full attenuation 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>
<!-- ========================================================================= -->
<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>
<!-- ========================================================================= -->
<A NAME="Q005"></A>
<H2><BR><B>Q5 : I'm using libsamplerate but the high quality settings
sound worse than the SRC_LINEAR converter. Why?</B></H2>
<P>
There are two possible problems.
Firstly, if you are using the src_simple() function on successive blocks
of a stream of samples, you will get bad results. The src_simple() function
is designed for use on a whole sound file, all at once, not on contiguous
segments of the same sound file.
To fix the problem, you need to move to the src_process() API or the callback
based API.
</P>
<P>
If you are already using the src_process() API or the callback based API and
the high quality settings sound worse than SRC_LINEAR, then you have other
problems.
Read on for more debugging hints.
</P>
<P>
All of the higher quality converters need to keep state while doing conversions
on segments of a large chunk of audio.
This state information is kept inside the private data pointed to by the
SRC_STATE pointer returned by the src_new() function.
This means, that when you want to start doing sample rate conversion on a
stream of data, you should call src_new() to get a new SRC_STATE pointer
(or alternatively, call src_reset() on an existing SRC_STATE pointer).
You should then pass this SRC_STATE pointer to the src_process() function
with each new block of audio data.
When you have completed the conversion, you can then call src_delete() on
the SRC_STATE pointer.
</P>
<P>
If you are doing all of the above correctly, you need to examine your usage
of the values passed to src_process() in the
<A HREF="api_misc.html#SRC_DATA">SRC_DATA</A>
struct.
Specifically:
</P>
<UL>
<LI> Check that input_frames and output_frames fields are being set in
terms of frames (number of sample values times channels) instead
of just the number of samples.
<LI> Check that you are using the return values input_frames_used and
output_frames_gen to update your source and destination pointers
correctly.
<LI> Check that you are updating the data_in and data_out pointers
correctly for each successive call.
</UL>
<P>
While doing the above, it is probably useful to compare what you are doing to
what is done in the example programs in the examples/ directory of the source
code tarball.
</P>
<P>
If you have done all of the above and are still having problems then its
probably time to email the author with the smallest chunk of code that
adequately demonstrates your problem.
This chunk should not need to be any more than 100 lines of code.
</P>
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