| Recently I came up with the following idea:
You get aliasing when doing FM-synthesis with high modulation
index, because the process creates harmonics above the Nyquist
limit which are the 'mirrored' back into the audio band. The
same when doing nonlinear operations that are not bandlimited, as
i.e. an tanh distorter.
Now my kinda coool idea is to turn up the sampling rate to, let's say,
441'000 (441 kHz), the filter all audio signals with a lowpass with a
cutoff of 22kHz, then resample the output to a normal sampling rate
of i.e. 44'100 Hz. I think that this could drastically reduce aliasing
effects.
Now I don't want to have Gigabytes of audio data with Megaherz sampling
rates, so I tried to link the downsampler (I want to use 'sox' for that
task) to csound via a unix-pipe (on a linux system):
csound -d -W -ostdout any.orc any.sco | sox -r441000 -c2 -twav - -r44100 out.wav
First problem is that 'sox' tells me that 441000 is a bogus sampling rate.
This is a minor problem, I think (just edit the source).
But the bigger problem is that csound doesn't like writing to a pipe:
soundfile write returned a bytecount of 3072, not 8192
(disk may be full...
closing the file ...)
0 8192-byte soundblks ....
Of cource, the disk isn't full at all.
I think the problem is, that when writing to a pipe, only so much
can be written as the reader of the pipe (here: 'sox') is reading out.
Any tips ?
Thomas
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To: csound@maths.ex.ac.uk
Subject: Re: oversampling
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Thomas Huber wrote:
> Recently I came up with the following idea:
> You get aliasing when doing FM-synthesis with high modulation
> index, because the process creates harmonics above the Nyquist
> limit which are the 'mirrored' back into the audio band. The
> same when doing nonlinear operations that are not bandlimited, as
> i.e. an tanh distorter.
I use oversampling for a rather long time and it works well (produces much
better quality)
> Now my kinda coool idea is to turn up the sampling rate to, let's say,
> 441'000 (441 kHz), the filter all audio signals with a lowpass with a
> cutoff of 22kHz, then resample the output to a normal sampling rate
> of i.e. 44'100 Hz. I think that this could drastically reduce aliasing
> effects.
depending on the audio signal, I render the files at 176400 Hz (4x),
352800 Hz (8x) or 705600 Hz (16x). The only problem with this is that
some csound opcodes does not like high sample rates (especially
some filters)
> Now I don't want to have Gigabytes of audio data with Megaherz sampling
> rates, so I tried to link the downsampler (I want to use 'sox' for that
> task) to csound via a unix-pipe (on a linux system):
> csound -d -W -ostdout any.orc any.sco | sox -r441000 -c2 -twav - -r44100 out.wav
I did not tried pipes, since I have enough disk space :)
Instead I render to a file which is then resampled by a shell script that
uses csound as a processing engine (sox is rather buggy and unreliable,
and also often produces poor sound quality so I avoid it if possible)
(I can send all shell scripts if anyone is interested)
> First problem is that 'sox' tells me that 441000 is a bogus sampling rate.
> This is a minor problem, I think (just edit the sou
> rce).
> But the bigger problem is that csound doesn't like writing to a pipe:
> soundfile write returned a bytecount of 3072, not 8192
> (disk may be full...
> closing the file ...)
> 0 8192-byte soundblks ....
Try to use headerless (raw) file format. WAV needs seeking which is not supported
by pipes/fifos.
> Of cource, the disk isn't full at all.
> I think the problem is, that when writing to a pipe, only so much
> can be written as the reader of the pipe (here: 'sox') is reading out.
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To: csound@maths.ex.ac.uk
Subject: Re: More bassdrums ...
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> now, this are bassdrums !!!
Well, there are tons of more (>100k), just did not want to spam the list ... :))
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Date: Fri, 24 Sep 1999 17:18:10 +0200
From: Josep M Comajuncosas
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To: Thomas Huber
CC: csound@maths.ex.ac.uk
Subject: Re: Oversampling overkill
References: <199909241455.QAA14135@giger.unibe.ch>
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Hi Thomas,
your idea is used everywhere ... even in the cheap portable CD players. U=
psampling
is a not too demanding operation and it will allow a gentler antialiasin=
g filter
prior to downsampling for example.
But designing such a filter calls for some care, and Csound certainly doe=
s not have
powerful filter design routines by now. Even filter2 is a crap ... you do=
n=B4t have
control over the way the filter structure is implemented (direct form and=
so on...),
and it is a limit of about 40 coefficients I think ... not to say the res=
ult is
truncated to single precission floats.
And I=B4m afraid you won=B4t find too much audio editors that can accept =
sampling rates
much bigger than 96 kHz !
So I would suggest you to take care of the usual FM parameters ... it is =
really
instructive ;-))
Thomas Huber escribi=F3:
> Recently I came up with the following idea:
> You get aliasing when doing FM-synthesis with high modulation
> index, because the process creates harmonics above the Nyquist
> limit which are the 'mirrored' back into the audio band. The
> same when doing nonlinear operations that are not bandlimited, as
> i.e. an tanh distorter.
>
> Now my kinda coool idea is to turn up the sampling rate to, let's say,
> 441'000 (441 kHz), the filter all audio signals with a lowpass with a
> cutoff of 22kHz, then resample the output to a normal sampling rate
> of i.e. 44'100 Hz. I think that this could drastically reduce aliasing
> effects.
>
> Now I don't want to have Gigabytes of audio data with Megaherz sampling
> rates, so I tried to link the downsampler (I want to use 'sox' for that
> task) to csound via a unix-pipe (on a linux system):
>
> csound -d -W -ostdout any.orc any.sco | sox -r441000 -c2 -twav - -r4410=
0 out.wav
>
> First problem is that 'sox' tells me that 441000 is a bogus sampling ra=
te.
> This is a minor problem, I think (just edit the source).
> But the bigger problem is that csound doesn't like writing to a pipe:
>
> soundfile write returned a bytecount of 3072, not 8192
> (disk may be full...
> closing the file ...)
> 0 8192-byte soundblks ....
>
> Of cource, the disk isn't full at all.
> I think the problem is, that when writing to a pipe, only so much
> can be written as the reader of the pipe (here: 'sox') is reading out.
>
> Any tips ?
>
> Thomas
--
Josep M Comajuncosas
C/ Circumval.lacio 75 08790 Gelida - Penedes
Catalunya - SPAIN tel. 93 7792243
e-mail: gelida@intercom.es
ET Informatica de Sistemes
e-mail: jcomajuncosas@campus.uoc.es
http://members.tripod.com/csound/
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To: csound@maths.ex.ac.uk
Subject: Re: oversampling
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Josep M Comajuncosas wrote:
> But designing such a filter calls for some care, and Csound certainly d=
oes not have
> powerful filter design routines by now. Even filter2 is a crap ... you =
don=B4t have
> control over the way the filter structure is implemented (direct form a=
nd so on...),
? and it is a limit of about 40 coefficients I think ... not to say the r=
esult is
> truncated to single precission floats.
What about convolve ?
> And I=B4m afraid you won=B4t find too much audio editors that can accep=
t sampling rates
> much bigger than 96 kHz !
> So I would suggest you to take care of the usual FM parameters ... it i=
s really
> instructive ;-))
the high sample-rate files are not going to be opened in any sound editor=
, since they
will be downsampled by some commandline utility before. Also, I was able =
to open
a 705600 Hz file in Cool Edit (Windows) without problems, I just was not =
able to play
it. On Linux I use a shell script for downsampling, which creates a tempo=
rary orc and
sco file, then runs csound.
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To: csound@maths.ex.ac.uk
Subject: Re: More bassdrums ... (ot)
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Josep M Comajuncosas wrote:
> Please send them to me then... any percussion instrument !!
>
> =3D?ISO-8859-2?Q?Varga_Istv=3DE1n?=3D escribi=F3:
>
> > > now, this are bassdrums !!!
> >
> > Well, there are tons of more (>100k), just did not want to spam the l=
ist ... :))
Hmm, these are techno bassdrums for those annoying 4/4 songs, and =
as far as I know you do not like this sort of music ... (there was a thre=
ad
about it a few weeks ago) but I can send them if you really want
(not sure that they will compile on your machine, as they use soundin
and include files, although it is possible to rewrite them to be "portabl=
e")
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Subject: Re: oversampling
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Now, here are some scripts (incl. downsampling, float -> short w/ dither etc.)
Warning: this is for Linux, will not run on Windows
note: this is originally written for my own usage, so do not expect it to be
portable, well documented, etc. It also assumes that the directories
"/Csound/Analysis", "/Csound/Samples" and "/Csound/Include" exist (SADIR,
SSDIR, include files).
---------------------------------------------------------------------------------------------------------------
begin-base64 644 Csound_scripts-1.1.2.tar.gz
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Message-ID: <001401bf06e6$2ad35180$5199170c@default>
From: Hans Mikelson
To: csound@maths.ex.ac.uk
MMDF-Warning: Parse error in original version of preceding line at UK.AC.Bath.maths.omphalos
Subject: Re: THE BOOK
Date: Fri, 24 Sep 1999 18:40:20 -0500
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Hi,
Latest from the web site is November sometime...
Bye,
Hans Mikelson
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