Hi Gary,
Yeah, the audio fundamental guide mentions the Nyquist rate, that was one term
I wasn’t aware of before then, but makes sense that Nyquist was named for that
concept, which itself, by the way, was named after Harry Theodore Nyquist
(1889-1976). Yes, inventors do fascinate me.
So my theory about actual sample data being between -32768 and 32767 and that
data being converted in a uniform value was correct then. I guess that makes
sense when we’re talking about files with different sample rates, bit depths
etc.
The way I understand it, volume is synonymous with amplitude. Is that right? If
so then I would assume that -1 would be equal to the minimum possible volume
(I.E. -160DB or whatever it is), and 1 represents the maximum (I.E. 0DB)?
In relation to adding samples, you say that, “if sample 47 of wave 1 is .3 and
that of sound 2 is .4, sample 47 of the new sound is .7.” If that’s the case,
my understanding is, since each sample is an amplitude (or volume), that would
change sample 47 from -56DB to –24DB (That is assuming we are working with a
-160DB minimum level).
Like I say, I need to know how these samples work before I can start to touch
Nyquist, otherwise I fear I’m going to be frazzled with information that I
don’t understand.
Cheers.
Damien.
From: Gary Campbell
Sent: Thursday, August 17, 2017 5:44 PM
To: audacity4blind@xxxxxxxxxxxxx
Subject: [audacity4blind] Re: Manuals and tutorials (Re: nyquist little manual)
Hi Damien,
It's been a while since I've used Nyquist, but I learned what I know about
Audacity and Nyquist from the Nyquist manual. On the wiki there is a Nyquist
Plug-ins Reference which has links to the Nyquist manual.
An audio sample in a 16-bit signed WAV file is a number between -32768 and
32767. In a 24-bit file it would be different. Nyquist abstracts this out as
a number between -1 and 1 so your program is not dependent on the way each
sound is stored. If you display a sound as a graph of voltage versus time,
each sample is the value of the voltage at that point in time. So in Nyquist a
sample is a number between -1 and 1 that is the amplitude of the sound at that
point in time. If you were to examine samples of a sine wave you would find
that they would start at 0, increase for a while until they reach a maximum,
then decrease through 0 to minus the maximum, and then increase to the maximum
again. The samples between the two maximum points would represent one cycle,
the amount of time that represents is the "period" (sec/cycle) of the wave, and
the frequency of that sound is 1 over the period. If you sum two sounds in
Nyquist, you make a new sound by adding the values of sample 1 of both sounds,
then sample 2, etc. So if sample 47 of wave 1 is .3 and that of sound 2 is .4,
sample 47 of the new sound is .7. If you multiply 2 sounds you take the
product of each sample, so you again change the amplitude in a different way,
so the speed doesn't change. There are some concepts used to work with sounds
that are unique to Nyquist, and you need to read the first sections of the
Nyquist manual to understand them. If I were going to do much with Nyquist I
would have to go back and read the Nyquist manual, time which my wife would
consider not well spent!
BTW: Another post talked about how you can only represent a sound with a
frequency of half the sample rate. That is called the Nyquist rate, after
which Nyquist was named.
Gary
On 8/14/2017 2:16 AM, Damien Sykes-Lindley wrote:
Hi,
I don't know why, but Nyquist kind of reminds me of the GoldWave expression
evaluator. Not sure if it's because the common upshot was I couldn't understand
either one of them, despite having a programming background. Lol.
Seriously though, I may be mistaken but they both may have this weird caveat
that you need a working knowledge of both programming, and digital audio
structure. The latter I have no experience with whatsoever, so when people talk
about adding signals to increase volume or multiplying signals to mix or
multiplying/dividing individual samples to manipulate speed...I haven't a clue
how that works in practice. I was always taught that 2*2=4, not chipmunk.
*Grin*.
At least making an attempt to be serious again, I think it's more a theory
behind digital audio processing is more what's needed. Everything I have seen
so far has either been very technical, or doesn't make sense. For instance, I
read somewhere, or at least understood it as, that an audio sample is a number
between -1 and 1. If that were the case, that could easily be stored in 2 bits,
yet audio is generally saved as 16, mixed at 32 bit, which I calculate as
providing ranges of -32768 to 32767 and -2147483648 to 2147483647 respectively.
Additionally, I can't seem to find anything regarding a logical explanation
as to what the numbers mean. The simplest explanation used most often, which I
understand to a degree, is that each number represents either an amplitude, or
a speaker position. I've seen both explanations, not sure how they link
together but I guess they do. But nothing explains why doing something to one
number and something else to another can change the output in a way that makes
me think that adding would logically change the volume, multiplying would
logically change the speed, using a square root of the inverse sign might apply
a filter, or raising to the power of 16, dividing by Pi and adding the number
of miles between NASA's latest rocket and the sun would cause a flange. In case
those weird formulas start a form of interesting debate, let me clarify for
those that didn't pick up on it that those last ones are completely made up
garbage...I've no idea what would cause those effects and I've no idea what
using those formulas might do - knowing my luck probably cause a lot of
distortion and unwanted hiss.
But as you can see. I personally think that's the kind of tutorial that is
needed.
I've always wanted to make a convolution plugin since neither GoldWave or
Audacity seems to have one - Wondering if Nyquist is up to such a task once I
can get all this theory learned first.
Cheers.
Damien.
-----Original Message----- From: Steve the Fiddle
Sent: Monday, August 14, 2017 8:48 AM
To: audacity4blind@xxxxxxxxxxxxx
Subject: [audacity4blind] Re: nyquist little manual
There is this page in the Audacity wiki that covers much of the
Audacity specific Nyquist information:
http://wiki.audacityteam.org/wiki/Nyquist_Plug-ins_Reference ;
The complete reference for Nyquist functions is in the full Nyquist
manual: http://www.cs.cmu.edu/~rbd/doc/nyquist/indx.html ;
and detailed information about LISP in Nyquist (with examples) can be
found here:
http://www.audacity-forum.de/download/edgar/nyquist/nyquist-doc/xlisp/xlisp-index.htm
Steve
On 14 August 2017 at 08:39, Paolo Giacomoni mailto:paolgiac@xxxxxxxxxx wrote: ;
Hi listers.
I.m looking for a little nyquist manual, specially for audacity
applications.
Thanks you
Paolo
The audacity4blind web site is at
//www.freelists.org/webpage/audacity4blind ;
Subscribe and unsubscribe information, message archives,
Audacity keyboard commands, and more...
To unsubscribe from audacity4blind, send an email to
audacity4blind-request@xxxxxxxxxxxxx
with subject line
unsubscribe
The audacity4blind web site is at
//www.freelists.org/webpage/audacity4blind ;
Subscribe and unsubscribe information, message archives,
Audacity keyboard commands, and more...
To unsubscribe from audacity4blind, send an email to
audacity4blind-request@xxxxxxxxxxxxx
with subject line
unsubscribe
The audacity4blind web site is at
//www.freelists.org/webpage/audacity4blind ;
Subscribe and unsubscribe information, message archives,
Audacity keyboard commands, and more...
To unsubscribe from audacity4blind, send an email to
audacity4blind-request@xxxxxxxxxxxxx
with subject line
unsubscribe
