Thread Tools Display Modes
Prev Previous Post   Next Post Next
  #1   Report Post  
Posted to rec.audio.pro
jaric
 
Posts: n/a
Default Am I understanding this correctly?

Ok, this is the way I see it: you have bit rate and sample rate.

The sample rate is the number of times per second that the signal is cut up.
The Nyquist Theorem states that the highest frequency reproduced is half of
the sample rate. This is too long and convoluted for me to understand or
explain. But so long as the sample rate is CD-quality (44.1 kHz), we'll have
a possible frequency of 22.05 kHz, which is out of normal human hearing and
definitely out of the range of electric guitar.

The bit rate is the dynamic range of a unit. Let's say the input is 1 volt
and the converter has 1 bit. It can be either 0 or 1. So if the signal is
closer to 1 volt, it will be 1, and if it's closer to 0 volts, it will be 0.
Obviously, you'd have full signal then nothing. With 2 bits, you double that
dynamic range. Each additional bit doubles that range so you end up with 2^N
different volume levels for each bit rate N.

Let's also say you have a guitar signal that comes in at only half the
volume the input buffer wants to see. You are wasting half the bits right
there, squashing your dynamic range. Likewise, if you have buzzing or
humming noise, it masks the lower bits squashing your dynamic range again.

A compressor will take a volume level over a certain threshold and reduce
it. You can make up for this reduction by increasing gain, making the louder
parts just as loud as before, and the softer parts louder. If done subtly,
you will just notice a smoother input.

A compressed signal has the POTENTIAL to increase dynamic range by ensuring
that your maximum peak levels are not too much higher than the rest of your
playing. Let's say that you hit the strings on a guitar as hard as you can
and it peaks at twice the normal playing volume. Again, you're wasting bits
and dynamic range right there. So a compressor will get your guitar
operating at a higher average level, increasing the number of bits used and
thus dynamic range.

Here I will BS a bit because I don't know exactly what happens (I'm an audio
enthusiast, not an EE major). A buffer will take a high impedance signal and
output a low impedance signal. This signal is less susceptible to
high-frequency and volume loss, perhaps?

Furthermore, impedance is not a flat number; it changes based on frequency.
I'm imagining that by converting this impedance, you are also flattening the
impedance of a guitars pickup, for instance. By doing so, certain
frequencies are not picked up louder because of different impedance, but
because of different output characteristics. In this way, you will not have
spikes or dips in the impedance of a pickup, which I propose, decreases the
dynamic range.

In my opinion, a well-designed, hum- and noise-free, buffer with a hint of
compression (i.e. a Valvulator), can actually increase dynamic range.

Am I understanding this correctly? Feel free to rip my comments to shreds.


 
Thread Tools
Display Modes

Posting Rules

Smilies are On
[IMG] code is On
HTML code is Off


Similar Threads
Thread Thread Starter Forum Replies Last Post
Understanding The Technical Market Indicators million dollar Marketplace 0 February 21st 06 02:30 AM
Understanding T-Racks... Mr.Composer Pro Audio 6 December 27th 05 12:45 AM
Help understanding Plate Resistance Joe Vacuum Tubes 9 February 3rd 05 02:38 PM
A little feedback worse than none at all? NewYorkDave Vacuum Tubes 211 November 26th 03 09:13 PM
Understanding Implementation of High Pass Filter Scott Soderlund Pro Audio 2 August 2nd 03 03:22 AM


All times are GMT +1. The time now is 09:07 AM.

Powered by: vBulletin
Copyright ©2000 - 2025, Jelsoft Enterprises Ltd.
Copyright ©2004-2025 AudioBanter.com.
The comments are property of their posters.
 

About Us

"It's about Audio and hi-fi"