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Digitech RP2K and MajDomo Mail letters 1.0(plus other related things)
Digitech, Genx, RP2000, RP units, and other sound related info.
(Some of the links, might not/wont work,
Im seperating the page, major domo letters,
into 3 pages. So please bare with me.)
yep, my set up, (for now) see--http://www.robertkeeley.com
Tons of stuff below
Im trying to collect letters from the Major-Domo server sent to me . I'll try to arrange them in topics
If you dont want your name listed here or have a favorite letter or information you think would help us,
As in all of history some things come and some go.
So our major-domo has gone.
These letters are some of the only traces of their existence
(sounds cool huh)
The digitech web site now has the "newer rp units" site,
on their message board. So see you there.
or the whole site at;
but these letters, this information, will remain here, so have a look!
It's unmoderated, so keep it clean , Tony
so skip over the "What is a Major-domo"--------
Hello everyone. After deleting almost all, of the last, weeks messages sent to the major domo. I thought id explain how this thing works. I don't run it, I have subscribed just like you.
When you send an email to firstname.lastname@example.org it sends the letter to every one who has subscribed. It is an automatic system run by a computer server somewhere. Occasionally a support staff person from Digitech answers a question, so I understand someone from the big D gets all these just like us.
If someone asks a question. and you wish to answer. There are two options for you to choose.
1 REPLY ALL to answer to the major-domo (we all get the answer) and in conversations like tubes and stuff its a very good learning tool. For me any way)
2 REPLY this sends your answer or comment to the originator only. and the rest of the members of the domo don't get the email.
If you want to tell someone to go jump in a lake use the second one please
This forum major-domo is for RP products discussions and questions (well the guitar questions are good to) ONLY. Political comments should not be included here, in this forum.
This forum has world wide members from UK, Switzerland, Canada, Brazil, USA, Italy and others, I personally think this is cool. And respect all those who answer. So Lets put this behind us and move on.
Greetings to all and thankyou for taking the time to post some great info.I am truly humbled by my lack of knowledge.
Just a few pointers on posting to limit duplicate messages:
1)When responding to the group,please make sure that the RP List address,
email@example.com ,is the only one appearing in your email To & Cc fields.Email messages are sent to every address in those fields,so the individuals listed will receive it twice as the list automatically routes it to everyone subscribed.
This will occur if you select "Reply All."It appears that when you select "Reply to Sender" the message goes to the individual (at least in my progy-Outlook Express).
2)When responding to an individual,delete the other addresses from the fields mentioned above.[Here I would ask you to please share your knowledge with everyone] :-) This may be useful for off-topic messages.
3)If you have Read Receipt Request enabled,(Darwin,thanks for your help,BTW)please disable it.Those read receipt messages appear to make it to the list and therefore,everyone on it.
Once again I would like to thank everyone for your participation.Thanks to those with questions others didn't think to ask or didn't know to ask-we all benefit.And thanks to those with the knowledge and spirit of a musician to take the time and effort to help.
One thing about musicians that has always made me feel part of a brotherhood is the willingness to help others for no apparent reason or compensation other than feeling good about doing it.No matter how advanced a player,we tend to be very unselfish,because we *all *were beginners and we* all* learned from those with more knowledge and experience.We pretty much all want to give back.In this spirit,I ask everyone,whatever your level,to not be shy.There are no stupid questions.Like I said,we've all been there.
YES YOU WILL HAVE TO RELOAD YOUR SAVED PATCHES AFTER UPGRADEING....
Sorry dude.Just clicked on link I sent,it brings up whole list.Here is the specific info:
There was 1 question found:
I just purchased an RP2000. I heard that there was a new upgrade chip. Is this true? What does the update do?
Digitech RP2000 submitted on 2/16/2001
Ver. 1.3 fixes low SPDIF signal and you don't have to send in your box.
Call this number.They will send version 1.3 free of charge.
Digitech Repair (801) 566-8800 ext. 626.
The upgrade is not a crucial one. If you want to upgrade to 1.4 it will be neccessary to have you send in your RP2000 as it would require a different treadle pedal. The update is simply an E-prom however, this would cause your existing pedal to stop working. If you decide to upgrade you will need to call us for a return authorization #.
Digitech Repair (801) 566-8800 ext. 626. Michael
You are correct.1.4 is the latest version.It involves replacing an E-prom chip,but makes your treadle pedal inoperable.(The upgrade involves improved treadle electronics)So you have to send it in to get the pedal replaced.
1.0 is the original,1.3 is the first upgrade.The guy at Digitech repair I spoke with said 1.3 is a bug fix for midi implementation.They will send you the upgrade chip @ n/c.
This is too funny check it out;
_______I wanted to upgrade my Digitech unit to version 1.4 to correct the spdif problem. After reading the FAQ (http://rushtabs.tripod.com/FAQ.txt) concerning this, I learned you need to return the unit to the factory so they can also upgrade the foot pedal. I called digitech and they said all you need to do is intall the 1.4 chip yourself and it will work. They mailed me the chip, I installed it, and guess what, the foot pedal did not work properly. So i contacted digitech and they said, yes someone made a mistake, please return the unit and we will upgrade the foot pedal.
So I had already installed the 1.4 chip. To avoid confusion, I enclosed this letter with my rp2000 when i sent it to tech support:
Dear Digitech repair:
You sent me the version 1.4 firmware upgrade chip, but did not tell me about needing to replace the foot pedal treadle assembly. I have installed the 1.4 firmware.
PLEASE UPGRADE THE FOOT PEDAL TREADLE ASSEMBLY TO MAKE IT COMPATABLE WITH THE 1.4 FIRMWARE!
I got the unit back in about 3 weeks. On the shipping address label and the tech support form, my name was "misspelled" as Dick Smith.
The only conclusions I can draw from this are:
1. Somehow they truly did misread my neatly typed letter confusing the name Ed with Dick.
2. When they issued the RMA, the person misinterpreted my spelling of "E-D" as "D-I-C-K" over the phone.
3. Someone got insulted by my letter and felt I was a Dick rather than Ed.
I did not act like a Dick on the phone when I requested the RMA. They made an error in shipping me the 1.4 chip, but I did not bring this up or act annoyed or say anything abrasive.
Now I ask the Digitech community, was my handling of this situation deserving of the name Dick? My fear was that they would plug in the unit, see the 1.4 flash on, assume that I bought the unit as upgraded, and ship it back to me without doing anything!
Isn't that immature calling me a Dick? If I owned the Digitech company, I'd be pretty embarrassed that my employees were calling their customers Dicks; even if they sent a letter specifying what needed to be done to the unit in bold-faced capital letters.
Being called a Dick like this really gives me a bad taste in my mouth concerning this company. I'd advise anyone sending their units back for upgrading or repairs to avoid using firm language or bold typeface in their requests, otherwise they too may "accidently" be called a Dick.
Thanks for listening
Ed "Dick" Smith
> What version of firmware do you have?
> I upgraded to 1.4--but I did it right after buying the unit so I can't
> really compare the before and after. Everything seems to work really well
> as> 1.4, but if you're doing it for spdif, i still think the recording level
> you> get is still too low to be worth using it.
I just upgraded recently. The 1.4 update did improve the spdif; as Ed
says, it's still not what you'd call a hot output. I use the digital
exclusively (home-studio, not live use), so I was pleased with the change.
>> For those that have done the RP2000 1.4 upgrade does changing the treadle
> assembly make it any more robust?>
> The current one makes a horrible phaaaaarrrrrtttt sound when I try to
> change the delay time on the fly using the expression pedal because the current
> assembly bends under the weight of my size tens...>
I'm not sure the digital delay is one of those things that is readily
modulatable (via lfo1/2 or pedal) -- some effects parameters just don't
sweep smoothly on a digital unit (still you can make an interesting racket this
way ;-) ). But I think you're referring to the physical assembly, not sound
output. I always wished there was an external pedal jack instead of the
built-in. It always seems a little stiff and creaky... perhaps it will
loosen up over time (I just got it back from the shop, and it does seem a bit more
solid--but the change in feel may be due to them re-tightening things)
what's the difference between digital and analog delay?
Analog delay uses electrical circuits to delay the signal.
Digital uses a digital program to delay the signal.
The RP2k uses a digital program to simulate an analog delay.
That's it in a nutshell
To add to Tony's answer,
I think of analog delay in this way,
The analog delay is accomplished with 1/4" tape moving in a loop across magnetic tape heads. The Binson Echorec, Echoplex and Roland made one I cant think of the name. This list is not inclusive as Im sure there were others made. These are the ones I've used.
The loop of tape similar to an 8 track tape with less tape. It doesn't have an end it just cycles. The signal is recorded using one head, then played back as the tape goes by another head. Some of these units physically moved the playback head to gain longer "echo". Some had an erase head just before the record head.
The Roland used different heads placed along the tape path, allowing the user to select the head i.e. amount of echo.
The debate of anolog tape to digital rom recording is one of the biggest debates going on today.
so anolog can be thought of as physicaly moveing tape and the magnetic elements involved with placeing sound on tape and then reproduceing that sound later in an Echo bounce. Some people used this output(from the echo unit ) to drive a secound amp. Gilmore,Beck,Page. Or did that again to a third amp creating a masive sound The guy in Queen. along with his guitar and this set up he came up with his signature sound(Cant think of his name) EDIT: Brian May
The digital does all this with electronics and ic chips.The early ones werent bad but the newer alogrithms are spectaclar. and the low mantenance well, no mantanance.
Where the debate over cleaning tape heads or not cleaning tape heads is gone. also the physical tape being recorded over and over led to degration of the electrons on the tape . these tapes had to be changed ocasionaly.
so in the rp2k unit, the selection in delay called analog, I think well turn on the tape.
The modern digital delays are so much easier, no more carting around an extra box. And hooking it up ect. Its just a selection on a list of selections on the floor unit. (I have since read an article which states a Binson was a wire recorder, They recorded on a wire loop.)Hope this explains something,
I'll add a few more things, to this discussion as well.
I love the analog tape discussion. That was, in fact, the first delay that we had and it was "invented" by Les Paul.
Later on, when electronics evolved we were able to, with simple IC's and transistors create a "bucket brigade device" which was able to delay the signal and feed it back, as well. The signal, in these devices wasn't sampled and processed, it was just fed back as it came into the device. The length of time that the signal was "held" before it sent it to the output jack. The more "devices" in the signal path, the more input information the thing could delay.
The other thing about these is that the signal was continually passed through the same electronics, over and over again. This has the direct result of degrading the output signal very quickly. The higher frequencies were rolled off quickly, they produced some "distortion" to the signal, etc. They didn't sound "clean" like the newer digital delays. However, again, this is due to the fact that the "raw" signal was fed through the circuitry over and over again for processing. (Similar to the tape delays that were used prior to this). The nice thing was, is that you could, w/ the use of electronics, compact a fairly good delay in a small place and not "suffer" the degredation of tape that the older EchoPlexes, among others, had. It was "clean" every show.
Digital delays are different. They sample the incoming signal and alter it, digitally, and add the altered sample to the original signal. In this way the delays are consistent and clean for every repeat provided by the feedback. As the amount of feedback (number of echoes) increased, the subsequent echoes were as pristine as the original one. There was no degredation in the output sound. This, of course, was a wonderful thing, until every decided the older sounded better. Digital was to clean.
The "analog" delay in the RP2K, GNX and all the newer guitar processors are simply digital models, that are programmed to degrade the overall output to match the degredation in the original units. They're still digital... but they sound "warmer" due the fact that we're degrading the signal over time in them.
Hope that answers your questions... go ahead and ask away... you guys are helping me to, finally, get my thoughts together about all of this gear... it's time to write a book....
dar excelent , I did forget the time element in tapedelay the delay time was also acheved through the slowing or riseing of the tape speed.
before we could afford a "real" delay, I would feed the signal into a reel to reel tape machine and output back to the pa. by turning the tape speed to 7-1/4 or 3-3/4 the echo sound would change . this was a crude version of the "boxes". the ehcorec, echoplex and the space echo (roland later )along with the feed back darwin mentioned also used tape speed to change the delay output.
I have 2 reel-reel machines. One a "newer" TEAC and an ancient Sony. Both
are 2 track. Can I make one or both of these into a delay machine. If so,
I've always wondered how the old studios would take a reel of tape and wrap
around mic stands and what not to increase/decrease the delay time.
want to sell one haha-- input to record in, tape deck. press record, output to pa/board ,source of input, different fader so you can adjust return volume or use the output levels on your deck (if it has one). It will run as long as the tape lasts remember this is a crude model. you will have three speeds? 1-7/8/ 3-3/4 7-1/4 these will be your delay times you may need to select monitor output on your deck. but play with it. you can do custom stuff like go "oh baby" and put your finger on the reel as it goed by, and slow it down or speed it up--live .
10/8/2001 6:03:29 PM, David D wrote:
>The purpose of compression is to make quite parts louder and louder parts
>Attack = how fast the input signal is grabbed by the compressor
>Ratio = how hard do you want to compress the signal. Ratio is directly
>related to threshold. Let say you set your threshold at 85. This means any
>signal that becomes louder that 85 will be compressed. If you have a 1.8 to
>1 ratio; for every 1.8 db of signal above the 85 threshold 1 db will
>actually be sent to the output channel.
>Threshold = at what point you want the signal to start being compressed or
>Gain = how strong your input signal is.
>I know this is confusing but if you start with the concept of a limiter it
>can make it simpler. A limiter will take a signal and cut off everything
>above the threshold. So if you set a limiter to 85 (from our previous
>example) all signal greater than 85 would be cut off or truncated. If you
>were playing an acoustic guitar and you played the strings very quite and
>then set your limiter at a very low threshold. No matter how hard you played
>your strings your guitar would not get any louder. A compressor works the
>same way except you have the added dynamic benefit that any signal above
>your 85 is not abruptly cut off but is nicely rounded to give it a more
>I hope this helps,
Thank you very much David. A very thorough explanation. I know how to get what I was wanting out
of the compressor now (infinity:1, threshold 50, and fast attack was what I was looking for, I
just didn't know it :-)).
Brent 'Goose' Towsley
The compressor on the newer RP units is kind of funky. It doesn't seem to respond quite like a normal
compressor to me. But, in answer to your question:
Attack- how fast (in milliseconds) compression occurs once the threshold has been exceeded by the
signal. A fast setting provides the quickest response, but it may suck some of the life out of the
sound. Transients (pick attack and the like on guitar) will suffer the most. A long setting allows
for more transient response and a more "open" sound, by you will not get as even an output. Medium,
of course, is somewhere in between. It'd be great if you could set the attack time in msec like on
most compressors, but Digi has preset this parameter for you. I do not know exactly what the number
of msec is for each setting.
Ratio- this is the setting (in decibels) that determines how much compression is applied to the
signal once it crosses the threshold. A setting of 4:1 means that for every 4 db's the input signal
goes over the threshold, the output signal only increases by 1 db. A setting if infinity is whats
known as "hard limiting". With this setting, the output is clamped to the threshold no matter how
wild the input signal gets. A ratio of 2 or 3 to one is nice for an even response out of vocals or
acoustic guitar. A ratio if infinity to one will get you Brian May's totally "squished" sound (a la
We Are the Champions).
Threshold- this parameter controls the level at which compression occurs (note where I used this term
in the above paragraphs- this is it). It is expressed in decibels. You can do some very drastic
things to your signal with this parameter. For light compression, I like it around 8-12 db. For a
more radical effect, try 20 db or more.
Gain- this parameter is used to provide "make up" gain for the compressor stage. As the signal
crosses the threshold and is compressed by the ratio selected above, there will be a reduction in the
overall loudness of the output signal. This parmameter allows you to make up for that loss so the
output signal is as loud as the input. Here's a little effects heresy for you- I use this parameter
on some patches to make the signal hotter on the output than the input. It adds volume and sustain to
some patches. It also adds noise and feedback, so be careful!!
On most compressors, there is also a "release" parameter that controls how fast the compressor stops
compressing after the signal has dropped back below the threshold. A long release time can add to
that squished sustain sound. Digi has preset this parameter- it would appear to my ear that it's set
to a fairly short setting. Again, as with the Attack- it'd be nice to have control of it.
Hope this helps. Compressors are sort of odd items. They are not really an effect like chorus or
reverb- they are essentially level controllers. When I record, I use them for that purpose a lot. On
the other hand, by using more extreme settings, they can become like an effect by significantly
altering your sound.
The Axeman (#(==>>
Here's a quick explanation of what all of those settings mean and what they'll give you when you set them to various settings. Ultimately, when using a compressor, you're sort of tweaking it so that you sound good, to yourself, and to others.
OK... that said, here we go.
First, what a compressor does and is... A compressor is what is called a "Voltage Controlled Amplifier" or VCA, for short. It helps to look at it in terms of a real amplifier. Let's say the real amplifier will amplify input voltages 100 times. So, with 1 volt in, I would get 100 volts out. 1.01 volts in would give 101 volts out. 1.10 volts in would give 110 volts out, etc. The gain is 100 and for every 1 volt of input change, the output changes by 1 volt.
Going further... the compressor really does two things. 1) It amplifies (remember it's a Voltage Controlled Amplifier) the lower sounds. 2) It actually reduces the amount a "hot" signal is amplified, bringing the two closer together in terms of volume. In other words, it's output is not linear, or straight.
When you've reached the range where the compressor kicks in, the output levels do not follow the input levels. Thus, let's say, in the above example that the "threshold" was set to 1.5 Volts. Compression ratio reduces 2:1 (and the reduction is in volts). In a regular amplifier going from 1.5 to 2.5 volts would increase the output from 150 to 250 volts. Going from 2.5 to 3.5 volts would bring the output from 250 volts to 350 volts. However, we're implementing a 2:1 compression ratio, now, which changes things. For every 2V of input change we're going to only increase the output by 1V. That's 2:1 compression. 3:1 states that for every 3 volts of input change the output will change by 1 volt. Infinity to 1 basically states that for all input voltages, above the threshold, the output level remains the same
So, let's look at this a bit closer... and view each of the parameters as they stand:
Attack - This is how fast the compressor responds to an incoming signal. Several things are happening here. First, when you play guitar you get an initial sound that is very strong. That's when your pick first hits the strings. In terms of power, it's the loudest sound that you have, and it's something that can overdrive effects, etc.
A slow attack means that this initial "pick attack" is actually "missed" by the compressor. It takes a while, in slow mode, for the compressor to activate, thus the only part of the sound that would get compressed is the sustaining strings, and not the initial pick attack. Part of what makes the guitar sound is the intial pick attack. However, that also produces a "sharp", "transient" sound that some folks don't like. Setting the compressor to slow allow all of that to go through.
If you're amplifying acoustic guitars, you're probably going to want to leave the attack set to slow. The reason being... the acoustic sound, overall, is very dependant upon pick attack being passed through for that "high-hat" sound.
Medium, will catch some of the longer transients, but not all transients. Fast Attack, basically, allows the compressor to compress nearly everything that hits it.
Ratio determines how much the output level is reduced. Many guitar sounds are based on a 2:1 - 4:1 ratio (depends on the player and the comfort range). Guitars compressed to this area, typically, have enough variation in overall dynamic range, to sound "natural, but enough to "smooth out" the sounds. Compressors, of course, have direct result of making a "bad" player sound a little better, as they even out the volume of the picking. Thus it makes a player that's not even sound more consistent (too bad they didn't fix bad notes in the process, hey? That's my dream)...
Setting the Ratio to infinity turns the compressor into a limiter (you've seen the terms). Basically, the function of a limiter is to take all input levels (that are above the threshold) and make them all the same output level. While a compressor isn't really a limiter, it does function very closely to one, when the ratio is set to infinity.
Threshold. This is the input level that finally triggers the compressor. Let's say that you want to be able to play very soft passages and hear the difference in volume between every single note... but then, in the same song, you want to beat the living daylights out of your guitar and play really loud... but, the problem is, if you're playing straight, without compression, the loud parts are hitting so hard you're overdriving your effects and your amps... and when you turn down to compensate for that you can't hear your quiet parts.... "Enter the Compressor..." Simply set the threshold so that you can play the soft parts w/o it kicking in... then, set your ratio higher (to start "limiting" the loud part so you don't overdrive your signal chain)... and voila, soft and loud... both closer to each other in volume, but niether of them causing the soundman any headaches. This takes a bit of practice, but you can get there! w/o too much trouble. The threshold, basically, determines when/where the compressor is going to kick in.
Gain... this is the OUTPUT LEVEL of the compressor. This is how "hot" it's going to hit everything else in your RP unit. Most compressors have this type of a feature. That way, if you're using it to compress all "soft and quiet" parts, you can drive the other parts of the RP unit w/ a strong enough signal that you still obtain a good signal to noise ratio, which is important.
That's about it... hope that this helps. If you have questions, let me know... we'll try to answer them....
But also compression can be likened to, How do you get 2 liters of milk into a 1 liter jug. ( I know I know)
It takes your sine wave, the one you created by making sound, and pushes it into certain limits. I attached two art creations, to explain (I hope best as I know how). The first is, sine wave2, thats sound. The vibration sound makes, so your ear can hear it. The second is compressed sound1, with in the limits that were selected. (I just drew two lines and yes they should be straight.) So that the compressed sound is kept with in these limits. That way, the volume level (example) can be raised without worry of spikes of sine wave on an uncompressed tone.
Rates the sound is compressed, as the sound enters the compressor can be adjusted. The speed at which the sound can be attacked with compression and the amount of compression, the volume pre or post of the compression. stuff like that there, are all adjustable. ah It has its uses.
Have a good one
barefootterry check out these high quality graphics
At 17:40 09/10/01 -0400, you wrote:
>I have really enjoyed this discussion on Compressors. I knew what a
>compressor was before, But i learned a lot more about them. I would like
>to request (and this is only my opinion) that say every week we discuss a
>different effect to this detail. I am sure everyone could learn something.
>The manuals just dont cover everything. I like hearing about how people
>apply these effects to suit their needs, and i like hearing the technical
>details. In the meantime, I am gonna start printing these out and
>compiling a binder with them. Even the "newbies" offer a perspective that
>you cant always get out of a manual.
>I would also like to take the time to thank everyone who posts here, as i
>hear a lot of enlightning things.
>Let me know what you think about my idea.
Off topic, Could someone explain what makes an amp a "Class A" amp.
I have heard this term thrown around, but no one gives me an explanation.
What is a Class A?
Class A is a term given to an amp that runs its tubes at full current all the time, unlike most tube amps that alternate between running one set of tubes and the other set, each for one half of the wave. The set not in use is turned off by a positive swing of the grid voltage. Single-ended out-put stages always operate in Class A. Most push-pull amplifiers, including the venerated Vox AC-30 operate in Class AB when overdriven, even if they are in Class A while clean. The upshot is that Class A operation has its own unique tone characteristics that set it apart from other tube amp classes. Class A amps sound great at low volumes, and even better as you turn them up. Thus, with the relatively low wattage of the UniValve you can turn up the amplifier to take full advantage of its stunning output distortion tone without deafening anyone
However, the Class-A/B thing I can tackle and give some insight into their
There are two types of power-amps out there, a Class-A and a Class-B. All
preamps, for the most part, are Class-A. Of the types of amps, Class-A power amps are the most expensive. Class-B are less so.
It's important to understand how AC works, before we get into this. I'll
explain a little bit of the important stuff. Questions can be asked, if
needed, to clarify certain points of understanding. This is a complex topic
so I don't expect everyone to understand everything, right off. If they
do... wow! That's awesome.
Anyway. AC voltage flows, if you will, in two directions, positive and
negative. AC voltage always varies and changes either regularly or in
irregular intervals. All positive voltages are voltages that are higher
than 0V. All negative voltages are those less than 0V. DC voltage either
flows one direction, or the other, and remains at the same voltage all the
When amplifying AC voltage, the amplifier must track all variations,
exactly, without adding any distortion (changes in the input waveform) to
the output. Even slight changes can cause something to sound very
different. There are different types of amplifiers that we can use to do
this, Class-A and Class-B amplifiers. The difference in the amps is in the
way that they handle the transition from positive voltages (those above 0V)
and negative voltages (those below 0 volts).
Another point about AC voltage. Let's say we have a symmetrical input
voltage, i.e. a voltage where both positive and negative peaks are the same
voltage. For example +10V (positive peak) and -10V (negative peak). This
represent a total voltage swing of 20V. If the amplifier had a gain of 10
(i.e. it amplified the input signal by 10 times) the output would have to
range from +100V to -100V a full 200V swing. That's large. It's just an
example to show some points about the differences and why we do what we do,
Pre-amps are voltage amplifiers. Power amplifiers are current amplifiers.
There are fundamental differences. When you amplify voltages, without
current, you end up w/ a quieter device, overall. The pre-amp is the
critical part of any amplifier, thus we use voltage amplifiers in them.
The "downside" to voltage amplifiers is that they can't "drive" anything.
Speaker voice coils are "current-based" devices and require a great deal of
current to get them moving. A pre-amp, because it's using relatively small
levels of current, and larger levels of voltage can't effectively drive a
The power amp, is a current amplifier. Voltages are low, current is high,
which is what is needed to move speaker diaphragms. These are, for the
most part, much more susceptible to noise (hiss) than what the pre-amp is.
However, given a well-designed pre-amp, coupling it to a current amp isn't
going to be that bad.
Let's look at the different types. In a Class-A amplifier, there's a single
active device (either a transistor or a tube) that amplifies both the
positive and the negative cycle of the AC voltage. Remember, from above,
if we have an amp that has even moderate gain, the output voltage swings are
going to be very great.
A tube, or transistor has a finite range of AC swing that it can do before
it distorts. This range is what we call a "load line" in electronics
terms. The load line is straight in the middle and flattens near the top
and the bottom. What this means is, is that as long as the tube, or
transistor, is swinging w/in the flat part of the load line it's amplifying
input signals exactly. Once it goes beyond the flat part of the load line
it's not accurately tracking the input. This is not a good thing as it
starts to distort the signal.
In a Class-A amp, as noted above, a single transistor or tube handles both
positive and negative swings of voltage. This takes a very "powerful"
device and something that can take not only large amounts of current, but
also fairly substantial voltage swings. Typical tubes and transistors are
designed for either current or voltage swings, but not both. That's hard
to do and drives the price up. Biasing normal tubes, or transistors, to
perform Class-A amplification is also done, but this results in a decreased
Power, for everyone's benefit is I squared E... that really makes a lot of
sense, doesn't it? It did when I went through school... basically, what
that means is, is that the power output of a device is a square of the
current x the voltage. Thus, current has the biggest direct effect on power
output. Remember I stated earlier that pre-amps are low-power devices
because they are voltage-based and not current-based, this provides some
background on why that's the case. In a Class-A amp, because voltage and
current have to be carefully balanced, you end up w/ a much lower powered
The other thing about a Class-A amp is that the output device (tube or
transistor) must be able to handle all the current in the output stage, as
well, which requires a very "hefty" output device. Finding something of
that handling capacity isn't easy. Current handling, which is required for
power amplifiers, is a critical factor, here.
Let's look at Class-B amplifiers, for a moment and then we'll talk about the
benefits, differences, etc.
Class-B amplifiers "split" the load. In a Class-B amp there are, usually,
two active devices that both share the load. One of them takes everything
above the 0V mark and the other takes everything below the 0V mark. The
"problem" is in the transition of the two devices. They don't "hand-off"
the voltage swings perfectly. Thus, around the 0 mark, in both the positive
and negative directions, there's a bit of a "glitch" in amplification.
This "glitch" is worse when the transistors, or tubes, are not equally
matched. Nowadays, it's pretty easy to match them, though, so it's not all
that bad. However, this does cause a minimal amount of harmonic distortion,
especially in the voltage levels that make the signal "hang" around that
The good thing, using the example above, w/ an input voltage of 10V and a
gain of 10... is that each amplifier device (transistor or tube) only has to
swing 100V, rather than 200V, as they are sharing the load, together. It
also means that they're sharing the current load, as well (they don't have
to both source all the current for the speaker load).
So, what are the good/bad parts? Class-A amplifiers are noted as the
"cleanest" sounding of all amplifiers. They amplify w/ the least amount of
distortion to the overall signal. Audio purists prefer Class-A amplifiers
as they don't "color" the signal at all. Of course that coloring is purely
subjective, for the most part, as very few folks would ever be able to tell
the difference between a Class-A and a Class-B amplifier, even w/ perfect
speakers in a perfect listening environment. To get tubes or transistors
to handle power-amp levels, in a Class-A is hard to do. They're expensive,
not easily obtainable, etc. Plus, Class-A's have the inherent "problem" of
not being able to build up a lot of power.
Class-B amplifiers, obviously have the inherent problem of "distorting" the
signal a little bit. It's, for the most part, unnoticeable by human hearing
standards, but test equipment will pick that up and show it. They have the
obvious advantage of being able to produce twice as much power output w/
cheaper devices, as both of the devices are "splitting" the load power.
That said, Class-A amps do sound a bit different than the Class-B amps, in
some ways. However, that's also a part of the pre-amp design, as well,
speakers, cabinets, etc. VOX are Class-A... Marshall's/Fenders are all
That's about it, for now. Hope that helps to get some basic understanding of
the differences. Questions/comments, please let me know. I'll be glad to
add some additional information to explain anything that's unclear in here.
Great info Darwin.What about class AB?How do they integrate the two?And what is push-pull and what is it used for?(Sorry for assuming you're an electrical engineer but you seem fairly well versed).Thanks!
Class AB is a modified version of the Class-B that, effectively, eliminates the cross-over distortion (passing from voltages above 0V to voltages below 0V and vice-versa). Class B amps have a bit of that distortion inherent w/in them. It's not a different type of amp, it's just biased (an electrical term for the way they apply voltage to the things) differently.
Push-pull is a Class-B or a Class-AB amplifier. One side "pushes" the other side "pulls" the current through the load.
I believe that when you set global cabs to ON, it turns on all preset
cabs for each user preset. To make them all BRIGHT, you would have to edit
each preset and save it with the BRIGHT cab chosen. Then when you enable
global cab they would all be BRIGHT. Not the quickest way, the only way,
Hope this answers your question.
This is correct... it's a bit of work on this thing... no way to set a "global emulation" that impacts all patches.
i just got me a midi cable at last but it isn't
working both with rp2kedit and ripper2000...
any help with the setup?
those are both available on the Patches and links page here. under patches
Or use the Susex Saver Programs--susex
I don't know what your problem is, but I had a problem that might be similar.
On MY cable the connector on the cable that is labeled out goes to the port
on the RP2K labeled IN. (of course it goes to follow that the connector
labled in goes to the RP2K port labled OUT).
If your computer came with a sound card on the mother board and you have
upgraded your sound card, make sure you plug the midi cable into the new
Those are the only two suggestions I have.
You might check the My computer/ control pannel/ system /device manager/ sound-video-game controler.
Make sure its turned on and working properly,(the sound card that is).
I finally got my MidiSport 4x4 midi interface working, and was able to hook my RP2K up to it's own port for data dumps and using the Ripper program (I can bring it up while running Cakewalk- very cool!). It works great. the neatest part, though, was accidental. The Ripper defaults to "Direct" amp model and nothing else whatsoever turned on, all values set to 0. I accidentally sent this to my RP and got a totally dry guitar through the RP. I started auditioning just the amp models (no other stuff in the way). Within minutes, I had built up several patches that had sounds that had eluded me otherwise (how about a good Angus patch? Tougher than you might think for such a basic, simple sound). Just thought I'd pass this on. It was an approach I'd never tried before. Usually I start with a preset that sounds "close". This seemed a lot easier in some ways.....YMMV
More sources on MIDI----MIDI
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10,815 messages since Sun, 8 Sep 1996 03:21:55 GMT+0100
If this list really goes, I for one will miss it. A lot.-
me to!! you will be missed, you list you, message boards are just not the same!!
The list is going bye bye, so if you have a favorite thread of thought, guestions, ect for the Major-Domo letters & other related things. please send them on over and it has been a real pleasure to have known all of you guitar nuts you. so take care and see you in the (ick) message board!!
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