Sound Clips n Stuff
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)
First, a little Shamefull Self Promotion.
Please play this file ?
then put it in a file, then open it.
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 email@example.com
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,
,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've done a TON of recording of guitars over the years. The "magic"
ingredient of most of the "big" sounds is the fact that there's a TON of
room ambience as a part of the sound. Very few of the guitar sounds
that we're all used to hearing were made direct to the board... Well,
if that doesn't just wreck the whole magic of amp/cab sim's like the
Digitech, POD, BOSS, ZOOM, Korg and many others, hey?
Typically, when doing recordings I will have the guitarist run through a
couple of cabs, many times in different rooms. I'll put the "magic"
SM57 in front of two different speakers in each cab (all 4 of the
Marshall speakers sound different, believe it or not). I then,
usually, put a few Nuemann U87's about 6-8 feet away from the cab and up
in the air... My favorite layout is an equilateral triangle - which
each side about 6-8 ft. I then take and mix the "direct" SM57's and
the "ambient" U87's together, adding just enough of the room to really
make the sound HUGE.
Most of the folks that I know use this direct/ambient method to really
capture the overall sound of the guitars/amps. There's no other way
than putting the natural ambience of the room into the mix to get a
good, quality guitar sound overall. The same goes for acoustic guitar.
Typically, what I do w/ that is to put take the pickup output and put
that through a direct box (SansAmp or, my favorite the new Yamaha AG
Stomp). I then mic the thing w/ an SM57 (mellow w/ glassy highs) or a
good, clean/crisp condenser (there are a few out there, Neumann 180
series being one of my fav's). I put a few full-range, flat mic's in
the room w/ it (again the triangle ambience) and pick up the sound of
the guitar and the room. I then mix the 3 mic's and 2 direct inputs
together to create a very full sound. It sounds larger than life when
Drums... Well, I put a mic in every tom... One above and one under the
snare... One over the high-hat... One in the bass drum... And overheads
(stereo X-Y pair)... I then sub-mix everything and EQ before I hit the
board... Great day!!!! Printing hot to tape... I break all the rules.
Bass guitar I love running through a direct box (SansAmp really makes a
nice one, Bass POD, etc) into an LA-2A compressor to tighten things up.
I'll EQ the heck out of the thing once it hits the board...
In the absence of this fancy stuff... Run what you can direct to the
board and use mic sim plug-ins. Also add some ambience to it (room
ambience w/ reverb) and then add all of that in. The ambience should be
panned hard left/right (I like all the effects on the "fringe" of the
stereo field. Pan all the other stuff closer to the middle.
That's a start... Have fun.
Here's some info on capturing the SRV sound Cold Shot. Kinda need the
GNX1/2 to do this... Stevie used a mixture of amps to get that sound. On
this album he used a Marshall 2x12, Fender Vibroverbs and Fender Super
Reverbs. In fact, I looked at the DVD of his "Austin City Limits" concert
and he has those in the background, as well, during that concert. That was
back before he even had the keyboardist. Most of the stuff I've read about
his sound is that he used the Marshall clean and the Fenders for distortion.
I stole this SRV patch from my LINE6 gear... this is pretty close,
actually, to the sound that he's got. I was using a Strat w/ Vintage Lace
Sensors and Vintage-Style electronics, just to give some background on how I
determined the overall closeness of the sound. Stevie uses the neck pickup
(at least he was on Austin City Limits) during the song.
The amp model of choice is going to be the Bassman. It's the closest thing
to the Vibroverb that you're going to get off the RP300 - or any other amp
modeling unit, for that matter. Amp Gain should be slightly about
mid-range... this just starts to break up the Bassman. Amp Level about 75,
or so. Unfortunately, Digitech forgot to put the Bass/Mid/Treble Controls
for the amp
The speaker cabs... LINE6 seems to want to graft the Marshall 2x12
Celestion Greenbacks on it. They did that in both the POD and the AX2 212.
I haven't checked the Digitech version of SRV, yet, however, they have only
copied his TS-9 sound, which came later, in most of their patches.
The original cabinet, for the Vibroverb and Super Reverbs were 2x12 cabs...
mic position, that should be at the cone, rather than the edge of the
speaker. The sound that Stevie uses is pretty devoid of "bass" frequencies
and those are accentuated the further you get from the cone, on the speaker.
EQ wise, Stevie, for the most part, runs pretty flat. From all the patches
I could dig up, on the web, and elsewhere, here's about what we have. I
also ran both my POD and my LINE6 AX2 212 through an RTA (Real Time Spectrum
Analyzer) to get the EQ curves out of it... you'll have to tweak to taste
and to your personal amplifier... I used the direct outputs on each of these
to get the curves rather than using a mic and having to deal w/ standing
waves and reflections in the room.
First, the EQ curves from LINE6... they use a 6-band, graphic EQ w/ the
80Hz - 2dB
240Hz - 0dB
750Hz - 0dB
2.2kHz - 0dB
6.6kHz - 3dB
Bass / Mid / Treble - Amp Controls
65 / 52 / 30
The mid-control, on the Bassman, actually acted like an addition to the
treble control. Most of the boost/cut was in the range of about 2kHz to
2.7kHz on those things... The value of 50 is flat. The Bass control
worked in the 100Hz range. So, what we're looking at is a slight boost at
80-100Hz... flat mid-range response and a bit of "presence" at the top end.
The overall EQ curve out of the amp looked like below:
/ \-----------------/ \
The "peaks" at 125Hz and 3.15kHz were about 2dB up from the flat response
shown in the middle. Low/High Frequency roll-offs were similar to those
presented by most amplifier models, w/ the majority of the frequency
response rolled off by 6kHz on the upper end and about 80Hz on the lower
end. This is characteristic of real guitar amps.
PreDelay - 0
Decay - 65
Damping - 80 to 90
Level - 50 to 60
Basically, you want to use the reverb to "fatten" the sound, but not so much
that you can hear it... if that makes sense. Most killer guitar tones are
mostly dry tone w/ just enough effect to change the sound but not enough to
hear the actual effect, itself. That's what you're going for, here. Many
try to "wet" their overall sound too much with effects and it has the
undesired side effect of either sounding too muddy, too harsh or buzzy.
Most of his overall sound, including amps, is actually, almost flat w/ very
little EQ. Stevie relied on pickup selectors, guitar tone controls and most
importantly his playing technique to get his unique sound.
Hope that this helps... happy hunting. If you'd like I can try to dial this
up on my RP2000 and see how close this comes to the actual sound that SRV
was getting. I can help you tweak, from there.
This is what let me to wrote this question! I can't play at higher levels
where I live, so I'd like to record directly. But it sounds too digital
because the sound is too dry. I can't emulate the natural reverb of a
room and the brightness and sustain of a real amp and a mic. Do you
think if just adding some reverb to my patch will solve this?
I don't have my unit here right now to experiment with.
> Check this two links out. They are not mine, but the were all done using
> direct recording:>
I'm gonna get there.
Adding reverb or delay can add space. The timbre of the sound, though, also
plays a crucial part in how your ears percieve the "space" a sound is coming
from. The relationship (both loudness and phase) of the lows, mids, and
highs all change with distance and room.
When recording direct, I usually start by trying to visualize in my head how
big a room I want the sound to be in, and then where I am in the room in
relationship to the sound. If you start with a dry patch that you like the
basic sound of (i.e you're fairly happy with the distortion, eq,
compression, etc), try turning the delay on. Keep it a subtle as possible
(start with 100msec or less, a real short decay, and keep the effect level
under say, 35%. Use a simple delay, not a multi tap or ping pong). A good
space (unless you're trying to simulate a live recording in a stadium or
arena) is very subtle. I heard it said once that the best effects aren't
really heard until you turn them off!! I find this to be true in a lot of
cases. Anyway, the delay time, feedback, and level parameters will all work
together to very the "size" of your room. You need to get comfortable with
how the delay parameters react and interact. Basically the delay time and
feedback (decay) parameters will determine the "size" and "length" of your
virtual room, while the amount parameter will simulate how much of the
"room" your "mic" is picking up.
But wait! There's more! Once you get a resonable approximation of the "room"
you're looking for, now tweek the eq. Notice how increasing or decreasing
different bands effects the sound and you're overall perception of the
"room"? Then, try different pan positions. Moving the sound even a little
off center gives it a depth and sense of dimension it didn't have when
panned straight up the middle.
The effects of reverb are much the same, only multiplied a bunch because
there's usually more parameters, and, on the RP, you can select different
room sizes as a starting point. Oh yeah, and remember, what sounds good when
you're playing alone may not be so great in a mix. Conversley, I have found
that what sounds great in mix very often doesn't sound so good solo. That's
why I usually don't record with a lot of reverb or delay- if you record with
it, you're committed to it!
Hope this helps,
Check out this epic on eq and get schooled! Thanks Darwin!!
At the request of many... I'll put this up here(on the domo)... Hopefully, this will make sense to ya'll. If it doesn't, please, lambast me w/ questions and I'll try to make some sense of whatever I didn't make sense of the first time.
Let's start here. There are three basic frequency ranges that we deal w/ whenever we use EQ. Most of the information, in this area, applies to anything, not just the guitar.
HIGHS - Above 3.5kHz
MIDS - 250Hz to 3.5kHz
LOWS - 20Hz to 250Hz
Many times the above areas are further broken down into the following ranges by us 'recordin engineers'...
Presence 3.5 - 6kHz
Upper Mid-Range 1.5 - 3.5kHz
Lower MId-Range 250Hz - 1.5kHz
Those ranges, of course, the smaller ones, are the ones that are easiest to reference when we're dealing w/ making the guitar sound "fuller", "brighter", "buzzy", "nasally", or whatever else you're looking to accomplish w/ your overall guitar sound.
Marshall amps, for instance, have a bit of a nasally sound... (lower mid-range). They have a bit of a bump, due to the interaction of the amps/speaker cabs they use, in the 800Hz range. Fender amps, of course, have a great deal of presence and they have a bit of a bump in the 3.5 - 6kHz range.
All right, let's take a look at each of these ranges and figure out what they contribute to the overall sound of the guitar / amp / cab.
FREQUENCY RANGE OF THE GUITAR:
The lowest frequency on the guitar is the Low E string... 82.4Hz is it's frequency.
Here's a range of frequencies, on the guitar, from Low A to 1 Octave above the Low A.
A - 110.0Hz
B - 123.5Hz
C - 130.8Hz
D - 146.8Hz
E - 164.8Hz
F - 174.6Hz
G - 195.9Hz
A - 220.0
For each successive octave, just double the frequency and you'll get the frequency for the note that you're playing. Hopefully, everyone gets that... Thus, if you're going to play an E, at the 12th fret, first string, you'd quadruple the 164.8Hz (2 octaves) and come up w/: 659.2Hz.
OK. Now that we've got that straight we need to understand a bit more w/ regards to this.
The guitar is an instrument that's comprised of fundamental tones (the original frequency of the tone that you're playing) and overtones (that's what some guitarists call them). Us recordin' engineers call the overtones "harmonics". In reality, the actual "power" of the guitar sound is not in the fundamental frequency, but it's in the harmonics. OK... so explain yerself, I'm hearing ya'll say.
Here's what gives... when you pluck a guitar string, you're getting the fundamental frequency (let's say we pluck the "A" string). So, you'll be getting 110Hz. However, because the string sorta bounces really funny (look at it close, sometime), you're also going to get a bit of 220Hz, 440Hz, 880Hz, 1100Hz, etc... These frequencies are spaced somewhat closely together, thanks to the fundamental frequency being so low. Thus, when you get up into the higher registers (i.e. 2-3kHz) there are, literally, 1000's of little harmonics all together.
When you add distortion in the mix (and we haven't done that, yet), you get even more harmonics produced.
OK. Now that we've had our string vibration lesson, let's look at some things, here.
THE FREQUENCY RANGES REVISITED
Let's take another look, here, at the frequency ranges (and, by the way, these are not the "definitive" names for these ranges). You'll see various other ranges w/ the same names. Sort of depends on which recording engineer that you choose to talk w/ as to what they name each of these areas.
Presence 3.5 - 6kHz
Upper Mid-Range 1.5 - 3.5kHz
Lower MId-Range 250Hz - 1.5kHz
It's easy to see that there are no frequencies, on the guitar in the sub-bass range. In fact, most of the frequency (fundamental) energy exists in what most folks would consider the "bass" range. Well, isn't that special. Lower mid-range gets the pleasure of having the upper fretboard fundamentals in it, and that's about it for the guitar.
The "bass range" is considered, by many recording engineers, to be the "thump" range. Basically, this is the range that you hear when you're walking down the street minding your own business and a teenager, proud of their Oakleys drives by! That's, essentially, the bass range, w/ a little sub-bass thrown in to make their car chassis vibrate.
Most of the time, boosting guitar sounds in the sub-bass range only enhances what we call "cabinet thump". It adds a lot of low-end noise that wreaks havoc w/ the bass guitar and the bass drum and adds no real sound and/or dimension to the music, overall.
However, that said, a slight boost at about 100Hz, can add a bit of "roundness" and "bottom" to your guitar sound. Most recording folks, however, limit the amount of boost in this range to about 1 to 2dB, as this has the potential, again, to "fight" w/ the bass guitars upper registers. Due to the fact that you'd like to hear everything, clearly, in the mix, you may not want to boost too much of this area.
Another boost around the 250Hz range, again, adds some warmth and body to the sound. However, it's important to note that the 250Hz to 300Hz region is also consider the "MUD" region. If your guitar sound is really muddy, it's best to cut a few dB out of this range to "separate" your bass from your mid/treble ranges. Tom-toms, bass drums, bass guitar, guitar and keyboards ALL exhibit frequencies in this range and they all conflict making a HUGE muddy mess of things if proper EQ is not attended to.
It is recommended, most of the time, that you boost around 100Hz, a few dB and cut in the 250Hz range to get rid of the mud. MOst of the time, that's the rule of thumb. However, if you need a bit more "roundness" in the guitar sound, boosting here can help that out, too. Just don't boost too much.
The next range to pay attention to is the lower mid-range. This is the area that is basically what we'd call the "nasally" area. The first order harmonics (many of them) live in this range. And they, for the most part, sound a bit "honky". This is the area, though, that a slight 800Hz bump might enhance to sorta get that throaty Marshall sound (or enhance it, a bit). You don't want to boost too much, here, or you'll end up w/ a honky sounding instrument and that may, or may not, be all that great.
Ahhh... the upper mid-range... this is where your guitar sound really starts to shine and stand out. You see, the vocal range is just on the lower edge, and just slightly below, this area. Thus, you can emphasize, here, a bit more, and stand out, even when the vocals are vocalling. Isn't that just great. When you have only one guitar, a quick bump (a few dB or so) at about 2.5kHz can really make the guitar stand out and shine. Be careful, though, if you're working w/ the Fender P-Bass, or similar, cuz they, too, have a bump (presence) at about 2.5kHz. Thus, if you just can't be heard above the mix, your bassist may be covering you up - albeit, if they knew, they wouldn't do it. Thus, to remedy that situation you can boost yourself at about 2.2kHz, or 2.3kHz and get the same effect w/o fighting w/ your bass guitarist over frequency ranges.
If you have two guitar players, one of the folks would want to boost a bit around 2.2kHz and the other one boost around 2.5kHz. This keeps you both separated spectrally so everything that's being played by both, gets heard. Stops that spectral fighting, so to speak.
Above that... well... most speaker cabinets (especially the ones w/ all those 12" speakers) roll off at about 4kHz, or so. The smaller amps, w/ the smaller speakers roll off a bit higher, say at about 5kHz. The older, vintage amps... well speakers and circuitry didn't really allow for much more than 4-5kHz anyway, so it didn't matter.
When you're using digital floorboard effects... emphasizing too much about the 6kHz range ends up just adding a great deal of quantization (those are the errors that get made by your D/A and A/D converters) noise and pre-amp noise to the mix w/o adding a whole lot more - even if you're using the broadband cabinet(s). Thus you end up w/ "hissy" sounds.
BUT WHAT ABOUT THE BUZZY SOUNDS I GET FROM MY GEAR????
Well, that's one that we've all seen a few times, isn't it? You hear the "tube versus solid state" arguements all start out w/ the "solid state amps all sound buzzy and the tube amps sound warm". Realistically, that could be true. There is a bit of a difference between tube and solid state distortion. However, the reality of the matter... these new digital units are emulating tube distortion (all the harmonic content) almost perfectly to what the original amps did. Thus, if you're getting "buzzy" distortions off of your digital unit, it's time to look elsewhere. Like your EQ.
The "buzzy" sounds that we all hear (those annoying sounds like chainsaws, flies, and swarms of mosquitoes) are all centered around that irritating frequency of 4kHz. Hmmm.... try it and see... take your trusty pedal and boost 4kHz at 15dB or 12dB... whatever it does... crank up the distortion and listen to your sound... while that's highly emphasized... it's going to be very irritating (HINT: Don't turn the amp up to 11 to do this unless you're into pain).
To round that out... remove the 4kHz bump... add some 80-100Hz to warm up the sound... give it a little clarity at about 2.2kHz... maybe a bit of "body" at 800Hz... and voila, you've got a pretty good sound, overall.
USING MY BOX W/ A GUITAR AMP - HOW COME THE DARNED THING SOUNDS HORRID BUT THE AMP SOUNDS OK BY ITSELF?
This is a common malady that I've seen on a lot of users groups. The important thing to remember is that the guitar amp speaker cabinet, amp and EQ controls are all pretty much "tuned" for each other. You add your little pedal into the effects loop and take away the pre-amp of the amplifier and voila', things get a bit more muddy, tinny, buzzy, or whatever else it is that you're experiencing. Results and mileage may vary based on the amp that you're using, the speaker cabs, etc. Thus, you're going to have to EQ the heck out of all of your patches to make sure that things sound right w/ the amp that you're using, Figuring out that "magic" EQ will take a bit of practice. Ear training is a good practice... there is no one magic way to make things sound good automatically w/ the amp. One thing is for sure, though... unless you're looking for a really new sound, keep your cabinet emulators TURNED OFF! The roll-off of the cabinet emulators will add to the roll-off of the guitar amp and make things sound very dark and muffled. This is isn't a "problem" it's just due to how the pedal and the amp are working together.
Another important thing to remember is that guitar cabs are FAR from being linear (flat) in frequency response. They are, for the most part, some of the most horrible things on the planet when it comes to linearity and frequency response. Of course, that's what gives us all that great tone everyone is striving after.
QUICK EQ GUIDE - I BORROWED THIS CUZ IT'S PRETTY GOOD
Using Your Portable Studio
Amsco Publications 1996
Page 148 - The Guitar Chapter
Bottom - Boost 100Hz
Warmth - Boost 250Hz
Body - Boost 500Hz
Pick/Percussion - Boost 1-2kHz
"Cut" - Boost 3-4kHz (usually most settle on 2.2-2.5kHz)
Presence - Boost 5kHz (some use 3.5 to 6kHz many center around 4kHz)
Buzz - Boost 7kHz
Clarity - Boost 10kHz and up (shelving)
Remove mud - Cut 200-250Hz
Remove harshness (cut 1-4kHz -depending on where the harsh sound is)
FINDING THOSE "PROBLEM" FREQUENCIES
How in the world do you figure out what the "problem" frequencies are? Let's say your guitar sounds WAY too buzzy, right now, for your taste? Take one of the parametric controls (those are the ones that allow you to adjust BOTH frequency and amplitude) and set the thing to +12dB or +15 (whatever your pedal allows). Then sweep it through the entire range of frequencies while playing your favorite chord or solo note. The one that seems to irritate you the most is the "problem" frequency. (There might be more than one). Thus, when you find it, rather than boost at that frequency you're going to cut. The same goes for something that sounds really good... you can sweep w/ a couple dB of boost (maybe 2 - 3dB) and find that "sweet spot" where your guitar sounds warmer, cleaner, punchier, or whatever you're looking for. Then you just tweak the boost control a bit, here and there, to get the exact sound that you want.
Ultimately, EQ is something that you have to play w/ a bit, to get that sound that you've always looked for. Some effects, like the use of pitch shifters in Heavy Metal, Thrash, etc. aren't really EQ-based effects, though it seems like they've EQ'd the hell out of things. It's just the addition of frequencies an octave below the primary that makes it sound like that. Other sounds include a judicious "blend" of bass guitar and electric guitar to come up w/ that "deep" sound. The "big" sounds are almost always a combination of room(ambience) mic's and direct mic'ing (we can get into recording techniques, if you'd like at some point).
Matching your Digitech RP2000, GNX2, GNX1, RP12, RP6, RP300, RP100, etc. to a sound that you're hearing on an album is, virtually, impossible (HUH????)... well, realistically, you've got a great amp modeler in the RPx packages, BUT, you're missing a HUGE pile of gear that they used in the studios that most of us don't have. Also, interestingly enough, most of the guitar tracks are recorded dry, no effects, and effects are added afterwards. What that means, essentially, is that you have the dry sound of the amp... then compression, gating, reverb, delay, chorus, etc, is added AFTER the speakers... micing... speaker sims, etc.... That presents a whole new way to look at things... the Digitech boxes don't allow for you to add that stuff AFTER the cab sims (at least not now).
Hope that helps... I realize that this was just a brief overview of the concept of EQ. I'd love to spend more time, if you guys need/want it, on the recording side of things. Just please don't expect answers right away - in real time - as I tend to travel around the world a lot (no kidding, either). My job is somewhat travel intensive, at this point, so if you can be patient, I'll type some of the things I know, have tried, and have had work.
Have fun EQ'ing.
Solid State Eaten up by Crowd?
I played a show last night, and i did a soundcheck (ina empty bar) and the
levels sounded great, My clean and Distortions sounded equal. Then after
people piled in and we played the show, I heard complaints that my
distortion wasnt loud enough, that the clean sounded "Great" but you could
barly hear the distortion. A guy in another band was explaining to me that
Solid state distortions have a tendancy to get "swallowed" up by crowds, and
when u are setting levels you need to make it sound louder then the clean,
and once people pile in, it will sound even. I was wondering if anyone had
any insight into this
That is very right because people are responding as low pass filters. So your distortion which is allot of frequencies and intermodulated products are attenuated by the people. Try when you play to have the speakers of the guitar as high from the floor as possible. That will make a big difference. You cannot simply do what your friend from the other band told you because then your low frequencies will dominate and your sound will be --it. Imagine also that your bass man is playing at low frequencies!!! Just keep your speakers aloft.
Yup. I've noticed this over the years in church, as well. Here's how I combat
You're right about the volume. It needs to be a little hotter. On Digitech
stuff, I've had good results by actually cutting the distortion back some, then
using the compressor level parameter to get the volume. Back off the reverb as
well. Try a little bit of short delay instead. I always mic my amp if I can, and
use the RP into the effects loop. Works good for me...
t 15:32 22/09/01 -0400, you wrote:
>The problem with solid-state distortions is they tend to sound compressed
>and don't have a lot of the high-order harmonics that are abundant in tube
>amp setups. Solid-state can sound pretty good (depending on your
>preference) for heavy metal distortion, and arguably great for clean sounds
>but in my experience lack considerably in live situations. I remember
>cranking my amp more and more throughout the night and although loud it just
>didn't usually seem to "cut" right.
>I used a solid-state setup for quite some time and was always wondering what
>was missing tone-wise when I played live and couldn't figure it out until I
>went tube. This may not be others experience but I'm not going to get into
>a tube/solid-state debate as it really is way too subjective.
>If you've ever noticed that your guitar sounds like crap when using anything
>other than full volume (on the guitar itself) you should try this with a
>tube amp - there just seems to be a whole response thing going on with the
>guitar and tubes.
>Anyway, whichever setup you use, you will probably find that a room full of
>people will tend to "soak" up higher frequencies that sounded great during
>sound-check at the beginning of the night. Just ask your soundman (in
>bigger rooms) who probably will tell you his mix changes as the night gets
>Anyone else care to add to this?
I don`t see a technical reason for tube amps sounding better live
than solid state, but then again I´m not a technician. I can`t seem to stop
thinking about the whole tube x solid state as something that can be either
simulated or cutted out. My experience with the rp6 and its parametric
equalizer taught me that there`s no sound you can`t get with a good
There are so many different things you could check first on your
gear before changing your approach to live rigs, maybe there was some
problem with your amp (I guess heating would be the case) or with the pa
system (maybe the guy on the mixer had a couple of drinks). But certainly a
crowded place eats up the sound, distortion could´ve sounded lower because
it got confused with all the public noise, a crowd speaking sometimes sound
like distortion. A clean sound stands out a lot more.
Maybe that`s where tube helps, it has that mellow on its
distortion as opposed to the solid state`s "roar". Write us later on if you
find what was the problem. Good luck.
Another idea to chase on the solid state thing. The amount of electricity available. I have noticed that when the solid state amps get full voltage from the ac they run great but add lights, fridges, or heaters that tend to drain amperage and some times drop voltages to say 105 or lower and the solid state amps don't sound very good. I have experienced this problem. Whereas the tube amps are not as sensitive to voltage drops.
A good voltage regulator is very welcome but are about $450.These hold the ac voltage to a constant 120. Even if the input is 90 to 130. but well worth the cash if you can afford it.
Another less expensive way is to run a power cord to an outlet you know for a fact is not on the same circut as the, say, lights, house sound, neon signs, another tube amp ( they suck more power than solid state), ect this is an effective way.
Most places we play have very inadequate power sources. And usualy are on the same run as every other plug in the joint.
Awhile back, Darwin posted some specs he got from doing a spectral analysis on
the bypass of the RP2K. I took the values and, in Excel, created a plot of the
values to provide a visual representation to work from. I have used this plot to
help me overcome some of the "darkness" inherent on the RP2K. I've found it
pretty useful. If anybody else would like a copy, email me and I'll send it to
The Axeman (#(==>>
Check Out the Tunes: http://www.geocities.com/gzsmuzik/The_CD.htm
I think this begs another question.Let me set this up.Say you are using the RP2000 as a preamp into a solid state power amp.From my point of view,the *tone* of the signal is a combination of guitar and preamp.Once that is to your liking,to preserve it,it should go into a solid state amp and full range speaker setup or the sound will be colored more and have to be compensated for again.Tube amps in no way provide a flat frequency response (although some would prefer that).
I think every link in the chain has to be accounted for.So, what if you run an all-solid state RP into the guitar input of a tube amp(RP>tube pre>tube amp>guitar speaker or cab),or effects loop(RP>tube amp>guitar speaker or cab) vs. RP with tube stage (RP tube>tube pre>tube amp>guitar speaker or cab), or effects loop(RP tube>tube amp>guitar speaker or cab) vs.All -solid state RP into solid state chain(RP>s/s power amp>full range cab)vs. s/s into guitar cab(RP>s/s power amp>guitar cab).I'm sure there are other configurations but this will give you some idea of what I'm getting at.
I like to keep things simple as I am a guitarist not a sound engineer.I think we all can agree that s/s has a
flatter,more extended frequency response than tube and that guitar cabs have a limited response curve as well.I think we have to keep all of this in mind.No one's rig is going to sound like anyone else's.If I'm running all-s/s and it sounds great,John Doe with the same setup substituting two tube stages will have to compensate for the coloration of those tube stages to get the same sound.One of my points here would be that solid state modeling and simulation is designed(correct me if I'm wrong)to be accurate thru a chain that provides a flat frequency response(as solid state).
I love the fact that I can plug my guitar into the RP2k,hook up the headphones and tweak away.Once I get something I like I can run it thru the PA (EQ'd for the room) and it sounds just as great(with perhaps,slight re-tweaking of room dependant effects and EQ).As you add more variables to the mix things become more complicated.
I know a guitar player who plugged his Parker into one of those little RedBoxes,ran that signal thru a home stereo and speakers,and proceeded to rip thru some Eddie Van Halen.It sounded so dead-on I thought he was playing the CD.
I seem to be losing my train of thought so I'll cut you all some slack and shut-up.Peace. Michael Hymer
I am enjoying this discussion, and therefore have to put
my two cents worth.
I do hear and enjoy the warm sound of a tube amp verses
a solid state amp.
Howeveer, some of the newer solid state amps that try
to emulate a tube amp do sound better than some poor
quality tube amps. Before you run out and buy just
any tube amp, listen to it and compare to a comparably
priced Solid State amp.
My tube amp, out of the box, did not sound all that great.
It required modifications, a better speaker, different
values of capacitor and resistors. My Electar Tube 10
sounds great now, but many would never touch a brand
new amp with a soldering iron.
Or to put it differently, listen before you leap.
I have what might be a real stupid question!! I am completely new at using a multi effects pedal (RP100), Have always just used my amp distortion, and clean channel (Carvin 100W head). My question is,, how is the best way to go about using the effects pedal? Should you start with a good clean sound on your amp,, and go from there,, or should you cut everything down,, tone, bass, presence, reverb.. etc.... and make all of the adjustments through the pedal? I haven't had a lot of time to mess with it much yet,, but so far, Just cant seem to get any good sound out of it "yet"!! Any advice would be GREATLY APPRECIATED.
Not a stupid question. Totally valid as you will see.
The tone you get is like making a cake. It is the sum total of all that goes in before it come out of the
speaker oven. One very important thing to keep in mind is that you will have more control over what you get if you make the tone flat on the amp as well as any distortion, drive, edge, etc.
Every amp and cabinet colors the tone so no two will sound similar or even through headphones.
Now if you find that many of your patches are too bright or don't have enough bass, you can adjust it on
the amp for all of the presets. This would be for a different room or club gig. Every room and acoustic
environment will have an effect on the frequency, i.e. bass, mid, and highs. So if you go from gig to gig, I would not change every patch to adjust for this. I would use the amp or PA to compensate for what you want to hear in the room.
This is what I have found but use what I am saying just like how you make your cake. It may not be the
flavor you are looking for. This news group has users all over the world of ages that span 5 decades playing music of all styles of guitars designs anyone can imagine using equipment that has ever been made.
There are folks that play and record at home and those that play for hundreds or thousands on a regular basis.
Now, if you like, give us a tone example you are looking for so we can get more specific. What kind of pickups, guitar, strings do you use?
I suggest setting all the EQ settings to neutral or 0 and disable all distortion ect. Then run the pedal through the amp, set the Guitar volume to 80-90 % and the pedal volume to about 80-90%, let the pedal handle the effects first and the amp the volume. I use an RP200 and let the AMP/CABINET settings handle most of the sound shaping before I try to adjust the the EQ on the amp.
Hope this helps you get the most out of your pedal.
These are all great points. But, if you are in my situation where your combo amp doesn't have an effects loop, your pretty much stuck with using the standard inputs. I do how ever try to use the RP200 to create my basic sound usually with headphones so I am not influenced by the amps natural EQ bias. Then I adjust the amp's EQ to match what I heard in the headphones. This usually gives me the sound I am looking for, and whenever possible I'll run the RP directly into the house system and use a wireless monitor.
When on a shoestring budget sometimes you have to improvise, and keep trying different things to get the sound your looking for. Don't be afraid to try anything, even the stuff that shouldn't normally work right. As long as your not doing something that can damage your gear, the sky's the limit. You just might find that special sound that sets you apart for the rest.
Good luck and keep playing....
Nice idea, the EQ pedal. I'm working on a eletronic device to do that job (just because it's cheaper...), but it will only control volume...It's really an nice idea, although I wonder if it has true by-pass (in cases where you might need just volume control, not EQ)
Marlon Hiraldo <firstname.lastname@example.org> escreveu:
Felipe is right the best thing you can do is bypass the preamp section. But I must say, that if you do that whenever you gig some place other from where you setup your rig it'll surely sound different, just like Bill says. So here is what I do. If I'm at home (where I regularly play, and set my patches) I connect the guitar to the rp, then the output from the rp goes into the f/x loop return input of my amp (Marshall ValveState). If I'm some place else than my house, and I hear the need for tone control, I pull out my eq pedal (Boss GE-7) and place it between the rp and the amp. I use it for tone and level control. This last configuration might sound weird, but it does the job just fine, and you don't get whatever preamp thingy is in the amp preamp besides the eq. Hope this helps and I'll like to hear what you people in the list think.
I just wanted to thank everyone that responded to my question. I got a lot of good,, and extremely useful information, and advice!! Again thank you all,, things are sounding better already!!!
The EG thing... Ive known this tidbit but it just worked its way to the output of my brain. -- graphic analyser---
This is a rack box that hears the ambiance of a room. it has the 32 points of the sound spectrum in lights., say of the 32 point slide graphic equlizer.
Has a microphone you set up in the middle of the room, (space) you are playing, then turn on , pink noise, this is the standard reference (noise) . you know the sound as oposed to white noise, (tv off, ant races) .you'll know it when you hear it.
This pink noise is heard by the analyser and the ambiant sound of the room is displayed in the lights on the front of the unit, by being up or down from center.
much like the positions of the graphic equlizer.
you adjust the 32 point graphic eq to the analyser's lights till all the lights are flat , or centered, or as close to centered as it can be.Then turn of the pink noise and your unit is set up for the place you adjusted for....
Some rooms were made for silent movies and cant be helped. haha yes thats per channel. you, If properly setting up, you need to do this to the pa. the guitar amps the bass amps the keyboard amps ect... we just worry about our guitar amps, with an graphic eq some where o the chain.
DOD made/makes one,3 or 4 hundred bucks, Ive used one several times, i tried to buy one from ams and ms but they were always out of stock and I never did recieve one. perhaps digitech (who is also a Harman company like dod), could install a primitive one(analyser) in the gnx3? (and for free) haha
I know this doesnt help the sender in his situation. but understanding a solution sometimes helps to understand what needs to be done to help the sound.??
I know Billy Gibbons uses one of these anylisers, ( well his crew does) on his guitar chain. It works in halls churches, ect anyware sound is spuewing forth.
Your ears know the difference, the time you heard the sound of a performance and it was clear and was a pleasure to listen to. vs the time you listened to the performance and your ears said something is wrong, this sounds like crap. not the music but the sound. (Arena sound)
Its the way, you know when your listining to the sound shape and just go well I think it sounds ok. but your ears are subject to being fooled, the analyiser isnt. and you can know with confidence its correct.
so take care,
just recently used one in our church, which at present is the chapel of a
funeral home. It's been to months or so since we did the pink noise
treatment, but now that you mentioned it, our sound as a band/worship team
is much better than it was.
thought i'd share that with ya!
One important thing to note, with regards to this... this DOES NOT
compensate for room modes and resonances. The other thing to note is that
the ONLY place, in the room, that is corrected is the "sweet spot" where the
mic has been placed.
Bass energy (the really muddy stuff) "piles up" in the corners of the room.
This is what is referred to as room modes. These modes are neither reduced
or eliminated when using EQ to "flatten" room response. In fact, what
you're doing is coloring the original signal to compensate for this...
flutter echo, reverb, reflections, etc. are still going to be make a mess of
things... in fact, EQ the room this way and then walk around. Some spots
will sound awesome, other spots will sound horrid... this is due to comb
filtering and modal resonances.
The only true way to compensate room modes is to eliminate them w/ bass
traps in the corners (tuned, preferably), diffusion, absorption and
controlled reflections. This takes time, work and energy as there's really
no "room in a box" way to do this (despite what Auralex says). Spending
time to pay attention to these aspects of the room will make life a lot
simpler when it comes to getting good sound.
In an auditorium, club, etc. you're doing to have to try to flatten response
of the board and the speakers, as they interact w/ the room. But this is
different that compensating a room, for recording or practice, by putting a
mic out and trying to flatten room response...
I'm looking at maybe getting a small tube amp, has to be "cheap" with a
nice tone. Where should I start looking? I don't have a big "guitar
center" to go and try a bunch of stuff out. Looking at small wattage
for in home playing.
Shameless Music Promotion...
Hi all, In regards to a small tube amp.....I've got an Ampeg VT-60 that I really really love!! It's got 3 channels that go from a relatively good clean, decent mellow distortion for bluesy sounds, to an all out great distortion that ranks easily above marshall distortion and a tad less than messa boogie distortion. It's an all tube amp so it's got tone out the wazoo!! :o) I call it small but it's a potent little 1x12 at 60 watts all tube. The 1x12 that came in mine was a celestion vintage 30 but I don't know if it was stock or not. If you can find any of these amps you may want to look at them...they're usually relatively cheap...I picked mine up for 200 smackaroos at a pawn shop. I've seen mixed reviews on this amp though on harmony-central. It just depends on how you treat the amp on how long it will last for you...but that goes for any amp. Search ebay for this amp sometime....mine may be goin' up there soon due to the fact that I wanna purchase a mesa now that I have some money finally!! :o) I use the distortions on my amp purely and don't bother with the RP distortions and only use it for effects in the effects loop. It's a nice little amp with alot of nice features...look it up!! :o)
Question, I've already asked you all about decent "cheap" tube amps and I'm looking at Laney LC15, Crate VC508 or maybe the Pignose G40V. The rest seem too expensive. I'd like to stay at around $200. Anyway, do I really need it? I use a Crate GX15 right now and through the RP2000 it sounds pretty good. I only play at home and record direct to the soundcard. In your professional opinion - Do I really need a tube amp? Is it worth it to trade up? Will it sound "incredibly" different? If I did switch amps, I guess I'd quit using the amp sims built into the RP? Only use it for effects? I'm geographically challenged and can't "just run out and try these things". Thanks...
Shameless Music Promotion...
That depends on how you use it. If your using your RP's distortion into a clean amp you probably won't notice that much of a difference. BUT, if you use the amps distortion and use the RP strictly for effects, the difference should be pretty substantial. Especially if you use an effects loop. BTW, the Laney LC15R has reverb & an effects loop, the LC15 does not.
If you eliminate the RP from the equation and play the 2 amps side by side, I'd be willing to bet that you'll be blown away by the difference. At least with the Laney. I've never hear the other 2.
That depends on the sound you want to do. Just for information, Pantera uses SS. If you want that sound, maybe you will not need tubes. IMHO, SS are only acceptable when playing some kind of jazz ( like Roland Jazz Chorus) or Pantera sound. ALL OTHER STYLES SOUNDS BETTER WITH TUBES. Just to name a few: Petrucci, Blackmore, Hendrix, Vai, Malmsteen, Santana, Van Halen, Brian May, Al Di Meola, David Gilmour, Dave Murray & Adrian Smith, Jimmy Page, Kirk Hammet & James Hetfield, etc, etc, etc, ALL OF THEM USE TUBE AMPS.
"Geographically challenged"? Where do you live? I live in Rio de Janeiro and have the same problem. Hear samples on the web, or only listen to your CD's. You will find tube sounds there. I'm also passing for the same situation - looking for a small cheap tube amp. How do you connect your RP to your Crate? If you use RP2000 distortion, remember that it EMULATES TUBE AMPS, so it's easy to realize that the real thing should sound better...
hi don, I never owned a tube amp, so I really shouldnt give you any advice. But I
knwo many senior "tube-heads" giving me advice as I was in search of an
economy amp. Anyways, they recommended I try the PEvey Transtube. My
situation is probably worser than yours. I cannot paly loud, but the beauty
of tube amp is when you crank it up, you feel the warmtness and the
saturation of the distortion. So, if you are not able to play loud, I dont
think the differance between the tube and solid would matter. (if you gig,
however there is no other alternative but to go tubes)
One of my friend traded in his Marsall 10" soldstate combo for a pevey
Transtube combo to only practice at home. He is very happy with it. I think
it is probably around $200. Try and see if you like it.
I own a Transtube (audition 110). Maybe it should sound good if you never heard tubes. It has a great solid-state distortion BUT IT'S DEFINITELY NOT TUBES! I use preamp distortion of my RP20 and it has a 12AX7 inside it. Many people say it's not the same as the power tubes such as 6L6 (FENDER), EL34 (MARSHALL) or EL84 (VOX). Anyway, it's still waaaaaaaaaaaayyyyyyyy better than solid-state. Tubes can "feel" dynamics in a way you will never hear with ss. You play soft, they sound soft. You play harder, they sound harder. Their sound is rich in lows and mediums while ss is rich in highs and mediums. I'd like to know a "home-playable/cheap" tube amp too, if anyone knows one plz email me. Anyway, I think it's better to buy a hybrid amp (pre=>tube/power=>ss) or use a tube stomp box with a ss amp than to go for a transtube.
I use a Transtube at the pratice hall, so I dont have to lug my FenderHot Rod deluxe around. The transtube is an ok amp but like Felipe said, its definitely not tubes, so if what you want is tube tone, I wouldnt advise going that direction. The untimate tube amp would be a mono block like a Manley or a Marshall then buy a couple of 4x12 cabs and plug the rp2k directly in to the mono block and blast in stereo.
Any one try plugging the rp into those Mackie powered pa cabinets ?
I bet those would totaly blast with tone.
but these ideas arent helping with the small amp on a budget post, as these anps cost $$$
just dreaming a little.
Two small tube jobs, is what players are all useing , especialy like on tv gigs.
From the ezboard site on tube amps
Anyone have Tube combo rcommendations?
I am still in the market for a combo Tube amp to use with my RP - Does anyone have any experiences to share? Has anyone used any of these? Hows the clean channel w/ the RP2K distortions?
Mesa Boogie Nomad 45 2x12
Marshall VS265R 2x12
Marshall JCM 2000 TSL 602 2x12
Marshall JCM 602 2x12
Marshall JCM 900 2x12
Carvin MTS100 2x12
Peavey 5150 2x12
Peavey Ultra 2x12
Crate VC6210 2x10
Crate VC5212 2x12
Thanks - Gotta have a Tube and not into the "Fender" thing even though it prob has the best clean channel
Reply Re: Anyone have Tube combo rcommendations?
Hey, Man, when I play out at Mono Gigs, I use a Peavey Ultra 112, and have had Very Good performance, consistency, and (I hear ya!), Real Tone. Hot Glass is great, and I'm like you on the Fenders. Your list showed the Ultra 2x12, but thought I'd let you know the 1x12 works magnificently, and it gets a nicely "bigger" with a decent 4x12 cabinet under it!!
Peace and lint screens - - - Dave
Reply Tube Amp
I would recommend a Fender "Evil" Twin 100 Watt Tube Amplifier. It's 2x12 combo provides ample headroom, and the RP2000 performs nicely, producing very clean and warm sounds and effects. I am currently using this rig and find it to be excellent and to my expectations. Any questions can be addressed to CMikeDB@AOL.com
Reply tubes, got to have tubes
HI, ya, hot rod deluxe user (fender), I know , i just cant help myself. Outside I add two bass cabs total 4x15s and just 1x12 and this thing kicks so hard. I run the preamp just about all the way open and have a volume pedal at the preamp / poweramp inputs.
The RP2K kicks it baby. And the bottom is moveing . Its 40 watts but so loud with the RP turned on 1/4.
Reply combo recommendation
I'm using a Marshal JCM 601 with my RP2000. Lotsa options for signal routing. 60Watts, mono.
Becareful, though, on shopping for a used one--they can be flakey.
Reply Tube Amps that work gwell with the RP2000
Since the RP2000 is so flexible there will be many tube amps that will work well for you. Finding the one that will work the best requires a thoughtful approach to what you are trying to do. In no particular order the three most important questions you must consider are: how loud do you wish to play? How big are the places that you intend to play at? And what kind of style of music you'll be playing? Search out local musicians and venues that are playing the same kind of music, using similar stages and volume levels you are planning to use for some of the best quick help. If there really aren't any local musicians that are close enough in any or all of those categories for a comfortable match that makes it a little harder. But if you're a little patient you can pick up tricks from other artists that may have a totally different style but are playing at a volume you're comfortable with or they have a cool amp sound. The trick is not to be turned off immediately but to look at everything and see if there is something to be learned.
I'm playing a mix of classic and modern rock in a three piece band. As the only guitar player I have to have a versatile sound. For small practice situations I use an early 80's Fender superchamp. For small clubs a 1995 Marshall JTM 30 and for louder experiences and a 2000 Marshall DSL 401. Sometimes (outdoors) I run both Marshalls in stereo. I use the clean channels exclusively on all an adjust the inputs and masters and the RP2000 to suit my needs.
Reply Best Amp to use With Rp2000??
Hi I currently have a small practice fender amp which is nice but is getting a little old and I am looking to buy some new equipment. I am in the market to spend $700 for a processor and amp what would be the best amp to buy for this processor? I am thinking of a Marshall or peavey heard they are good amps but never tried them. Also I am planning a trip to a place in DC that I heard is HUDGE and has cheap prices. So I was also wondering how much the rp2000 cost and might be if it's a good sale?
Reply Re: Best Amp to use With Rp2000??
Dave (SirGawain21), submitted a response to a similar question in the "General Info" forum. Here's a copy of it (hope you don't mind Dave!)
"Guys, just a possibility, here. I agree with the expense of two amps being restrictive, or quite frankly impossible, for my budget. However, there are some newer products out there that offer a solution. I am currently saving for one that I've tried and LOVED. Tech21NYC makes a great little unit called the Power Engine 60. I hooked up a pair to my RP and was amazed!! First off, they're designed to be VERY sonically transparent, so the nice Modeling job the RP does actually comes through uncolored. Second, they're only about 500-600 bucks a Pair (compare that to a pair of AMPs). They sounded great, are daisy-chainable, and very simple (only controls are Level, Bass, Mid, and Treble. They work GREAT on the outs of the RP, for fairly low bucks! My current setup is the RP through a RackMount Peavey 260 (Stereo Power Amp), which was only 250, but you need to have a pair of good cabinets "laying around". Loads of Luck, and btw, Tech21 has a good look at the PE60's on their Website: www.tech21nyc.com/index2.htm Try'em out, there's a reasonable number of dealer locations out there. Loads of Luck Peace & Rock Hammers - Dave"
I've tried the PE60 with my RP2000 and I agree, it's the best solution I've seen so far.
Reply Re: Best Amp to use With Rp2000??
Randall RG100SC... 100 watts, 2x12 Celestion Seventy80 speakers, stereo....
Reply Tech21 Low-End?
Just curious if the Tech21 PE60's have any low end to them. I looked at the website and they were open back cabs if I recall correctly. Thanks in advance for any info.
Reply Re: Best Amp to use With Rp2000??
My RP2K sounds its very best, hands down, thru my old Peavey KB300 keyboard amp. Since all the amp characteristics are already present in the RP sound, you don't need or want additional amp coloring, which is what you will get with a straight guitar amp.
I also own a Peavey Prowler all tube, 1x12 combo. It is a super little amp for straight in thru a couple stomp boxes, but totally sucks with the RP2000. No low end, quite brittle and harsh sounding, thin. Again, it's trying to redo eveything the RP has already done and the results are not too great.
I would think the Tech21 amps would be good, since they are designed to be transparent. Btw, my KB300 has one 15" and a horn, closed back, and will totally kill you on the low end if you want it! (no, it is NOT muddy, either).
Godspeed =-=-=-=-=-=-=- Sedjwik =-=-=-=-=-=-=-
Reply Keyboard Amp...I agree
I have also played my RP2000 through the Crate KX100 keyboard amp (effects return), and it sounds really good. It has a closed back, ported cabinet which gives it lots of low end. Like you said, the more transparent, the better. However, I just purchased a Marshall AVT275, and I've been running the RP into the stereo effects loop. So far this setup is working, but I'm not using the amp modeling on the RP anymore, because I've been experimenting with the distortions on the AVT275. It takes some tweaking to get a really good 'growling' crunch (including adjusting my LP's volume knobs), but once you do, wow. I'm using the RP on the 'Direct' amp model w/ cabinet emulation ON. I'm also using it for effects and EQ. Overall, this setup is much better for me. I like it better than the RP through the Crate, but I'm not using the RP distortions right now.
TRY THE PA>
I've tried my RP2000 through a lot of different amplification devices
and find that it SHINES through a nice PA, especially one that has
stereo output. When you go through an amp, you can run into probs using
the amp and pickup sims. You basically have to just use the effects
only, at least with a tube amp. You also get the coloration of the
particular amp/speaker combination you're using. What you do miss is the
feedback you can get with an amp set-up. If you can get along without
that, patch into the PA. Try it out during practice one time (if you
have a vocalist or keyboard player, I'm sure you'll have access to a PA)
and you'll probably not want to go back to using an amp, as long as the
RP2000 provides everything you need sonically. One note of caution, I
caught my string on fire last week when me and my rhythm guitarist hit
our strings together. I had not trimmed the ends of my strings and we
were flailing around a bit and the untrimmed part of my strings touched
the strings on his guitar. He was plugged into his amp and I was plugged
into the PA. WOW! I've never seen a string burn like that before. Burned
right to the locking nut, but it still held! That really did freak us
out! I don't think my buddie's garage's electrical system is grounded
properly ;) Anyway, all that BS aside, if you're using a mono PA, be
sure to set the RP2000 to mono. It sounds obvious, but I made that
mistake and it didn't sound as good, until I came to my senses and
turned stereo off. Also, If you haven't tried playing with the RP2000
beyond using the presets, you're missing out on 95% of what it can do.
The presets are only a general guide. Take the time to play with the
settings and you'll be blown away by what it can do. It also sounds
pretty damn good when I go digi out into my ZA2 digital card and play
through my computer. I can play at reasonable levels for practicing at
home and still get the full balls out sound of the RP2000 in stereo.
Doesn't sound quite as good with the analog out, but still is pretty
damn good that way also. I've played at friend's houses using the
RP2000's analog out to my son's 65W/chan Sony bookshelf system (yes, I
bought the Sony extended warrantee ;)) and it rocked for the lower
volume we were looking for. It beat the hell out of turning my Peavey
Ultra down to living room volume and having the sound suffer maximally!
These are just my experiences. YMMV. Sorry, I've gone beyond what you've
asked, but that's where it took me.
Happy Holidays to all in the RP newsgroup,
Harmonics Generated by Tube and Solidstate amps
The widths of the fundamental and the harmonics, are pretty much the same. For the most part, because I'm using pure sine-waves to derive this data, the sidebands are low enough (on the order of 100 times) to be discounted out of the equation and counted as noise floor. During no measurement were the sidebands at a level that could be counted. The harmonics generated, by the distortion, were pretty much a single spike on the RTA.
In fact, the values that I provided, in the example below, already took that factor into consideration, i.e. all power under the 1/24 octave intervals that I used for sampling was integrated, for both tube and solid state. Even order-harmonics are, generally, at a much higher power level in tube-based gear than in solid-state-based gear.
I would agree with you, though, that if I were using a guitar or a complex audio source, as an input, I would need to integrate the area around the fundamental, and the harmonic, to determine overall level(s) in each area. However, that said, you're, typically, going to end up w/ the two or more fundamentals and their respective harmonics all spread out... and then the sum/difference frequencies and their harmonics spread. In most cases, the harmonic content/structure is still the same, tube versus solid state, you just have a lot more data to stare at on the analyzer.
Interesting, too, that you bring up the design issues, as well. The newer tube amps are designed to be much more linear than the older ones. They don't distort as nicely because they're smack in the middle of their load lines, rather than biased one side or the other. They're much more accurate. The tubes, too, are made to tighter specs w/ less HF leakage. All of this has made the newer tube amplifiers sound a little more "tinny" (listen to the difference between the old Fender Twins out of the 60's and the new ones, today). They're generating more odd-order harmonics... due to the fact that they "tightened" things up a bit in the overall design.
I'll add a few comments in here and then run. I've been doing spectral
analysis on tube amps, distortions, distortion modelers and a bunch of other
stuff, just for own edification, recently. So, rather than spend time w/ a
bunch of theory read in a Gerald Weber book, I'll set out what I've actually
learned through some pretty in-depth testing and research on this type of
Solid State distortion, as many of you have probably read in the "tubes are
better rags" is somewhat different in harmonic content, structure and
makeup. Basically, what this means is, it sounds different.
Most tube-amp references would have you believe that tube amps generate more
even order harmonics and solid-state's generate odd-order harmonics. That's
crap. If you do the math, odd-harmonics can and do exist. However, if you
take even order harmonics out of the picture, the fundamental can no longer
exist, either (and that's the note that you played, originally).
In reality, distortions have the following characteristics:
Even-order harmonics (2nd, 4th, 6th, etc) are fundamentally lower in power
than the odd-order harmonics. This is true in BOTH tube and solid-state
The spacing of harmonics will be multiples of the fundamental frequency
played. Thus, if your note was 150Hz, all harmonics will be spaced 150Hz
from each other.
The doubling of any frequency is an octave.
All harmonics are related by "frequency".
The level of each harmonic determines the "timber" of the instrument (not
the pitch). Timber is the way we differentiate different sounds. Marshall
amps/cabinets emphasize different harmonics than say, a Fender Twin being
driven by a Pro-Co Ratt. This difference in harmonic levels changes the
timber of the instrument, thus allowing us to hear different sounds for the
same input "tone", or fundamental frequency. That's also what makes the
difference between the sound of a guitar and a violin playing the same note.
Harmonics are all mathematically related to the fundamental (in other words,
you can take the frequency of a harmonic and divide it by the fundamental
frequency and come up w/ a whole number (no decimals).
Let's look at a short range of harmonics generated by a 150Hz signal being
overdriven. This applies to both tube and solid-state distortions.
150Hz - Fundamental
300Hz - 2nd Harmonic
450Hz - 3rd Harmonic
600Hz - 4th Harmonic
750Hz - 5th Harmonic
The 2nd, 4th, 6th, etc. harmonics are called even harmonics.
The 3rd, 5th, 7th, etc. harmonics are called odd-harmonics.
To get the full-range of harmonics generated you can keep adding 150Hz to
each result, as shown above.
The unique thing about solid-state distortion is the fact that even ordered
harmonics are at a much lower level. For example, when I did a quick
spectral analysis of a solid-state distortions in the RP12 being driven w/
150Hz, the difference between the 150Hz and the 2nd-order harmonic was about
50dB. The 3rd Order and 4th order harmonic difference was nearly the same.
The only difference, between these distortions was the amount of HF,
upper-order harmonics that got added to the overall sound. The ratio's of
the lower-order harmonics remained the same when I backed off on the levels.
Just the amount of high-order harmonic content changed.
Tube distortion is different. I sat and measured this on a couple of Fender
amps I've got. A Hot-Rod Deluxe and a Deluxe Reverb. They both had the
same results. Interestingly enough, my POD model of the Deluxe Reverb did
nearly the same thing.
In tube-based amplifiers the even-order harmonics are much closer to the
fundamental and the odd-order harmonics. There was only about a 25-30dB
difference, on the tube amp, versus a 50dB difference on the solid-state
gear. This variation (higher levels of even-order harmonics) occurred across
the entire spectrum.
Also important to note, at this point, that overdriving a tube amp, harder,
has the same effect of adding more and more higher order harmonics to the
sound. Backing off on the thing removes these... the same as solid-state.
The difference is the relationship between the even/odd harmonics.
Another important thing to note is that as the harmonics get higher, they're
going to sound more and more like the noise floor (that hiss you hear when
you turn your amp all the way up). The levels are random and they're
spaced very close together. Random fluctuations and close-spacing of
frequencies, well, that's noise!
I go through all of that to bring us to part of the reason why tube amps
stand out a bit more on the stage... In the studio, with creative EQ,
multi-band compression and other things, we can compensate so much easier.
As noted in a lot of the previous e-mails, human beings and clothing, etc.
that fill a bar are high-frequency absorbers. They literally suck the life
out of the top-end. Having worked as a sound man for a while, you have to
compensate by making the mix a lot brighter, as the crowd gets larger, just
to deal w/ the HF loss.
The second issue that we're dealing with is the background noise. As more
and more folks come into the club, talking, moving, etc. they're generating
a HUGE amount of random noise patterns that ends up being similar to pink
noise - which masks most sounds. Because much of the noise is in the
audible range, and much of it in the guitar range, we're masking some of the
sound, just due to the ambient noise generated by lots of folks walking,
talking, drinking, clinking, etc... (for a demonstration on how pink noise
masks sounds... turn on a fan and listen to a tape, or, turn a boom-box
tuned between two stations up and try to listen to another boom box right
Here's the point. The even-order harmonics are CRITICAL... as they are
lower in frequency than odd-order harmonics. Solid-states emphasize
odd-order harmonics and they have a higher frequency component that tube
sounds. That's why we're losing some of the sound in the crowd. Tube amps
bring up the levels (they don't actually emphasize them over the odd-order
harmonics) of the even-order harmonics, thus giving a bit more "power" to
the sound, overall, in the area that we can all hear at, nearer the
fundamental. A sweepable mid-range, on the mixer, should also help out with
this. They can set the Q and frequency range on it to be approximately
where you play the most so that the harmonic content, closest to most of the
fundamentals you're playing, comes through.
Adding a little "edge" to the guitar in the 2.2kHz to 2.5kHz range will make
it stand out in the mix, no matter what type of distortion you're using.
If you have a rhythm guitar player that's also playing w/ distortion, set
your boost in the 2.2kHz range and give your rhythm guitar player the 2.5kHz
(that usually works out very well). It keeps your fundamentals and
even-orders a bit stronger.
Another method to combat this "problem" w/o buying a new amplifier is to use
an Enhancer/Exciter. These boxes actually add harmonic content to the
material. If you get a box that does both, you can mix the levels, as
needed, to get the right balance between even/odd ordered harmonics.
Enhancer/Exciter gear, now, can be obtained for about $150.00. Plus, it has
the direct effect of adding extra clarity to both clean and distorted sounds
w/o muddying things up. They work great and are the cornerstone pieces of
gear on both studio and soundstage.
I hope that this has helped you guys out, a bit. I've been getting pretty
deep into studying this whole spectral signature thing and I find it
fascinating. The darned thing is, I'm still learning... This is sort of a
"chapter 1" in this stuff.
Have a good one. Questions/comments, please ask.
That is very true. But you forgot to mention the width of the fundamental and harmonic signal. If you integrate the peak you can get the power included by each order.
So it is not only the type of the intermodulating products you will get but some more other characteristics for the sound.
Generally you will see that tube amps are not longer used in systems just because they add a big level (dB) of harmonic. BUT this is sometimes good for the quitars where you can get a rich and warm sounding.
When you saturate a tube you will get a much higher levels of sound that a semiconductor amp. It is the summation of the integrals of the spectrum components.
But there are sometimes different amps(tube) that the power level of the harmonics are not linearly decreasing with frequency. In that case you get different type of sound characteristics. For e.g. your 3rd order harmonic can have higher power than the 2nd.
So it is more about design issue.
Engineers do not like working with tubes. They like transistors as they have a better signal to noise ratio. But if you play hard core and stuff you want the opposite one.
That is when you have to choose an effect. It is more about if the design characteristics match with your needs.
Excellent work Darwin and Mounasgelaki (sorry but I didn't see your name) !
Now, although I can follow what you guys are discussing, I won't pretend to have any real expertise in the technical department here so please feel free to correct me if I'm way off base.
The only thing that seems to be missing here, or just to flesh out (so to speak) the topic of this discussion is perhaps the "human" factor when comparing solid-state to tubes. Again, the end result is purely subjective as the listener can decide what sounds good to him so before you write me off as another "tube ownz" guitar-head just keep in mind that I can only really speak for my own experience as vast or limited as that may be.
Anyway, I get the impression that your tests have likely involved analysing the spectral characteristics of a single tone through these amps for the purpose of scientific comparison, however, guitars are played in real time in three dimensional space which I don't think can be discounted entirely. I'll just list my points below:
The guitar signal going to the amp often is a signal comprised of multiple strings (chords obviously and such) and other noises (muting, etc.) which affect the response of the amp and can't be totally written off as simply the characteristics of a particular guitar. Tube & SS amps will react differently to these signals.
Volume, including the attack of notes and guitar volume has a completely different effect on tube amps than solid-state. I don't exactly know how technically (maybe you guys can tackle that) but I know when I play softer on a tube amp or at lower guitar volumes the sound responds way differently.
Feedback is quite different as well. Again, I can't put it into words but it sounds more natural on tube. It almost plays itself. :)I suppose it would be pretty tough to scientifically analyse the same person playing the same thing in exactly the same way on both tube and solid-state (in real time) but I think there is a physical difference (varying voltage as you play?) between tubes and amps beyond the spectral characteristics you describe. Again, it's just...different.
Well, I just thought I'd throw that into the mix and see what you guys think. :)
You'd asked about a comprehensive volume on EQ and how it works... well, there's a lot to that, actually. If you're looking for a general overview, Sound on Sound magazine ran an article last month and is running another, this month, on that very topic. In fact, I think, they're going to be running a few more articles in the series.
If you're looking to go into this thing in depth, the best thing that I've found is the "Golden Ears" program. This thing is written by David Moulton and is a series of 8 CD's (if I remember right) that take you through EVERY aspect you could ever dream of w/ regards to EQ. It's in-depth and each CD is about 74 minutes of lecture, exercises and time. The thing even comes w/ workbooks. If you're not into being very advanced about EQ, I wouldn't suggest spending a lot of time w/ this one. But, if you want to be a master at it, by all means, get this series.
Now, more focused works, if you will:
It really depends on the topic of EQ that you're looking to cover. Here are a few books, that I have, or have read, personally, that I've found to be quite helpful. To date, I've found, unfortuneatly, no single treatise on EQ, other than the "Golden Ears" series. The rest of it I've found scattered in bits and pieces in 100's of various books on the subject.
If you're just looking for guitar gear EQ tips, tricks and ideas, here are some good references:
The Recording Guitarist - A guide for home and studio
This book is easy to read, discusses a lot more than just guitar EQ, including gear setups of a few folks, some interviews w/ guitarists as to how they got their sounds, etc. It's not very technically written, but easy to understand w/ some good information about other effects, as well.
The Audio Pro Home Recording Course - Volume 1
If you're going into recording this entire 3-book series is a MUST read for everyone. Written in a non-technical, in-depth way, this bugger covers EVERYTHING from studio setup, gear, recording/mix-down/bouncing techniques, through to mastering. It's authoritative, well-written and this stuff works!
Book 1 covers guitars, guitar EQ, guitars direct, guitar signal chains and a bunch of other stuff that works both in the studio and on the stage. Book 2 covers vocals, bass guitar, pianos, etc. Book 3 covers mix-down and mastering technique. EQ and effects specific/unique to each instrument is covered in their respective chapters. And yes, each instrument type has its own chapter in this series.
The Art of Mixing
The title doesn't necessarily imply that EQ is a part of the overall subject matter, but it is. While this book is dedicated to teaching a person how to mix-down recorded tracks for burning, the information, here, readily applies to a variety of different situations, including the live sound-stage. Loaded w/ tons of full-color pictures, this thing will have you thinking about how/where you sit in a mix faster than you can say "mix". Very well written. If you're looking for in-depth EQ-specific information, this book isn't going to give it to you, but EQ as a whole, in relationship to a song and all instruments, this is very good.
The Musicians Guide to Home Recording
Peter Mclan and Larry Wichmann
This book is CHOCK FULL of EQ settings/ranges for various instruments. In fact, that's the only reason I have this book is because of the information on the EQ settings for various instruments and their ranges. Doesn't really explain to you why it works that way, but they do, at least, give you the information that you need to make it work.
This book, if you're looking for hard-core, how it works stuff, is going to be the bomb. You'll even be able to figure out, after reading this, the problem frequencies that you're going to face in every club, just by looking at the dimensions of the club, and how to compensate for them. This is a companion to the "Golden Ears" series that I'd mentioned above. This book, however, does not specifically focus on EQ for various instruments, instead, it gives you the highly technical details about EQ and it's operation so that you can apply it any way that you want. Plus there's a ton of other useful and technical information that you're not going to get any where else, that I've been able to dig up, thus far.
Hope that helps. If you have further questions, let me know.
I thought long about posting this, cause I think it answers it self, and its
not a joke. But....
In analyzing the harmonics and waveforms, has any work been done, on these
odd/even harmonics and the effect they have on the human ear. Meaning, does
the ear discriminate between the two? would an even harmonic cause a
different response by the ear. Do different sizes and shapes of the ear
prefer different combinations of these odd/even harmonics. They must
we hear them, but does the ear "like" one over the other? does our physical
make up, size/shape prefer one over the other?
Mine likes the sound of old tube amps . :)
hey thanks, you guys are amassing. Ive printed almost every one on the
Darwin thanks for the heads up on the 6 kHz. That was right on the mark.
Shape of the ear?
With response to your question, there have been, to my knowledge, no real "tests" done, that we're highly subjective in some way, shape or form about the sensitivity of the human ear to variations in harmonic content/levels. As in many applications, the determination of what "sounds good" is in the eyes of the listener, rather than in the content of the harmonics, at times. Plus, I've been able to creatively EQ solid-state distortion and it's hard to tell the difference, once you've done that.
Even/odd harmonics, we know, are processed in fundamentally the same manner, by the ear (given that they're at the same frequency - which, in terms of the original signal, require a frequency change). There isn't anything different. So, a 3rd order harmonic at 1kHz would sound, if we could isolate it, the same as a 2nd order harmonic at 1kHz, again if we could isolate just the harmonic.
What we percieve as timbre is the balance and distribution of the harmonic content of a signal. The timbre of a tube amp basically means that even and odd harmonics are at nearly the same level, when the thing gets overdriven. Solid state amps, when driven to the same level of saturation produce even harmonics at a lower level that odd harmonics. You can tell the difference, because, of course, the timbre of each amp would be different, given the same listening room, speaker cabinets, apparent volume and EQ settings.
Many of us, who are trained musically, and some who are not, will be able to determine the difference in timbres. There are some that are what we could call "tone-deaf" that either can't distinguish, or just don't care, between the two. They just haven't picked up on the differences.
Could I be more vague and yet try to answer you question? The answer, truly is that I haven't heard of any "real" studies being done on the ability to tell the subjective difference(s) between the two types of distortions. However, most claim that they can tell.
This is a really...really silly question...but I will go ahead and ask.
What character of the sound makes the guitar sound differnt than other
>>> In reality, the actual "power" of the guitar sound is not in the
>>> fundamental frequency, but it's in the harmonics.
What do you really mean by that? Are you implying the overtones
(harmonics) are responsible for this character? I am not good in physics,
but what is "amplitude" of a sound? is it the same is "volume" or
intensity of a sound? What is "quality" of sound? Is it something to do
with the Harmonics/overtones? And is Harmonics and Overtones the same term.
I am planning to write a research paper for my Software Engineering class
about Digital Sound modeling. It shoud be fun, but i have a feeling it will
be painstaking? Does anyone have any websites where I could do some
research about "digital sound modeling"
The tone of the guitar lies in a couple of different areas.
1) The resultant fundamental tone, from plucking the string, is, basically,
a sine wave. That means it well-rounded, curved and has a period that is
very predictable. Oboes, for instance, are more of a triange wave. Other
woodwinds have square-wave, or nearly so, shapes.
2) The harmonic content of the guitar is another thing that makes the sound
unique. When you pluck a guitar string the string vibrates along it's
entire length. Let's say that you pluck a Low-E. The combination of the
string guage (how thick it is) and the string length make it vibrate at a
certain rate of speed. Low-E is, roughly, 80Hz (I don't have my frequency
reference charts in front of me this very minute, so we're approximating).
If you can picture the string bouncing up and down 80 times per second, for
a minute (and you can simulate this w/ two folks and a jump rope), if you
could look at the waves leaving the string you'd note that they rose and
fell very "smoothly". You'd also note, looking down that string, that it
appeared to be "divided". In other words, the string was not only vibrating
along it's length alone, but also the string would appear to be "divided in
1/2" and that both halves would be vibrating at twice the speed of the
entire string. Each of those halves would be divided, as well, and
vibrating at 4 times the rate of the full string, etc... Those "divisions"
that are vibrating, seemingly, on their own, are the harmonic intervals of
that string (overtones) if you will.
The fact that the overtones are all bouncing in the same "sine wave" fashion
and that the number of overtones are limited, physically, by string length,
string guage, string tension, etc.gives the guitar its unique sound - that's
why you can always tell a guitar from a piano, harmonica, etc. Nylon
strings have different laws of physics that govern their vibration (i.e.
different material, thickness, etc) and different overtones are generated.
On certain types of strings, the fundamentals are more apparent (thicker
strings, nylon, etc) due to the fact that the amount of damping (how much
the output is reduced) is greater. On strings that aren't damped as much
(higher strings) more of the overtones are present. How's that for fun?
A piano, because it's using copper strings rather than the nickel-based
stuff that we use on guitars, it's string length is fundamentally different
for the same pitch and the sound board is different is going to sound
different when a note is plucked. The string vibrates with the same laws,
but, a longer string will be somewhat tighter, generating less fundamental
and more overtones... cool, hey? That's why they all sound different.
That brings us to various guitars and thier sounds. Fenders have a specific
string length that Leo gave them back in the early days. That string
length, coupled w/ the wood he used and the design of the overall guitar
body gave it the unique sound it has. Certain woods will damp high
frequencies at a greater rate (i.e. muffle them) than what other woods do.
Thus you have differences in sound between Alder, Ash, Maple, Rosewood,
Mahogony, etc. Les Pauls are made out of different woods, have different
string lengths and hardware, etc. and, thus, damp or attenuate the higher
frequencies at a different rate of speed.
> >>> In reality, the actual "power" of the guitar sound is not in the
> >>> fundamental frequency, but it's in the harmonics.
> What do you really mean by that? Are you implying the overtones
> (harmonics) are responsible for this character? I am not good in physics,
> but what is "amplitude" of a sound? is it the same is "volume" or
> intensity of a sound? What is "quality" of sound? Is it something to do
> with the Harmonics/overtones? And is Harmonics and Overtones the same
Amplitude - For the most part you can think of this as the "volume" of the
sound. Amplitude, for all practical intents and purposes is a level
measurement. Higher amplitude means more volume, lower amplitude means
Quality of sound, as I mention it, has to do w/ the overall "balance" of the
fundamental and the overtones, together, rather than the subjective version
"that's a good quality sound".
Hope that helps.
Gosh....Thanks so much for your concern. You really shouldnt have put that
much effort into it. I can understand how hectic can be. But thanks
anyways. I have read your email several times. Yeah things are a more
clear by now. I was more interested in the software aspect of sound
modeling. Yeah you do need a processor (DSP) but the software is there as a
mediator between your sound source and the processor. I have found quite a
bit of information, and even some C++ codes on how software can simulate
pitch, vibration, ambience, amplitude and all other aspects of sound, but
the DSP is there to process it all. Whew....this porject will be very time
consuming....hehe..if I cant make it within the end of the semester, I might
write something about Windows XP...so I can BS about it as much as you
want....without even any research ;)
Hey if you are interested...I have compiled some links to review for
later...i think you might find them quite (way too much for me :)
Hey once again thanks a lot for ur help :)
Agreed w/ the software side of things being a bit more "flexible" than what
the hardware is... the problem w/ software is the fact that every time you
add a single line of code, you have to recalculate the number of processor
cycles taken by that code and change every timing loop in the gosh-darned
program... that is, if you want to keep everything accurate... You can do
it w/ processors... it's just that the coding overhead gets to be very
intensive... There's a lot of math that'll need to be done, in software,
that can be done automatically w/ a DSP ASIC...
Harmonizer (get it--Harmon---izer?)
From the Ezboard forum (Thanks Frank)
Hi,Thanks to all responses to previous questions. Here's another one! When you create a harmony patch and set it to, say, Cmaj and a 'third up', what does that really mean? So, if I play non-Cmaj scale tones what's gonna happen? What are the implications for playing this setting live I wonder.
hi, depends on what you are going for.
The blues scale is most useable especialy for beginers. The 1octive up or down is the other good place to start. If the unit has presets check out the Suboct patch, its awesome. for scarry heavy.
A little musical knowledge is useful to get on quite right.
I'm not the most knowledgeable in this, but I'll try a intro to the subject.
To set up the pitch shifter is to already have an idea in mind.
say ,, a 5th in c# minor. im useing this so we can play in E.
All major keys have a relative minor (related to them) so your lead can be played in eighter the key the song is in or the relitive minor.( well most of the time)
I used an 1-4-5 progression This is The basic blues patern In E the 1 is E the 4-A the 5-B.
The relitive minor for the key of E is C#m, so to play a blues in E and solo in C#m you set up your shifter to blues, C#, and minor. And then select oct up or down, or a harmony in a 3rd, 4th or 5th ect... Or course you can also play the lead in Em, so you would set it up that way. or of course in good old E or use just open chromatic( all the notes). but when you go and solo in each key over each chord change , the solo in E or C#m, for A -F#m, for B- G#m.
You can see there are going to be trade offs. also the other different scales are mixed into the fray. In the set up above I was useing the , whats called Blues scale, this is a pentatonic scale with the added "blue" notes. so you can use it for both scales.
There are tons(well a lot) of different scales. about 7 basic..includeing, mixolidion, Iodian ect...Indian, Equiptian, asian,.and the buttscratcher scale (the one I use) .
so sometimes the intelligent shifting is a pain in the butt. but when used effectively, sounds great (for a crappy sound use the up 2nd),. where The hell could a person use that?
so set it up the way I showed above, use up a 5th, and play Robert Palmers "bad case of loveing you" those fills you couldnt play? With the pitch shifter, set this way , your there.
so good luck I hope this gets you in the door to shifty pitches,anyway.
MP3.com - Barefootterry
Reply Re: Harmony
Hi tmccallquteduau, welcome to the forum.
To use your example, if you play note that is not in the Cmaj scale, the harmonizer will still play a note that's a third above, and it may (or may not) sound ok in the song you're playing. It depends....
Below is a snippit of a previous post I made discussing the RP2000's "Harmonizer" effect. Hopefully, this will help explain my answer to your queston.
Unfortunately, the RP100, RP200 & RP300 do not have a "harmonizer" effect, only the RP2000 does. They all have the "Pitch Shift" effect, but there's a big difference when trying to play harmonies. In any case, I'll explain both, but be prepared, it's gonna be long!
A "Pitch Shifter" takes your guitar signal, digitally transposes it (based on the interval you set, i.e. a 3rd up or a 4th down), mixes it in with the original note, and plays both back at the same time. While this may sound like it's what you're looking for, it probably isn't. That's because the interval is FIXED. Think of a chord... The simplest chord is a Triad, which consists of three notes... The "root", a "3rd" above the root, and a "5th" above the root. Depending on the key (and scale, but I'll talk about that later) you are playing in, these 3rds & 5ths may need to be "Sharp" or "Flat" for any given root. Since the pitch shifter doesn't know what key you're playing in, it doesn't know when to adjust the pitch of the harmonized note, so it always plays an absolute interval, which means that the harmonized note stays parallel to the original note being played. The result is that your lead sounds like "Chinese" music.
Here's where the "Harmonizer" comes in. Sometimes called an "Intelligent Pitch Shifter", the harmonizer works just like the pitch shifter, but knows which notes to make sharp or flat so it keeps the shifted pitches within the key that you specify. Most harmonized leads you hear in songs today are based on either the "Major" or "Minor" scale. In fact a major scale in one key is usually the minor scale for another key (example... The notes that make up the "C major" scale are the same notes that make up the "A minor" scale). So, all you need to know is what key the song is in, and what scale you want to follow, right? Wrong!
The problem: Most songs tend to drift in and out of the key they're written in (especially during the bridge, where most leads take place). I've never worked with the Eventide, but I'd bet one of the advanced features is MIDI control, so you can change the key setting (among other things) via footswitch. Another issue is that your guitar MUST be perfectly in tune (and tuned to "concert pitch"... 440 Hz for the note A above middle C). This is because the harmonizer needs to be able to interpret exactly what note you're playing and where it fits in the key it's set to. If this weren't bad enough, you need to give careful thought to the notes that you're going to play, because playing a note that falls outside of the scale the unit is set to will most likely sound really bad with the harmony added in.
If all this seems like a pain in the ass, trust me, it is! It basicly means that the harmonizer must be set to a unique setting for each song in which you will use it, and you have to map out your leads ahead of time. If you're not up on your music theory (keys, scales & such), it can be a real headache. You'd have to try out your lead using each key setting (and several scale settings) until you find one that sounds correct. I use it at certain points in leads in several songs we do, and I had to give each one it's own patch, not to mention it took quite a while to get the settings right.
I guess by about now, you're sorry you asked, huh?
P.S. Can you give me an idea of some of the songs where you think Kirk is using the Eventide (I'm not up on my Metallica)?
BTW, here's a link that has great information on triads, scales & keys:
That's what I meant, if you're messing with the amps, as soon as you turn the wheel they all go to a default. So, say you have the rect at gain 10 to clean it up, and you turn the wheel, and look at another amp, say classa, the rect gain goes automatically to 80 or something like that, dirty sounding. You need to put it back to 10 before you save it or it will be 80. And sound VERY different from what you originally had. I hope it's not just my unit doing this!
Shameless Music Promotion...
My unit does this too. It is very annoying. I have the same experience. I was looking for a good tone that matched a song I was learning. I had gain of 50 with Boutique. Then I found out Satriani plays through a Marshall. So I flipped over to the JCM 900 model and bam! My gain poped up to 85 or something. It made getting that ideal tone more difficult. I hope the GNX pedals don't do this. I don't see why the software guys would reset the gain when the model is being changed unless gain is part of their AMP class and when it flips they reload a new structure into the unit.
Speaker Hum and Other Buzz'es
hey trace the problem to isulate the hum source. Donot use shelded cable for speakers. The load is to much for them to handle and you will fry your amp.
start with the amp, plug other speakers in and see if there(it probaly isnt). then unplug every thing from the amp. If no hum its not the amp. next unplug every thing from the mixer and plug it in to the amp. if no hum check the mike cords ect..if it hums it could be the power supply (probaly its that). or a bad or crummy cord plugging it to the amp. You say hum not noise or his. hums are grounding problems or powersupplies . a fixable condition.
any way something to try
> --- dutch <email@example.com> wrote:
> > Boy do I feel like an idiot..> >
> > After reading everyone's comments about ground loops
> > and grounding all of
> > the various pieces of equipment together, I
> > remembered that when I plugged
> > in the mixing board ages ago someone had snipped off
> > the ground part of the 3 way plug
> > I replaced the end with a 3 prong plug and the hum
> > has pretty much gone away.
> > Thanks for all your comments and
> > suggestions...sometimes the easiest
> > solutions are the simplest ones.
> > Dutch
- especially when you can by one of those "ground lift" adapters at just about
any supermarket (look by the lightbulbs) for a buck and a quarter for a package of
Buzzing might be from a grounding problem. Or crossed cables. Or
Or, it might just be the nature of the beast. Ths DS_1
Get one of those 7 band DOD Eqs and roll off the top end. That'll help a
>>> florescent lights.
??? Really. I didnt know that they were allergic to floroscent lights. But
when I was playing bedisde my computer it was creating that noise. After I
shut it off, the noise cut back a lot. Well, it only happens when I dont
play and keep the DS1 on. I hope the noise doesnt get mixed into my
Yes, i did try the noise gate. I have to push the thresh hold upto 70 to
get rid of the problem, but it is impossible to play at that level.
I think it might be my cheap cable. I will try with the better expensive
cables I have. I think tony might be right about cross cables.
But thanks for helping :)
Q. But when I was playing bedisde my computer it was creating that noise. After I
shut it off, the noise cut back a lot.
You answered. Video Monitors Cause Interference. Respect a little distance or use a noise gate - or even a noise gate plug-in if you're recording in your computer.
I sometimes learn new licks on the web.
www.riffinteractive.com and www.emplive.com
are a couple of sites. Anyway when I turn my guitar
so that the pickups face to computer / monitor mine makes
an annoying hiss. If I back away or turn it away it quits. Maybe it's
the Radiation? :o Just kidding. It's something the pickups are
"...........Radiation? :o Just kidding"
Actually, Terry B., I think you're correct. The RF (Radio Frequency [?])
the monitor puts out is being picked up by the PUP. The wave is rather
direct and not very great, otherwise moving further away or turning away
from the monitor wouldn't help.
One could find an RF filter in say "AUDIO" magazine or any reputable Hi-Fi
source, although I'm not sure this would solve the problem.
You could always put your monitor in a cage of chicken wire........break up
the signal some......but who the heck would want to stare at a monitor
through chicken wire?.... not me.
p.s. "Terry B." not to be confused with "B. Terry." ;-))) For those who
got the two mixed up ....... well, that's for you!!
It's called EMI, or Electro- Magnetic Interference. It's a cousin of RFI,
which is Radio Frequency Interference. EMI is usually an annoying growl that
goes away as you back away from the source, turn the source off, of change
the angle of you axe relative to the source (in this case the source is the
degaussing coil in the computer monitor). RFI can also be a buzz or a growl,
or it can be that annoying Latino station that keeps playing Mexican Ooompah
music that you get coming through your guitar and amp (not intended as a
racist remark- I have nothing against anybody no matter what color they are-
I just can't stand polka music!! How'd the beer barrel polka get to Mexico,
anyhow? I never saw a Hispanic wearing lederhosen.....). The only real fix
for these sources of interference is a hit or miss combination of chokes and
expensive cables. Some people shield the pickup cavity on their guitars to
try and cut back on the effects. I bet Bill Bores would have something
useful to add here........
The reason I make the distinction is because, during all the discussion on
this thread, nobody has made the distinction between the above types of
interference, which travel through the air and are really hard to fix
reliably, and your plain old garden variety 60HZ hum which can be caused by
bad cables, ground loops, poor electrical wiring, dirty power, etc. Those
have a completely different set of fixes. The test for which is which is
pretty simple- if you can pretty much make it go away by rotating slowly in
a circle until you hit a spot where it quits (or diminishes greatly), you
probably have an EMI/RFI problem rather than a power/grounding problem.
All that said, I think Sonnet's root problem is that he's feeding the input
stage of the RP too hot a signal. Remember, the input to the RP expects to
see a wimpy little signal from the magnets in a set of guitar pickups, and
the signal from the distortion pedal is probably considerably hotter than
that. This is called an impedance mismatch. It will really aggravate the
noise coming out of the distortion pedal (which is inherent in the design of
the pedal). Try turning the output of the distortion pedal down. Someone
suggested a DOD 7 band Graphic EQ pedal between the distortion and the RP-
this is probably a good since, as I recall, the DOD unit had an overall
level input as well as seven bands of EQ. This would probably help a lot.
The DOD is a good unit- I used to have one and I'm sorry I sold it.
Hope this helps everybody-
The Axeman (##(===>>
gnx3 pics below
Here is a good essay Darwin wrote on the improvements in the GNX series.
The GNX is, hands-down, a much more powerful unit than the original SDISC powered units, that's for sure. They've cleaned up the sound, overall. My RP6, for the most part is a good sounding unit, however, things like the phaser/flanger produce a bit of "noise" when everything is supposed to be quiet. The RP7, I believe, had the SDISC processor and not the SDISCII, so it would do some of the same things. Even my SDISCII (in the RP12) is a bit noisy.
Overall, the RP6/7 is a more "mellow" unit than the RP12 was. The RP12 sounds "digital" if you have to pigeonhole the overall sound. The RP6/7 have a bit of "mud" to their overall output, so they are a bit better. The cab emulators aren't bad, either, for the most part, on either the SDISCI or SDISCII.
However... once you get into the GNX series of processors there's a world of difference. They have a lot of horsepower under the hood for digital effects processing and the effects are MUCH cleaner than the SDISC's of old. Four band EQ, as opposed to your current 3-band... allows you to custom-taylor the sound a lot more than before. There's also a great deal more options for Q (bandwidth) control on the GNX series which will allow you to "target" specific ranges of frequencies much more accurately w/ the band.
The effects, themselves, are more versatile. Compressor actually has settings for threshold, ratio, etc. rather than the old "single control" where all of that was done for you. Reverb, has Decay, Pre-Delay and other functions not available on the RP7 units, as well. The delay, too, is much less "noisy" and more "crisp" in sound, making the overall delays a bit nicer (I like them clean).
All in all, the processing power that you have with the GNX, in the digital section blows the RP units away, completely (including the RP2000 - although it has its good points, too).
Moving on... Amp Models, well, they're better than the RP2000 and you don't get them in the RP6/7. The cab emulators are darned good, too and you get a heck of a lot more cabs w/ the GNX2 (even the GNX1 has a lot, but fewer than the GNX2). The ability to mix/match amp models is really cool. Basically, that's starting to get where most players are, today. If you look at most setups, in the studio and on the stage, they consist of a mixture of amps to get the tone that the players want, rather than being stuck w/ a single amplifier. The GNX allows you to replicate the same types of setups. Obviously, the RP6/7 doesn't allow for that.
One of the critical "tone" derivatives, if you're playing rock, is the ability to feed an overdriven amplifier w/ a distortion pedal. Hendrix did it... most of the players, today do it, as well. The GNX1 does not allow for that. However, the GNX2 has built in emulations of most of the popular distortion pedals out there. With that, you can put the distortion pedal before the amp model... overdrive the distortion, overdrive the amp model and voila, you have more distortion that you know what to do with... I, personally, at times, use this type of thing. With my RP2000 and/or the GNX1 you'd have to purchase outboard distortions pedals to feed the thing. This allows you do that, right in the unit.
Overall, the GNX series of processors is a great buy. The things have a lot of power, etc. for the money, that's for sure. Not having to need a room full of amps/speakers is really the "alluring" part of these pedals. However, that said, if you're looking for AWESOME amp emulations this thing isn't going to give them to you. They are "processed" amps and they're not all that accurate (yeah, I owned a few vintage amps, too)... The tone controls are more like "snapshots" than actual controls. You don't get a model of the Bass/Mid/Treble/Presence knobs like you would on the real amps... rather you have to guess where the overall tonality is, on them, and then tweak the Parametric EQ to emulate what you're looking for. Unless you're a master at EQ, that can be frustrating. Plus, many times, you end up w/ the wrong EQ settings for the amp model you're working w/ and you can't copy all that well.
Amp modeling is still the best in the LINE6 series of gear. BOSS's COSM (Composite Object Sound Modeling) is the next best version, in my opinion. Both BOSS and LINE6 have done stellar jobs in creating great amp models. BOSS's GT-5 pedal sells for about the same price as the GNX, has all the same features and a ton more... Effects loop (yes and effects loop on a multi-effects processor), guitar synth for duplication of strings, bass, brass, etc... totally (and I mean totally) programmable effects order... lots more patch storage, etc. The one thing I don't like about the BOSS... actually a few things... changing patches is a horror! Manual mode... not much better... the amps that they modeled, while slightly limited (11 models I think, plus 2 acoustic guitar models), were a bit on the "dark" side... they don't have the "punch" that you'd expect to find in the amps. However, that said, they are very good, overall.
OK... so what's best... if you're looking for sheer signal processing power, quiet and pretty good emulations, the GNX series is something to seriously consider. They are ultra-quiet, powerful and have some pretty darned good presets. Amp models are "so-so" but good, as are the cabinet emulators. If you're looking for drop-dead, killer, amp models, LINE6 is the only way to go, but you give up some good things in the signal processing chain... depends on what you're really after, for the most part. For me, I use, a lot, the BOSS GT-5 w/ a POD2.0 in the effects loop. It gives me killer pre-amp sounds w/ pretty awesome effects processing power. However, it just depends on the application.
Have a good one. Hope that helps. Questions/comments? Let me know. Always glad to answer.
Well, that's about it...
For live situations, Its hard to beat the Digitech line of Gnx processors.
I feel like a Treckie with this next section. Haha
this was the first stab at Gnx3 ver.01 lousey pic but you can see the layout
(well its ver .0 2 now) ver .0 3 has different switch functions, see lower down
This is a different pic from the one in there brochure (its ver1) we'll see what the final product looks like.
GET A BROCHURE on line for the genex line!
page 10 has the original look at the gnx3.
check out page 8 it will blow you away.
stomp/amp/effects/delay/play? wheres the control switch? cant have it all.
(They have since changed the pedal assignments to stompbox,amp,choris/mod,
delay/?/tap and CONTROL....YES)
ver .03 (this pic from a web site and not to clear)
This is an awesome machine
8 tracks of digital recording! whow wonder how it outputs?(floppy? wav?) .
well its a smart mediacard!. With some Cakewalk software.
Hey, im a Man, where's my remote? hehe...
Heres a site that has sound clips from the 8-digital tracks on board the unit
DigiTech's new GNX3™ Guitar Workstation has been awarded EQ Magazine's coveted Blue Ribbon Award for 2002!
8-track digital recorder
2-track simultaneous recording
Footswitches mappable to recorder shuttle controls
JamMan2 full-featured loop recorder
GeNetX amp and cab modeling with 'Warp" function
Stompbox modeling with expression pedal
Guitar and mic ins
1/4" stereo and S/PDIF outs
You've got an excellent point Tom. I was looking to find a new RP2K for
my drummer's kid. He couldn't wait (drummers are always in a hurry, except
to show up at rehearsal :) and got his kid the GNX1. That kid is spoiled
rotten!! Anyway, got to play with it for a couple of hours. I'm impressed
with digitech's ears. The stuff that folks didn't like about the RP2K, that
I heard about and experienced on this egroup, were fixed on the GNX1.
The unit has an on/off switch. It has a power plug like a stomp box. The
tuner is accessed by pressing and holding the first two buttons. Parameters
are adjusted by knobs. I like being able to go from red to green channel
seamlessly. The amp sims sound a touch meatier. The expression pedal has a
metal post going down inside instead of the clear plastic thing.
I'm impressed so far...
whats old is new
This is the update you've been waiting for. The 1.0 firmware has been released to production. We have several production GNX3s built and many more in progress right now. We plan on shipping a small quantity by tomorrow and mass quantities on Monday. Obviously, it will take some time to get to your local dealers, so I can't say when you will see them. I would expect those in the Western states to get them by the end of next week. Thanks for waiting everyone! You'll be glad you did. And to any who have bought a competitor's product in the meantime, I highly recommend you go check out a GNX-3 before it's too late to return it. They're gonna have to pry my prototype unit out of my cold dead hands!
this is the link to the picture of the GNX3 team. (the ones who designed and took a woopin from some fanatic players who waited long and--well a long time any way.)
Ver.1.0 in pic
Pictured left to right: Jim Lambrick, Roger Mann, Paul Howard, Jim Pennock, Joe Malocha, Andrew Lovegrove, Ron Baker, Billy Clements, Josh Kapp, Mark Gerberding, John Hanson, Jeff Pace, Troy Birch
Hats off for the most excelent unit ever made. B.Terry
p.s. some if not most, of these guys, developed the older rp units
Re: GNX3 Engineer Team
I didn't want to steal the limelight. I played a very minor role with the GNX-3, so I thought it appropriate to blend into the background. Some of the other guys in the photo have some serious battle scars from the GNX-3. It took some serious engineering talent on their part to put this thing together. Frankly, I'm still amazed when I think about the unit. I can't beleive they packed everything in there for such a low price. The 8-track smart media recorder itself is probably worth the cost of the unit.
Hey digitech send me a free one for all this free advertisment?-- BTerry
also Sample Rates Why? or Quantizing
The very basic's,
HI yep Terry here, Im recording through Cakewalk Pyro
just for saving ideas and Pro tools free
for 8 tracks of demo fun, but this is useful for any audio into the aux./line input on your card. I fixed the too loud to soft issues. The computer (im assuming you have Windows or all the following will be, well, meaningless) so click the Start button on the bottom row. Then Accessories, then Entertainment, and a thing called Sound Recorder will be listed along with other stuff. Put your arrow on the sound recorder spot and right click your mouse. A little box will show up, click on Create shortcut.
A copy of your Sound Recorder(2) will be there with the original, click the left mouse button and hold the button down and drag the Sound Recorder(2) onto your Desk Top screen( the one with all the Icons on it ) , and release the mouse button. It should now be on your Main desk top screen. Close the Start stuff. You can now access the volume controls with a click.
Your sound card has 3 (trs 1\8 I think there called) jacks on it. The middle one is usually the line in.
I use the head phone out on the Rp2k unit just because I can get stereo and in one plug.
Click and open the Sound Recorder icon, a little window will pop up. Click on the Edit button and a list will appear, go to the bottom on Audio Properties, the properties will open, you click on Sound Recording (volume) and the sound control counsole (recording console) will appear. You can now close all the other windows.
Usually when your done, The input selection resets to Mic. You need to click the select box, Line In, everytime you start. Now you can click on the bar there and adjust the volume level.
Remember your output level on the RP unit should be up about a 1\4. The second line up on the sound recorder is about all you can get.
Dont let your input volume clip, or go red, that will sound bad. It takes a little pratice to get to know how to adjust this, the loudest volume, you will have in the tune, is where you need to set it. Its not like the good old vu meters, where you want to go over the line for the peeks, over the line on the digital is bad , dont let it clip at all.
On my Cakewalk Pyro, I have a line graph between an upper and a lower line, if my sound hits either one, it clips(distorts) that's a no no . so even if the whole song is recorded way under volume, if that one section that's a popper doesn't clip you got it.
By the way at the sound recorder box you can record right from Windows up to 60 sec. of sound.
I hope this gets you going and allows you to control a few things.
Digidesign.com has a free version of Protools, but you have to have a huge system. Pentium3, 256 megs mem, fast hard disc under 10ms seek time, and for the best set up you need a seperate hard drive for the audio files, thats alot to run a "free" system. Others dont require so much.
The Cakewalk Pyro, This is a CD copy program, but in its tools section, it has an audio record that lets you record anything audio,and as long as you want. In stereo. Its 40 bucks and runs on most peoples systems. But no seperate tracks or "overdubbing". Great for long speaches or preaching, and then burning it to a CD.
Now a lot of disc space is needed to record these .wav files. I have one file, that's 438megs, thats just 1 file, you will have a seperate file for each take/ instrament. most are 40 to70 megs. so a huge hard disc is a good thing. The 438 meg is the 40 min preaching from a sunday morning. Yours will typicaly be under this.
Also if you are going to do alot of mixing and CD burning and don't want any digital skips, you will need a separate hard disc for the song storage, with the program file on the other.The good thing is, you can start small and grow as needed. Have a listen to what can be done almost for free, If you own a p3 and have a little patience--- http://artists.mp3s.com/artists/cds/195/195204.html
Think about a SCSI Hard drive at 10,000rpm, (250 bucks at Fry's) 7800 rpm is the minimum, the faster the better. The scsi (sku-ze) is way better and if you get an external drive, you can take it to the studio, or the band, or send the tracks to the vocalist, handy, as theaudio, fade, ect, files are to big, (often) for 1 cd (800 megs) and definetly to big for jass drives.
Wait, dont go SCSI (unless you have very deep pockets),
check out this New, Fast, ide, ata100, 8meg buffer HD, at Western Digitals site.
as soon as you scrape up the bucks, pick one of these HD's.
They come in 100gigs, and 120, I have heard about a 150gig but whow, looseing a 150 gigs in a chash, Would try a mans soul......
check out the poop they have on it.....
Western Digital Unveils 'Super Fast' Special Edition Hard Drive First-Ever EIDE Hard Drive With 8 MB Buffer Delivers Data Faster to Users Than Ever Before, With Performance Rivaling 15,000 RPM SCSI Product, According to Industry Press
WD Caviar Special Edition 7200 RPM EIDE hard drive delivers data to users faster than any desktop hard drive has before, rivaling 15,000 RPM SCSI hard drives. Perfect for high-performance applications like games, digital video editing, and file server applications. (and Didital audio workstations)
From me__They make a pci (i call a spliter)card, for when you have two HD's allready,
so you can add(with this pci card ) two more HD's (ya baby)
You can also buy a Mastering program, for final burn. This gets all the levels the same and aranges tunes. but you dont need one for demos there just nice to have , but not a job stoper. at the digidesign/ store.com. called Master list.
******TURN OFF/ Disable, YOUR TRS Programs, LIKE VIRUS SECURITY.
or your DAW/sound recorder, will run lousey, if at all.
Hey, I'm wondering, does digital recording really sound that much better
than analog? I have a crystal sound card (like $5) and it sounds pretty
good. Should I really go out and spend like $80 to get a SoundBlaster live
with that digital interface card? Is it really worth the extra cost? BTW,
can you connect a (or make) a cable from the SoundBlaster card CD S/PDIF
input (small jack) that has a RCA jack on the other end and save the cost
of the interface card? Thanks!
Shameless Music Promotion...
If it sounds good enough to you,stick with what you have.There is
certainly a difference but everything is relative.
What do you mean by "sounds pretty good"? Have you ever recorded an audio
track with your soundcard? The main factor in judging a sound card is the
quality of its A/D/A converter, if you never recorded an audio track with
your soundcard you still did not tested its converter... BTW, Soundblaster
is great for gaming but "so so" for audio...Its converter is
16-bit...unfortunately, it's the one I have...
I think there is a new Soundblaster card with 24bit audio......soundblaster
live Audigy???? Anybody back me up on that, or am I thinking something else?
The Audigy has 24 bit playback. Not recording. Recording is still 16 bit.
I don't get why Creative did that. If you can't record and only playback in
24 bit, you need a 24 bit source. The only 24bit sources I know of are on
DVDs. Who wants to watch a DVD on their PC with 24 bit audio on crappy PC
speakers? Probably 1% of the computer world does NOT have their PC hooked
up to crappy PC speakers.
Here is something I find odd with my SB Live. I had this problem with both
the MP3+ card and the Audigy. When I play my RP through my PC via S/PDIF,
the audio is quiet compared to my backing tracks which are wave and midi
based. If I turn up the amp level parameter I can tell that the Audigy
inputs can not handle the levels as I get tremendous amounts of clipping
if the levels are similar for both the RP and backing tracks. But, when I
playback what I recorded from the RP it sounds fine. So, to clarify, the
tracks I recorded sounded louder than when I played them originally. (I
have the same problem if I use a line in instead)
I wonder if this is why Digitech created updates for the RP units that
increased the digital output. Because of some odd quirk with SB cards
which most people were using for S/PDIF input.
Just a thought.
you need to upgrade your rp2000 firmware to 1.4, which i have done and now
the audio levels are much better, although still not up to full
signal----you have to send your rp2000 back to digitech for the 1.4 upgrade
because they need to also change something in the footpedal. ("treadle
supposedly if you upgrade to just 1.3 you can do the firmware upgrade
yourself, but i do not know if that version corrects the spdif problem.
i would have to say though the difference between the analog line-in and
spdif recordings on the sblive to me are not noticable quality-wise.
excellent info concerning firmware upgrades is here:
Yes you can connect the rp2000 SPDIF OUT to the SPDIF IN of your soundcard using a standard yellow video cable.
You would need to buy a daughtercard or the livedrive accessory for your sblive though to utilize the SPDIF IN.
And you would need to upgrade the firmware of your RP2000 to make it at least 1.3.
Overall improvement in soundquality is not noticable to me, thus not worth the hassle in my opinion.
Just use the headphone out. On your unit, it has a line level output signal and its in stereo.
And the speaker simulator(cab) works very well in this situation.
Barefootterry's copy bar for spaceing things
This idea works for other DAW's also
What's Latency All About?
Latency is a frequently misunderstood factor in host-based systems. It is basically the delay incurred by audio that passes through the system (it takes some time for the computer to process it, and send it back out). This is not typically a problem until you start doing overdubs and you notice your new tracks aren't quite lining up with the previous ones. Check out some tips on dealing with latency in the latest Mbox FAQ:
Here are a few tips regarding latency issues with Mbox:
When recording into Pro Tools LE:
1) Mute recorded enabled audio track in Pro Tools. When recording on Mbox, you are actually monitoring the audio signal going to Mbox. If the record enabled audio track is not muted, you will also be monitoring the signal going out of Pro Tools back to Mbox. This will be heard as a doubling effect because you are hearing both the input and the latent output at the same time.
2) Adjust the "Mix" knob on Mbox to prevent any echo or latency sound.
3) Press the "Mono" button on Mbox. This will prevent the input signal from sounding hard panned left or right when adjusting the Mix knob on Mbox. The "Mono" knob does not affect the mix playing out of Pro Tools, just the incoming signal.
1) When you record into Pro Tools there is a latency of 164 samples. All tracks recorded into Pro Tools have 164 samples of latency, but you will not notice this with the first track unless you have a MIDI click, or a recorded click to play along with. Thus, when you record the second track/pass, you will hear latency when the two tracks are played together. Tip #3 explains how to deal with this issue.
2) Tip #1 just helps to prevent hearing the latency while recording into Pro Tools; however, there will be latency when recording into Pro Tools/Mbox.
1) After recording your second track/pass into Pro Tools, adjust the newly recorded audio track to compensate for the 164 samples of latency.
2) Select the second pass of audio with the Grabber Tool. Edit Menu > Shift. Click "Earlier" and type in 164 in the "Samples" field. Click "OK". Now, Pro Tools will locate the audio 164 samples earlier.
3) "Shift" each track or recorded pass back by 164 samples immediately after recording. If you are stacking up tracks, you definitely want them all to be in sync as each new one is recorded.
4) If you recorded your audio at the very start of the session, you will have to trim in the audio before the waveform starts to allow space for the audio to be shifted 164 samples ahead.
Now, you may ask, are all other passes 164 samples apart from each other too?
Not exactly, each recording pass will be 164 samples behind everything already recorded. If you record one track at a time, and mute recorded tracks as you go along so you're only listening to the most recent track, then yes, your final track (say, 24) will be 23 x164 samples behind the first track, at least in terms of what you're hearing. Technically, the last track is still only 164 samples behind the first track but listening to these delays building up will cause your playing in time to the tracks to start falling behind noticeably.
So if you add one track at a time and listen to all previous tracks on each recording you'll be hearing:
Track 1: On time
Track 2: 164 samples behind Track 1
Track 3: 164 samples behind Tracks 1 and 2, so play in time to Track 1 you're off by 164 samples, but if you play in time to Track 2 you're 328 samples off.
Track 4: 164 samples behind Tracks 1, 2, and 3, but the delays build up differently depending on what track you play off of.
Other sound card problems--
ok I listed it here from harmony central
Guys I just bought a Music Studio G5 pack today. Al these years I have been fiddling around with my 4 track tape recording machine.
Anyways, my problem with my Music Studio G5 is that I can record as many tracks as I want, but I cannot hear myself when I am recording something. I have tried the manual and even the help files but came up with nothing. Yes, I am recording Audio (guitar) not midi. The funny thing is it can play back all the tracks when recording a another new track, but it really does not help cause I have no clue what I am recording in the new track.
HOw could I play something and listen to it at the same time witout having to record it and then play it back??
Please help guys?
Just to let you know that the same thing happened with my Cakewalk demo. I could only listen my guitar after I recorded, but not while I was playing it.
Is it my sound card. Why cant I hear my guitar. I just want to play and it before I record something. Pls hlp me
Just wanted to let you know that the same thing happend to me with my Cakewalk demo. I couldnt hear anything while I was recording. Is it something to do with my sound card? It is full duplex. So it shouldnt happen.
Common guys pls hlp me
In my experience with cheap soundcards, you wouldn't want to hear the sound as it is being recorded because of the 'latency' of the soundcard. This is the time delay while your audio is being processed, and with cheap cards this amounts to a disturbing delay that will disturb your timing.
Some of the good cards can get this latency down to a few milliseconds, but I would still recommend that you monitor your recorded sound before the soundcard, and not after it. Some of the good soundcards or convertor box's now offer this 'zero latency monitoring' as a feature. It means that you have to split the signal somehow, so if all you have is an Audio Buddy you have a bit of a problem. I use a Mackie mixer to solve this problem, and in the meantime this is also my preamp/s.
So now I need a Mixer.
So should my set up be:
1) Guitar --> mixer --> soundcard (for recording)
2) guitar --> mixer --> soundcard
|-----> Headphoneout (for my personal monitoring)
if that is the case. How will I monitor my other tracks? Only my guitar will sound throgh the headphones, but other tracks will comeout through my computer speakers right? HOw can you set it up so that all my tracks come out throgh the same spekers or headphones?
Or should it be like this:
2) guitar --> soundcard (line out) --> mixer (line2) ---> headphones
Assuming line1 will be for my guitar input.
So the base line is I will need a mixer now. Oh boy...
So you all have the same problem then...hmmm....
Author Topic: Help guys!! I'm totally new to this digital recording
Is your guitar acoustic or electric and do you intend to use mic's? You will get best results mic'ing up an acoustic or a guitar amp,but I appreciate you don't always have that luxury.
Sometimes when I record with 1 or 2 mic's I use my Mackie mixer for preamps, and I use the insert outputs from each channel to directly feed my PC soundcard. This means I can use the rest of the mixer for monitoring and for my synths & samplers & stuff. I plan on getting a better mic preamp for this, and just using the mixer for monitoring only.
If you were thinking of plugging guitar pickups into an Audio Buddy or mixer - think again. Mic pre's and mixers usually have lo-z inputs designed for mic's and DI-boxes that use lo-z balanced cables. Guitar pickups use hi-z unbalanced (a pity, but that's what we're stuck with) connectors. Also, pure uncolored sound is not what guitar players want to hear. You want distortion and eq and effects and cabinet simulators and reverb, etc.
A popular solution is a guitar processor, such as the rps. A RP would let you hear you're playing while you record it - but you still couldn't mix in the backing track without sending that to the soundcard, so you would have to listen to that from speakers while you wore some light headpones that let the speaker sound in.
A mixer does solve a lot of monitoring problems ...
I'm looking seriously at soundcards that provide a zero-latency monitor mix because this is an important issue for people like me who want to eliminate their mixer from the recording chain ...
Oh yea, i forgot the mic pre amp's of course , thats a givin. Mics, guitar direct, things with a low line level, need a preamp. And the behringer mixers are very guiet and inexpensive. $60-$100 for 4 or 6 channels. so you could output the headphone of the sound card to one channel, panned left. Rp unit in another channel panned right. Then in to your sound card use just the right input on your software, out put the mixer to the aux input on the sound card. it will work and you can listen with just one set of phones.
The rp is a pre .so nothing is needed to up the line level Yes,the cheep sound card is true, on the delay, but you said you got nothin?
The idea mentioned is brobaly the best ,, live monitoring , put one of the headphone's from the mix on one ear and one of the rps headphones on the other ear and play that way, or headphone from the mix on one ear and a live amp for the other ear. That should do you fine.----or buy a $200 sound card or a layla or digi-001 with a pci card and a breakout box w/ pre's and phantom power. but my $20. crystal audio sound card does an adequite job of this and with the buss selected as send output, it takes the echo delay away.
but for recording on the cheep as we do (and with great results) you move with the punches, nothing is with out its quirks and untill music pays for the better units, well this does ok for now.
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
I dont use it, cause I still havent figured out how to hook it up to my
comupters. I dont think my sound card has a Midi in out port. I almost
made an attempt to buy the USB midi in-out cable. But I think they wanted
to $76 for that damn cable, which discouraged me a bit.
But hey, you have done I great job programming the Ripper. I know a little
bit of VB and C++, and I know how frustrating it can get. HOw on earth did
you program that thing....I wouldnt even dare try...ya must be genius.
What sound card do you use? Does SB live have midi in/out with it? Is SB
the best out there (man it is really expensive!!) I have soundMAX which came
with the computer. It sounds fine. I am not sure if I should buy another
one. If I did, what qualities would I look for in the sound card. Is 24
bit ADA the best out there now, and is this all that matters in a soundcard?
What is 96 hz or something? Or far easier, is there a website where I can
get all this info?
Man where is time to play when you have to do so much research....hah...
But again....great job with ripper. I am sure it is appreciated by all
RP2000 users :)
> I dont use it, cause I still havent figured out how to hook it up to my
It's not difficult. You need a MIDI adapter and 2 MIDI cables.
>I dont think my sound card has a Midi in out port.
Probably not, most don't come with one. You need a MIDI adapter that
connects to the game port. I think I have a link in the FAQ
(http://rushtabs.tripod.com/FAQ.txt) to a place that sells them for around
>I almost made an attempt to buy the USB midi in-out cable. But I think they wanted
> to $76 for that damn cable, which discouraged me a bit.
That's not bad. I want a multiport MIDI thing. I have too many MIDI devices
in my studio and not all have MIDI thru. When I want to use the RP, I have
to unplug the drum machine.
> But hey, you have done I great job programming the Ripper. I know a
>little bit of VB and C++, and I know how frustrating it can get. HOw on earth
>did you program that thing....I wouldnt even dare try...ya must be genius.
I lost a lot of hair. :) It's up in the tens of thousands of lines of code
by now, and I'm not finished yet. I program for a living. I used to do a
lot of hardware device interfaces for lab test equipment, and this is
similar. It's all in Visual C++ by the way. The MIDI implementation was
the hardest part. My next big project is a VXTi/VST effect host for Tuareg
(www.brambos.com) that'll keep me busy.
> What sound card do you use? Does SB live have midi in/out with it?
SBLive Platinum. With the LiveDrive which has the MIDI in/out.
> Is SB the best out there (man it is really expensive!!)
No way. It's actually not that wonderful a sound card (16 bit/48khz, crappy
latency with the SB drivers, PCI problems.) It's definitely an amateur
level card. Except for the sound fonts, I'd get an M-Audio Audiophile 2496
for $150. Actually, that's on my wish list. Right after the new mixer and
a bass. :)
You can get decent recordings with the SBLive, but there are better cards
out there for the same price.
>I have soundMAX which came with the computer. It sounds fine.
> I am not sure if I should buy another one.
>If I did, what qualities would I look for in the sound card. Is 24
> bit ADA the best out there now, and is this all that matters in a
I'd get one with 24 bit, 96khz recording. The Echo Mia and the Audiophile
are both reasonably priced (under $200) and quite nice. Make sure it's got
ASIO and/or WDM drivers. be sure it's compatible with your recording
> What is 96 hz or something? Or far easier, is there a website where I can
> get all this info?
where they had sound card reviews, but I can't find it anymore.
> Man where is time to play when you have to do so much research....hah...
Dunno. I have too many things to do already! That's why it took so long to
update Ripper. Too much fun stuff to do!
> But again....great job with ripper. I am sure it is appreciated by all
> RP2000 users :)
> Thanks man.
some ideas for beginers and pros
Hello RP friends,
Bill Bores here.
A 15 / 30 dual watt soldering pencil from Radio Shack will work well.
P.N. 910-3695 Price $9.99
I have about a dozen soldering irons, pencils, Weller station, miniature
torches, and the 15/30 watt unit is my most recent purchase.
Here is something that a new learner will likely have trouble. Too muchheat in
the wrong place. The heat melts the insulation, destroys semi
conductive components, transistors, IC, diodes, etc. and removes the
The problem we encounter when making our own cords, in the list above,
is melting the insulation.
Solution: Use a heat sink. A heat sink is an aluminum clip or tiny
needle nose pliers. Put on on either side of the solder joint and this
will give you a ton of time to watch the solder flow.
Follow Kevin McQuade's suggesting soldering. Get some scrap pieces of
wire and have a good time practicing.
Get a sponge, wet it, and use this to keep you tip clean. Each time
before you solder, wipe the tip on the sponge. This will remove
oxidized solder and contaminants and prepare the tip. Apply a small
amount of solder to the tip and quickly put the tip so that it touches
both pieces of work to be soldered.
Heat will flow in both directions. Keeping the tip in place, touch the
solder at the point where all three meet.
Kevin's point about the good flow is what you will have to learn from
doing this. The cooling to a solid shiny joint he mentions is the heart
of a good solder joint.
If you move either piece before it cools, it will look frosty and not
Good luck Sonnet,
The brown stuff is called flux. It helps the solder flow. It's very useful,
but it's also corrosive, so you have to use an acid brush (a small stiff
bristle brush) and some alcohol or flux remover to remove the excess when
you're done. Also, it'll give off vapor when you solder- try not to breathe
too much of it. I doubt that it's good for you!!
There have been some good tips given here, so I'll add mine. As far as I'm
concerned, you absolutely have to have well tinned wires to start with.
Tinning your wires means wicking some solder on them before you solder the
joint. It goes like this:
1. Strip the insulation off your wires back about 1/2 inch
2. Take the bare wire and twist it between your thumb and forefinger so it's
nice and tight and straight and all the strands are nicely twisted together.
3. Run some flux the length of the twisted wire (I keep some flux in an old
nail polish bottle- the little brush in the cap is perfect for this job!!)
4. Pick up your iron, hit the wet sponge with both sides of it, the put a
nice little dab of solder on the tip of the iron
5. Now, take your iron in one hand and the twisted, fluxed wire end in the
other, and run the little dab of solder along the wire from underneath,
starting just short of the insulation and going all the way off the end in
one smooth, slow, motion (not too slow or the insulation will melt!!). The
capillary action of the wire along with the flux should draw the solder up
into the twisted strands.
The result should be a nice shiny finish, with the individual strands just
barely visible beneath the solder. The wire end should be stiff, solid, and
should not break up when you take a pair of small needle nose pliers and
bend the tip of the wire into a nice little hook to go into that little hole
in the jack end. Slide the tinned wire hook into the little hole, add a
little more flux to both pieces (a small tabletop vise or clamp will free up
a hand!), pick up your iron, hit both sides of it on the wet sponge again,
tin the tip a little (you can either use solder to tin the iron tip, or you
can get a little can of tip tin, but always keep your tip tinned- it aids
good heat /solder flow, and it keeps your tip from burning out. You know
your tip is burned out when solder slides off of it rather than sticking to
it! Tin your iron before you put it down, and before you turn it off, too).
Then lay the iron along side the tinned wire/jack joint, touching the jack
right up along side the wire, add a little solder in between the two, the
watch the whole thing heat up until you see the surface tension of the
solder break up and the whole thing just flows together around wire and
evenly onto the jack (extra points if you get a nice even flow that
completely fills the hole without making a big blob!!). As soon as you see
it flow, remove the iron, then hold everything nice and steady without
moving until you see everything solidify again. Clean off any excess flux
with an acid brush dipped in some alcohol. A good solder joint will be
smooth, fairly thin, and nice and shiny. You should be able to see the wire
underneath the solder, and, contrary to the old technician's joke, "the
bigger the blob the better the job" just ain't so!!!
The Axeman (##(===>>
One of the most important things the flux does is to isolate the soldered
surfaces from the air. This discourages any oxidation in the soldering
process due to heat and ambient air reacting during heat up. Helps for
making a solid long lasting joint. Copper contacts like foil strips and
components for boards are really susceptible to oxidation and must be kept
perfectly clean when your working on them. A little finger grease or oily
residue can make a joint impossible to solder right.
Just my 2 cents in the mix here.
Rosin is brown. If you buy solder at Radio Shack, note that is is rosin
filled. So you won't have to add any. The purpose of rosin is to keep help the
solder flow easily onto the metals parts.
The big gun was a soldering gun which typically range in wattage from 120 Watts
to 240 watts and up. I have one that is 500 Watt. These are necessary for
soldering on a chassis or large connectors and wires where the mass of the metal
takes a lot of energy to get the place to be soldered hot enough to melt the
When soldering, we should make sure the solder touches the parts to be
soldered. When they get hot enough for the solder to melt, and both of them are
at least the proper temperature, a good solder joint can be formed.
Theoretically, a guitar cannot be in perfect tune. An neither can a pedal
steel guitar. A steel guitar has no intonation adjustments.
So here is a contest question.
Why is it that a pedal steel guitar which does not have intonation adjustments,
and can have the same scale length as a 6 string electric, and string gauge range, sound
more in tune and the chords more rich and harmonious?
String density is part of this tuning phenomenon, but so is the size and
design of the piano. If you were to use the same size strings on two different
piano's they would have to be tuned differently in order to 'sound' in tune.
There are personal preferences that some owners of piano's have that will
change how a piano is tuned. They might express this by saying that it doesn't sound happy
enough, or it just sounds too sad or dark. Even though the piano is theoretically in tune
with itself and properly in pitch with a tuning standard, A440 might need to be A442 or A443.
This is really an odd thing that harmonics do with our brain and psychology.
You have no doubt noticed the following:
Guitar has a new set of strings, intonation is set properly, action and everything
is great. The guitar will tune to open strings but on some songs, can be tuned
a bit better based upon the key you are playing in. This is related to
overtones, harmonics, and your particular instrument and the key you are playing in.
Personally, I just know all this stuff, tune with a tuner, and occasionally
adjust the B string a bit sharp, sometimes and it depends on my mood
and the song. Mainly, if it sounds good and pleasing that is all I care about.
Smaller digital recorders use the virtual track scheme to get higher
recorded track counts with limited processing power. It works like this:
Say you have a digital 8 track like a Roland VS 880 or something. You are
limited to 8 tracks in the final mix because that's all the unit's processor
can handle. This is fairly limiting if you want to record several passes of
the lead vocal and comp (assemble) the final track from the best bits of
several passes. So, what you do is record several virtual tracks and
assemble them into one track using the boxes editing functions. The virtual
tracks only use up storage space on the hard drive or zip drive- since you
don't play them all at once, they don't require the processing power needed
for the higher track counts.
Another use is if you want to record 16 seperate tracks and squeeze them
into the final 8 track count. Record your 16 tracks, and do some bouncing to
get the final 8 tracks. Unlike tape, bouncing to free up additional tracks
is not irreversable- the original seperate tracks will still be available as
virtual tracks, at least until you start to run out of disk space.
Clear as mud, huh?
The Axeman (##(===>>
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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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