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Sound Card Recording (mail1.15)

<< need a player?  WINAMP
 Sound Card Recording
also Sample Rates Why? or Quantizing
back to Digital Audio Workstation-- Digital Audio Workstation
also see   E.Q. & Tubes vs SS  & Recording Guitars?
 for some excelent  letters from Dar, and Free Music Hosting
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. 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---

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/  called Master list.
 Have fun,
Terry (barefootterry)
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...
 Michael Hymer
 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.
 Tom G.  
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.

from digidesign on latency here or at
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:
Tip #1:
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.

Tip #2:
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.
Tip #3:
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.
ID:7868 Created:04/24/2002

Download a D.A.W. today at--Digital Audio Workstation

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.
Thanks greendoor,
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.
oh yea.              b.Terry
  E.Q. & Tubes vs SS        to Tubes vs SS    to More EQ
back to Sound Card Recording

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'...
Brilliance >3.5kHz
Presence 3.5 - 6kHz
Upper Mid-Range 1.5 - 3.5kHz
Lower MId-Range 250Hz - 1.5kHz
Bass 60Hz-250Hz
Sub-Bass <60Hz
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.
The lowest frequency on the guitar is the Low E string... 82.4Hz is it's frequency.
E 82.4Hz
A 110.0Hz
D 146.8Hz
G 196.0Hz
B 246.8Hz
E 329.6Hz
Here's a range of frequencies, on the guitar, from Low A to 1 Octave above the Low A.
A - 110.0Hz
A#- 116.5Hz
B - 123.5Hz
C - 130.8Hz
C#- 138.6Hz
D - 146.8Hz
D#- 155.6Hz
E - 164.8Hz
F - 174.6Hz
F#- 184.9Hz
G - 195.9Hz
G#- 207.7Hz
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.
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.
Brilliance >3.5kHz
Presence 3.5 - 6kHz
Upper Mid-Range 1.5 - 3.5kHz
Lower MId-Range 250Hz - 1.5kHz
Bass 60Hz-250Hz
Sub-Bass <60Hz
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.
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.
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.
Comes from:
Using Your Portable Studio
Peter McIan
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)
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.

 Tubes vs SS             to  More EQ
back to   Sound Card Recording

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
Damage Incorp

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.
<> Mounasgelaki

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?
>Steve Rollins

See graphic analyser

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
parametric eq.
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.
Rafael Polanczyk

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.

good luck,

Hi Guys-
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 (#(==>>
GZS Muzik
Check Out the Tunes:

Thanks guys.
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.
: jreid
 More EQ
back to Sound Card Recording

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.
Thank you!!
Hello Kevin,
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?
Bill Bores
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 <> escreveu:
Hi everyone,
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.
Marlon Hiraldo
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!!!

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.
Donnie Vazquez

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.
good luck

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.
      Take care
B.Terry    or this pic       monoblock
From the ezboard site on tube amps

Unregistered User
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
Any others??
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

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.
Good luck.
Richard Wessels
Unregistered User

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?
Frank Genus
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: 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.
- Frank.

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.
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,
 Sounds & Misc Curves
 Recording Guitars?           back to  Sound Card Recording
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
all done.

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
following parameters:
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:
125Hz 3.15kHz
/--\ /--\
/ \-----------------/ \
/ \
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.
Spring Reverb
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,

 Graphic Analyser

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...