Tone Quest /

SOURCE: VETTAVILLe.NL

Tone Quest - How to create a sound like "so-and-so" artist?... 22-07-2008

There are a number of ways to create patches to emulate a specific guitarist. This Tone Quest article gives you some directions on how to achieve that.

Perhaps the quickest/easiest way is to find a patch in your Line 6 unit that is similar to the sound you are trying to emulate, make the modifications necessary and then rename/save the patch (see your manual for the specifics on how to save your patches).
You can also download thousands of user created patches at Custom Tone and install these tones on your Line 6 unit using the Line 6 Gearbox or Line 6 Edit programs.
We recommend doing a bit of research on the guitarist you are trying to emulate if you are interested in creating patches “from scratch”. The first item of interest would most likely be what specific amplifier the guitarist used, followed by speaker cabinet type and effects. Microphone type and distance from the speaker can change the sound the model quite a bit, so experiment with different options. Also take note of the guitar(s) used by the artist, as single-coil pickups sound different than humbuckers, semi-hollow guitars sound differently than solid body guitars, etc. Websites such as Guitar Geek and uberproaudio have many well-known players guitar rigs listed, and printed guitar and recording magazines often have specific diagrams and layouts (sometimes including the actual microphones used and distance from the cabinet), which may be available at your local library.

Here we have listed some well-known players documented rigs. Most of the rigs are taken from live performance situations, although many guitarists use similar gear in recording sessions. If you do not have the specific model available on your particular Line 6 gear, Gearbox will often recommend a similar substitute. The source of the information is sometimes documented at the end of the entry for further research.

Many of the item descriptions have been abbreviated to save space, but you should be able to find more information with a little research on sites like Harmony Central . Some are the common abbreviations included in the list are:
HB = Humbucking pickup
SB = Solid Body guitar
SC = Single Coil pickup
SH = Semi-hollow body guitar

NOTE: All product names are trademarks of their respective owners, which are in no way affiliated with Line 6. These product names are provided for the sole purpose of identifying the specific products that the artist(s) have purportedly used.

A Derakh/Orgy: Jackson SB (HB), FZ-2, PH-2, PS-3, CF-7, Marshall JMC2K, TC2290, Marshall 412 w/ V30s (GG.com)

A Holdsworth: Custom SB (HB) > (2) PCM 41s, (2) Yamaha 1500, (2) SDE-3K, (2) Delta Lab Effectrons, Boogie Mk I combo for Clean into 1x12, Dual rectos for solos (into Dual recto 212). (Guitar shop 1994).

A Lifeson: mid 70s: Gibson SB (HB) volume pedal, Cry Baby, BOSS CE-1, Mistress flanger, phase shifters, tape delays, Plexi Marshall, Marshall Cabs

A Lifeson: Early 80s - Super Strat , volume pedal, Cry Baby, BOSS CE-1, Mistress flanger, phase shifters, tape delays, Fender twin and Hiwatt 100w > Marshall Cabs (internet research)

A Young Gibson SG (HB) > Marshall JTM 45 or 1969 SLP reissues, Marshall 412 cabs.

Al Mckay/EW&F: Tele (SC) and ES-335 (HB) > JC 120 w/ phaser, Rat pedal, CE-2> JC120.

A Summers/Police: MXR Dyna Comp, an MXR Phase 90, an Electro-Harmonix Electric Mistress Flanger/Filter Matrix, an MXR Analog Delay, two fuzzes (believed to be an MXR and a Electro-Harmonix Big Muff Pi), and an Musitronics Mu-Tron III Envelope Follower. MXR Analog Delay into echoplexes, Telecaster, Marshall head and 412 cabs JC120 Jazz Chorus close-miked with a Sennheiser 421

B J Armstrong: Black Les Paul, Les Paul Junior, Fernandez Strat (w/ SD JB) > TS-9 > Ibanez Phaser > Boss Tremelo > Marshall 100w Plexi heads (w/ Dookie Mod w/ better preamp and biasing) > Marshall 412 w/ V30. Also has a CAE Preamp > Marshall power amp rig (Jan 2005 G1)

B Gibbons: Les Paul (HB) > 1959 Marshall Plexi through a 2x12 cab

B May: Custom SB w/ 3 SC, BM treble booster > Dunlop Cry baby > Eventide H3K > Vox AC30 w/ TB (no EQ or Cut). H3K uses Stereo flanger, 800 & 1600 ms DDL w/ no repeats (Guitar shop 1994, GG.com)

B Nowell/Sublime: Santeria: Ibanez S-470 > Boss OS-2 > Boss DD-3 (splitter) Marshall JCM 800 combo 212 and JC-120 (GG.com)

B Setzer: Stray Cats: Gretsch Semi-hollow > TS9, Roland 301 Echo unit or Echolplex, fender tremolo, 63 Fender bassman head w/ 212 cabs w/ V30 (internet)

B Whitford: Crybaby wah, Marshall guvner, Boss CD-2, Boss CE-5, Chandler Echo, Bogner Extacy head, Bogner cabinet (internet).

Coheed and Cambria (Claudio Sanchez/ Travis Stever): Gibson Explorer with EMG-81/85 pick-ups > Dunlop CryBaby Wah > Line 6 Tremolo > Bogner Uberschall and/or Mesa Triple Rectifier >Mesa 412 Recto Cab.
Clean amp: Fender 65 Twin Reverb amp

D Donegan/Disturbed: Les Paul Standard (HB) > E Ball Volume > Whammy Pedal > PH-2 > Cry Baby Wah > Mesa Triple Recto > Mesa 412 w/ V30s (GG.com)

D Grohl/Foo Fighters: Gibson SB (HB) > TR-2 > EB Volume > DD-2 > Phase 90 > BF-2 > Mesa Road King > TC G Force > Mesa 412 w/ V 30s (Jan 2006, Guitar One)

D Gilmour: Strat w/ EMGs, Tube driver, Rat, CS-2, MXR comp, HM-2, GE7, univibe, 2290, PCM 70, MXR rack DDL, Hiwatt head, Marshall 412, Doppola (miked w/ u87 and SM57s) (Guitar Shop December 1996).

D Mustaine: ESP signature Solid bodies w/ SD HBs > Roctron Prophesy > Marshall 100/100 (EL34s) > Marshall 412 w/ V30s (May 2005 G1)

E Johnson: Strat w/ Dimarzio HS-2>fuzzface>MXR DDL/Echoplexes> 66-68 100w Marshall plexi > Marshall 412 (Guitar Shop, October 1996, September 1997)

E Van Halen: Wolfgang SB (HB) > SD-1 > OC-2 > Phase 90 > Cry baby Wah > 5150 > (FX loop H3K > (2) Roland SDE-3K > PCM-70) > Peavey 412 w/ 75w speakers (Guitar Shop, Spring 1994)

Edge/U2: (early years) Strat > Big Muff > SPX 90 > Rev 7 > Boss Pedals > Memory Man > Vox AC30 (july 87 Guitar World).

Korn: (composite of both rigs): Ibanez 7 string SB (HB)> Dunlop Wah > Whammy Pedal > Univibe > Boss PH-2 > Boss CE-5 > TS-9 > Big Muff > Rocktron tremolo > RV-3 > Mesa Triple recto > Marshall 412 w/ V30s (GG.com)

H Garza (Los Lonely Boys): Fender Strat (SC) > Vox Wah > TS808 SPLIT: Marshall 2K (w/ Fender reverb unit) > Tone Tubby 412, Fender Twin w/ Jensen speakers (Holiday 2005 G1)

In Flames (Björn Gelotte/ Jesper Strömblad): Gibson Les Paul Custom with EMG 85 active HB, Peavey 5150 amp > Line 6 Effects> 412 cab (internet)

Iron Maiden (D Murray/Adrian Smith)
84 tour book
Superstrat > MXR EQ > Boss FA-1 Preamp > MXR Dist + > MXR Phase 90 > Cry baby Wah > Yamaha analog delay > DOD Flanger > Marshall JCM 800 > Marshall 4x12 w/ 75w speakers
Guitar world February 07 (D Murray)
Super strat > JMP 1 > Marshall JFX > JCM 2K (power amp) > Marshall 4x12 w/ 75w speakers (for power amps)

J Christ/Danzig: BC Rich SB (HB) > VHT Pitbull Classic Head, Rocktron Intellifex > VHT CHromeface Power amp > Marshall 412 cabs (Guitar Shop 1994)

J Cantrell/AIC: 5150 stacks and/or Mesa Dual recto. Marshall 412 w/Celestion 30-watt Vintage and 25-watt Greenback speakers. Fender Twin for Clean tones. Dunlop Crybaby wah, ProCo Rat and a vintage Electro-Harmonix Big Muff Pi.

J Frusciante (RHCP): Rig 1: 50s or 60s Fender > DS-2 > Big Muff Pi > Phase 90 > Ibanez Wah > CE-1 SPLIT to Marshall Major 200w head and Marshall Silver Anniversary head > Marshall 412 cabs. (GG.com)

J Frusciante (RHCP): Rig 2: 50s Gretsch Falcon > DS-1 > Fender showman head > Marshall 412 (GG.com)

J Greenwood (Radiohead):
RIG 1: Fender Tele Plus > Marshall shred master > BOSS RV-3 > SD-1 > Volume > Fender Deluxe (SS)
RIG 2: Fender Tele Plus > Marshall shred master > BOSS RV-3 > SD-1 > Volume > Fender Deluxe > Whammy pedal > Trem (home made) > DOD envelope filter > EH Small Stone > Space echo (no model) > Mutron Mutator > Vox AC-30.

J Hendrix: Late 60s Fender Strat (SC) > Vox Wah > Arbiter Fuzz Face > Octavia > Univibe (w/ pedal) > Marshall Plexi 100w > Marshall 412s (Guitar shop 94)

J Hetfield/Metallica: ESP SB (active HB) Mesa Boogie Mk IIC+ , Roland JC-120 combo. Celestion Vintage 30s (web research)

J Jorgenson/Hellecasters: G& L ASAT (tele style) Ibanez TS5, TS 808, Boss Dimension C (preset 2), Boss DD2, Boss RV-2, Matchless SC-30 (Guitar shop 1995)

J Petrucci/Dream Theater: Custom Ibanez SB (HB) > Mesa Triaxis > DBX 166 > TC 2290 > PCM 70 > Mesa 2:90 > Recto cabs (Guitar Shop, Summer 94)

J Satriani: Superstrat > Cry Baby wah > Whammy Pedal > Boss DS-1 > CH-1 > DD-2 > (2) Chandler DDLs (450-550 ms, 600-800ms) > Marshall Anniversary head > Marshall 412 w/ V30s. (Guitar Shop Summer 94, Nov 97, GG.com)

J Winter
Early rig: Fender Mustang > Fender Twin
Later Rig: Gibson Firebird (HB, open D tuning for Slide) > CE-2 (rate = 10 pm, depth = 2 pm, always on) > Musicman 410 w/ Celestion Vintage 10 (all tones @ 0, treble at 10, no reverb) (April 2005 G1)

Jet
C Muncey: Gibson flying V (HB) > TS-9 > SD-1 > Boss Tremelo > 50 Marshall head > Marshall 412 (March 2005 G1)

N Cester: Gibson ES 335 (Semi hollow, HB) > Z-Vex Fuzz Factory > Hot Cake distortion > Hiwatt 50w Custom > marshall 412 (March 2005 G1)

K Cobain: Fender "Jag-stang" (HB) >Boss DS-2, Sansamp (as distortion) >EH Poly Flange>EH Ply-chorus (heavy rate and depth)>Mesa Studio or Quad Preamp>4x12 cabs

K Thayill/Soundgarden: Guild SG w/ HB > cry baby > DOD FX 10 preamp > CE-2 > Mesa Dual Recto head > to Mesa cabs or Fender Leslie (Guitar Shop 1994)

KW Sheppard: Late 50s/ early 60s Strats (SC, 11-58): Dunlop CB wah > TS 808 (modded) > TS9 (modded) > Octavia > L6 DM4 > L6 DL4 > Alalog Man chorus > SPLIT: Blackface Twin reissue (EVL speakers), Fuchs 100 OD supreme > Fuchs 410 (v30s), Marshall 1959 SLP reissue > Marshall 412 (V30s) (July 2005 G1)

Lenny Kravitz: Gibson SB (HB) > rack splitter > OC-3 > Fat boost > Dynacomp > Phase 90 > L6 Echo Pro > L6 Mod pro > L6 Filter Pro > L6 POD pro SPLIT > (2) Deluxe reverbs > 112 Greenback, (2) Marshall 50w Plexi Heads > Marshall 412 w/ 25 greenback, (2) Twin Reverbs > 212 80w G12s. (November 2005 G1)

M Izinger/Incubus: Pardon Me: Semi Hollow PRS > H&K Rotosphere > (2) PH-2 > DOD Gonlulator (ring modulation) > RV-3 > Phase 90 > DOD FX25 (envelope filter) > CS-3 > DOD FX75 Flanger > GeE7 > OC-2 > DOD overdrive > Mesa Dual Trem-o-verbs (1 clean, 1 distorted) > mesa 212 or 412 cabs w/ V30s (Sep 2002 Guitar One, GG.com)

M Schenker: Flying V SB (HB) > Dunlop Wah (rack mounted) > DD3 > CE-5 > JCM800 50w (dist) SPLIT JCM 2k (clean) > Marchall 412 w/ 75w Celestions (June 2005 G1)

Maroon 5
A Levine: Gibson SB (HB) > L6DL4 > TS9 > NS2 SPLIT: Fender tone master and/or Triple Recto > Marshall 312 w/ 75w celestions.
J Valentine: Fender SB (SC?) > EB Volume > wah > L6DL4 > Phase 90 > Fulltone Deja Vibe > Z Vex SD Fuzz SPLITTER: Tonemaster 100, %13 FTR 37, %13 RSA 23, Two Rock Custom Reverb > %13 112 (V30) and %13 212 (V30 and Alnico Bulldog). (October 2005 G1)

N Schon/Journey: Les Paul in Crybaby Wah, Boss DS-1, Roland DDS, and Lexicon reverb Hiwatt head w/ 412.

T Dumont/No Doubt: Gibson Explorer or V (HB) > Cry baby Wah, Dunlop tremolo, univibe > Fender Tone Master head, Mesa Dual Recto head > (2) Fender 2x12 and (2) Fender 4x12.

O Rush: custom strat > Mesa Mk III > Mesa 412 cab (Guitar Shop Winter 1994)

P Hamilton (Helmet): ESP SB (HB) Vox Wah > MXR Dist+ > Dyna-comp > Tech 21 double drive > Z Vex Fuzz Factory > ProCo Rat > PE COB > octave Pedal > MSC Microamp > MXR Bass octave > Whammy pedal > ADA FLanger > MXR Stereo chorus > DD6 > VHT Pitbull Head > VHT 412 w/ Eminence 50w (September 2005 G1)

R Cray: early 60s strats & Gibson 345, (2) Fender Super Reverbs
Fender Twin Reverbs (web)

R Rhoads: Les Paul > MXR Distortion plus > MXR EQ (heavy midrange emphasis)> MXR Chorus > MXR Flanger > Korg Echo unit > Cry Baby or Vox wah > Marshall 1959 Super Lead Plexi heads > Marshall 4x12 cabinets with Altec speakers. Shure 57s close-miked on cabinet, Neumann U87 20' away.

S Gossard (Pearl Jam):
1: Les Paul > TS-9 > mid 70s Twin reverb (EV 75w speakers)
2: Les Paul > TS-9 > Matchless HC30 head > Marshall 412 w/ 25w speakers (Guitar Shop 1994)
3: Les Paul > TS-9 > 1968 Marshall Plexi Super bass > Marshall 412 w/ 25w speakers (Guitar Shop 1994)

S Ian: Among the Living: Jackson guitar > TC Distortion > JCM 800 > Marshall cabs.

S Seals: Gibson ES-335 > mid 60s Fender Super reverb (Guitar Shop 95)

S Stevens/Billy Idol: Hamer SB (HB), cry baby wah, Rat dist, Phase 90, Boss CS-2, 69 Marshall Marshall, PCM 41, PCM 70, XPS 90, Eventide 969(?), Marshall 412 w/ 25w speakers. (July 87 Guitar World)

S Vai
Guitar Shop 1994
Ibanez SB (HB) > SD-1 > Cry baby wah > Whammy Pedal > JCM 8/900 > SDE 3K > H3K > Marshall 412 w/ V30s

Guitar Shop 1995
Ibanez SB (HB) > DS-2 > Cry baby wah > Whammy Pedal > Laney and Bogner xtacy > SDE 3K > H3K > H4K > Marshall 412 w/ V30s

December 2005 G1
Ibanez SB (HB) > Morley Volume > Digitech whammy > DS1 > TS9 > Phase 90 > Phase 100 > Carvin Legacy amp (loop w/ Eventide Harmonizer®, TC G Force) > Carvin 412 w/ V30s.

Slash: Les Paul (HB) > EMB Remote Wah > Marshall Anniversary series head, Marshall 412 w/ V 30 speakers (amp settings From Guitar Shop 1995, June 1996)

Slayer (composite): Jackson w/ EMG HB > Bogner Shark Preamp, Eventide H3K, Boss EQ, VHT Power amps (Wet), Marshall JCM 800s (6550s) (Dry) > Marshall 4x12 w/ 75w Celestions (Guitar Shop 1995)

ESP SB (EMG pickups) > Dunlop Wah > Boss RGE-10 EQ > SPX 900 > Eventide H3K > JCM800 w/ 6550 > Marshall 412 w/ V30s (August 2005 G1)

T Delong/Blink 182: Superstrat SPLIT to Marshall JCM 900 (clean) and Mesa Triple Recto (distortion) > Mesa 412 cab (GG.com)

T Morello: RATM/Audioslave: Superstrat w/ EMG humbuckers> crybaby wah> Whammy Pedal > Boss DD3 > DOD EQ > Ibanez digital Flanger > Marshall JCM 2205 50w > old peavey Cab (November 97 Guitar Shop)

T Petty: Tele or Rickernbacker (SC)> Vox Wah > Red Llama OD > TS-9 > CE-2 > RV-3 SPLIT TO: Fender Bassman and Vox AC 30 (GG.com)

Y Malmsteen: Strat w/ HM3 pickups > BF-2 Flanger > Marshall JMP 50w MK II heads > KORG DDL (DDL and Chorus), Marshall 412 w/ 25w celestion speakers. (GG.com, Guitar Shop, Winter 95)

Z Wylde/Ozzy Osbourne, Black Label Society: - Les Paul w/ EMG HB > Dunlop Rotovibe > Crybaby > SD-1 > CH-1 > JCM 800 > Marshall 412 w/ 75w speakers. (Guitar Shop 94, GG.com)

Source: Line 6 / vettaville.nl

Tone Quest - Double Tracker 1... 09-01-2005

DOUBLE TRACKER

The Original Vetta Double Tracker Examined 

I have been surprised at how disappointed some are about the Vetta's Double Tracker. Some claim that they cannot hear the effect, or that the effect does not work right. Whether you're a novice or a pro, it has been my musical experience, that when someone claims they cannot hear a particular effect, it is simply because their musical ear has not yet developed the ability to distinguish the effect. And how could anyone say the DT doesn't work "right" without actually knowing how Line 6 engineers define "right" according to the intended design. Therefore, it would be unfair to harshly criticize Line 6 or the Vetta for an effect that one doesn't fully comprehend.

Now I'll be the first to admit, having already understood the recording production technique of double tracking, what I had envisioned in my head was going to be this miraculous, 100% accurate simulation, just like it sounds on a CD effect...but it wasn't. But then again, I realized my expectation was un-realistic, because no one else has attempted to simulate this effect to this degree in an amp before. So technically, there was really nothing to compare it to other than some "ideal" I had floating in my head about how I thought it was supposed to sound, especially in comparison to real double tracked guitars.

I suppose I was really expecting the DT to make a SUPER HUGE difference in the sound, or make one of the modeled amps sound "exactly" like a separate guitar player was playing (doubling) a track alongside my playing. After experimenting with the DT, and thinking realistically about it...double tracking is a big call to try and reproduce with DSP (digital signal processing) technology. The sheer fact that the Vetta comes as close as it does to simulating double tracking is extremely impressive. And now that I've worked more with it, I'm so glad the effect was included. The Double Tracker is another digital tool in the toolbox that can be used...or not used...but at least it's an additional tool of choice.

"the Double Tracker is another digital tool in the toolbox"

For those who may not already know, "double tracking" is a popular studio technique which has been used extensively in most styles of music in order to create a thicker and more interesting guitar sound and image in the final mix. This is the process of recording a guitar track, then having the guitarist record to a second track while listening to the first and duplicating it. When the two tracks are played back together, the result is a slight "chorusing" and fattening of the signal due to minor pitch, timing, and dynamic (volume) differences between the two performances. In addition, double tracking helps to create a "stereo" guitar sound when those individual tracks are panned to different positions within the stereo field. And the Vetta works much in the same way. By allowing you to separately adjust the panning position for both AMP1and AMP2 anywhere in the stereo-field, from hard-left to hard-right, and applying the Double Tracker, you can actually create a stereo image that sounds like two separate guitar tracks.

How can you accurately simulate a second guitar player or part?

The reason double tracking works as well as it does in multitrack recording, is because of the inherent differences that occur when playing the same guitar part multiple times. I'm not referring to melodic differences, but the subtle performance differences of: timing, dynamics, and pitch. In the real world, there is just no way to play the same guitar part twice and have both parts be absolutely identical. So Line 6 obviously faced a tremendous challenge when they decided to include a double tracking effect. But how does one go about trying simulate a second guitar player/part in real time? Answer...with proprietary digital technology...the Vetta Double Tracker.

Obviously, it is extremely challenging to manipulate an incoming guitar signal, digitally apply aspects of the "human factor" and mirror what occurs in the real world during a guitar players performance. You can easily split the incoming guitar signal into separate paths, but these are still identical clones. Even if you send these cloned signals to separate amp models, such as the Vetta does, and apply different effects and equalization, you will still be missing the three most fundamental differences: timing, dynamics, and pitch. Without these three, tonal differences alone are still not enough to produce a signal that really simulates studio double tracking.

Let's take a deeper look into the "major" differences that occur.

TIMING DIFFERENCES: First, there is no way to play perfectly in sync with another guitar player or pre-recorded guitar track. When jumping from chord to chord, or note to note, there will always be subtle timing differences (in the milliseconds) occurring that differ each performance. The sloppier your timing is, the more dissimilar the two guitar parts tend to sound from one another.

DYNAMIC DIFFERENCES: Second, there is no way to identically match the picking and strumming dynamics of a guitar performance. There will always be times when you strum chords or play notes with varying degrees of intensity. All those subtle little volume differences add up to distinguish both parts. Also consider, that different amps, pickups, and strings, all react differently and will impose their dynamic effect upon the signal as well.

PITCH DIFFERENCES: And thirdly, there is no way to have absolute identical pitch occurring at all times. There will always be micro-differences of pitch occurring between each performance. Differences of pressure applied against the fretboard, intonation differences between guitar necks, and differences in string tuning all play their part in creating pitch differences.

TESTING PROCEDURE: For all you tech-heads out there, here's a detailed explanation of how the testing took place. In order to more accurately determine what kind of signal processing the DT applies, I thought it best to start with a known constant. In this case, the known constant was the input test signal, which was created in Sound Forge to be a "mono" square wave impulse response, with a set amplitude (volume), a set impulse length (250ms), and a set duration of silence (250ms) between each impulse. What this gives you, is an easy to see waveform that looks similar to bar graph (see illustration below). If the DT processes both AMP1 and AMP2, then we should be able to easily see any changes to our test signal.

OBSERVATIONS: Keeping all the above background theory of timing, dynamics, and pitch in mind, let's take a look at the DT's effect upon our test signal. We can draw some tremendous insights about what is taking place. First, the DT apparently uses some form of input threshold sensing, because it appears to react to the incoming signal dependent upon how it sees that incoming signal. Which amp models you choose, and how you play your guitar will affect the DT differently. In other words, if you play palm-mutes, or very staccato, or in a manner that has very distinguishable start and stop times, the DT appears to look for those small dynamic gaps of near silence and/or intensity between chords/notes as a trigger mechanism. Long, fluid, non-dynamic, sustained signals, especially when using the naturally compressed higher distortion amp models, present fewer instances for the triggering mechanism, thereby requiring higher parameter settings in order for the double tracking effect to be more dramatic.

DT TIMING PARAMETER: The four images below show a close up look of how the DT's Timing Parameter affects each impulse of our incoming test signal by applying a randomly changing offset delay to either AMP1 or AMP2. There appears to be a wide window of offset delay times. And the Timing Parameter adjusts just how wide this window will be. At lower settings, the offset delays tend to be between 1ms. and 5ms. As you increase this parameter, the offset delays can vary anywhere between 1ms. up to 22ms. and more. I did not try to discover what the maximum offset delay might be, but as the owner's manual describes, it's enough to make you think you're playing with a drunk guitarist.

DT DYNAMIC PARAMETER: The two images below show a close up look of how the DT's Dynamic Parameter affects each impulse of our incoming test signal by applying a dynamically fluctuating (semi-polarized?) volume offset to both AMP1 and AMP2. In more simple terms, it appears that as the DT applies a volume increase (or decrease) to AMP1, there is an opposing volume change occurring in AMP2. As you increase this parameter, the volume differences can become noticeably greater to the point of sounding very unbalanced and unnatural.

..

Let me show you another example. The first image below shows the left and right output signal of the Vetta with the Double Tracker turned OFF. Take notice of how the waveforms appear the same. Even though we have each amp panned hard left and right, because the output signals are virtually-identical, the sound is MONO, and appears at the very CENTER of the stereo field.

The additional image below shows how the Double Tracker's Dynamic Parameter can affect an incoming guitar signal when turned up to extreme settings. Once again, take notice of how the DT applies opposite volume adjustments to both AMP1 and AMP2. As the red lines indicate, as AMP1 is processed with a volume increase across time, AMP2 experiences a volume decease. Extreme settings produce very unnatural and unpleasant volume shifts between the two amp models.

PARAMETER SETTINGS: The parameter settings for the Double Tracker can really vary depending upon which amp(s) you decide to use. You can sonically abuse your output signal by using more extreme settings, so try very minimum settings when first tweaking the parameters. Using two different amp models will help to produce more realistic results for stereo recording. And your style of playing has a direct affect upon how the DT works.

The TIMING parameter seems to be the most sonically useful. Just one TIMING knob "click" past OFF is enough to make a noticeable difference, by subtlety widening the two amps into a stereo image. At higher settings, the timing differences increase, and produce an even more distinct stereo image. TIMING settings up to 12 o'clock work best, and you may find some use for settings above this, but set too high, and you may find too much "sloppiness" between the two amps. You may also notice that when using "identical" amp models and settings panned hard left and right, the timing differences that occur between AMP1 and AMP2 cause the signal to jump around in the stereo field. But keep in mind, this is typically not the sonic result when two different amp models are used, and it may be best to actually use two different amp models to achieve full potential.

The PITCH parameter simply adds a subtle detuning between the two amp models. Minimum settings can go unnoticed. The detuning really becomes apparent only at higher settings.

The DYNAMIC parameter is the one most difficult to adjust. Minimum settings up to 9 o'clock seem to work best and sound more natural. However, if you adjust the setting too high, you will get very unnatural increases and decreases of volume between the two amps.

FINAL CONCLUSIONS: The Double Tracker is more easy to recognize while listening to a playback recording of the Vetta then when you are in the midst of playing the guitar (with the exception of using an extension cab). When you are sitting in the "sweet" spot between studio monitors (or between the Vetta Combo and a Vetta 212S Extension Cab), the effect is clearly heard, and impressive enough to appreciate when you get the parameter settings adjusted properly. Anyone with a good ear and attention to sonic detail will definitely notice it immediately. But realistically, how many end users are sitting in front of studio monitors playing the Vetta direct (with the amps speakers turned off). Some users may actually be hearing the effects of the DT, but not recognize the effect depending upon their degree/level of experience working with stereo signals...and so many guitar amps are mono (with stereo fx), so many users may be less experienced with the stereo capabilities of having two configurable amp models.

I'll be the first to say that the Double Tracker may not be distinctly recognized if you are playing "only" through the Vetta Combo's speakers, this is simply due to the fact that the amps speakers are located so close to each other, and the listener is surrounded with sound. However, the effect becomes much more distinct when recording from the DIRECT OUTPUTS while listening to playback on studio monitors...you are able to point to an exact location in the stereo-field from where the sound of each amp originates, and hear the timing, dynamic, and pitch differences.

And the DT can sound downright awesome when using the Vetta 212S Extension Cab (or any additional extension cab) to actually create some physical space between the amp models when they are panned hard left and right. Having an additional Extension Cab is the "only" and best way to compliment the Vetta Combo so you can hear the full potential of sound that it is capable of producing. An Extension Cab provides for a very wide "stereo" separation of the amp models, because the Vetta Combo is handling the LEFT side of the stereo outputs, and the Ext.Cab is handling the RIGHT.

With the right settings, the Double Tracker can be used most effectively with both direct recording and live performance. The DT can produce some fantastic stereo results provided you have the parameters adjusted to optimize the effect for whichever amp model(s) you decide to use. Used in the home recording studio, it can really sound very similar to a double tracked guitar. Used on stage, it can add a "wide" 3-dimensional quality to the sound...especially if you are using the PING PONG DELAY which will jump between the cabs LEFT-RIGHT-LEFT-RIGHT etc. Can it accurately re-create a double tracked guitar 100%...almost. The technology is there and can only get better over time. Line 6 has created a winner effect in my book, and I look forward to the day when their digital technology reaches the point of completely fooling out sonic senses.

Source: vettaville.com / vettaville.nl

Tone Quest - Double Tracker 2... 09-01-2005

Source: vettaville.nl

Tone Quest - Equalization... 09-01-2005

EQUALIZATION

Introduction
	Have you ever found yourself listening to a car or home stereo and didn't like what you were hearing and immediately reached for the Treble (high frequency) and Bass (low frequency) knobs to adjust the sound? If so...then you just stepped into the world of equalization.
	Now it's not my intent to try and cover every aspect of equalization in this article, because there could never be enough written. I'm also not going to focus on too much technical theory, because we're guitarists right - just give us the basics. So I'm going to try and write this article to be as simple to understand as possible.
What Is Sound Anyway?
	In order to better understand the principles of equalization for tone manipulation, one needs to first understand the underlying nature of the audio spectrum. To put this all simply, sounds are made up of different frequencies. Different instruments sound the way they do because of the complex manner in which the frequencies they produce all add up. There are frequencies we can hear and those which we cannot hear. Why? Because the human ear is designed to only hear sounds that occur within a given frequency range.
	So what frequency range can the human ear hear? A good, young, healthy, and undamaged ear can hear frequencies in the range of approximately 20Hz - 20kHz. Frequencies are often expressed in Hz (hertz) or kHz (kilohertz). For those who need to understand the math, 5000Hz = 5kHz, or 1.2kHz = 1200Hz, etc.
	Take Note: Things start to get a little more interesting when we start to look at how frequencies play their part in the guitar world. It helps to know that the lowest E-string on a standard 6-string guitar is tuned to 82.41Hz. It also helps to know that guitar amp speakers are not capable of producing too many frequencies above 4kHz. Guitar speakers are designed with a limited frequency response and cannot playback a full-range of frequencies such as recording monitors or a P.A. system. I know this is over-simplifying things, but Vettanarians should initially start off concerning themselves with the frequency range between 80Hz - 4kHz (and perhaps a little further on either side of this range if needed), when equalizing patches on their Vetta.
Guitar String Frequencies
	E = 82.41 Hz A = 110.0 Hz D = 146.8 Hz G = 196.0 Hz B = 246.9 Hz E = 329.6 Hz When you consider the fundamental pitches of each guitar string, you will notice that they are towards the lower-mid end of the frequency spectrum. However, it is the actual harmonic overtone content of each string which pushes its tone spectrum into the mid to high frequency ranges.   20Hz 40Hz 80Hz 160Hz 320Hz 640Hz 1.2kHz 2.5kHz 5kHz 10kHz   20kHz Low           Mid       High
	The Ten Octaves of the Audio Spectrum The audio spectrum which in which we hear is often divided up into 10 octaves to help define the particular musical, acoustical, and psycho-acoustical qualities within. 16 kHz 20480 Hz Octave 10: Extreme highs, hiss, very little musical content here. 8 k 10240 Hz Octave 9: Highs, treble, metallic brightness, brilliance, upper musical content of guitar strings. 4 kHz 5120 Hz Octave 8: Presence, upper end of tone spectrum for many instruments, brightness.  2 kHz 2560 Hz Octave 7: Upper Mid-Range. Hardness, bite, intensity, loudness, definition. Major range of harmonic content for most instruments. 1 kHz 1280 Hz Octave 6: Mid-Range. Highest fundamental pitches reside here. Beginning of upper harmonics. Major overtones for most instruments. 500 Hz 640 Hz Octave 5: Lower Mid-Range. Body and richness of sounds. Fullness. Warmth. The primary treble of musical pitches resides here. 250 Hz 320 Hz Octave 4: The "mud" range. Thickness and muddiness, thump.  125 Hz 160 Hz Octave 3: Upper Bass. Musical foundation for many instruments. All speakers play back this octave. 62.5 Hz 80 Hz Octave 2: Lower Bass. Bottom of musical pitches, primary bass energy. Sonic foundation. Most loudspeakers can play back this octave. 31.25 Hz 40 Hz Octave 1: Very Low Bottom End. Little musical content here, too low to be played back by most loudspeakers. Non-pitched bass sounds. 20 Hz
Cutting Instead of Boosting
	One of the biggest patterns of EQ behavior that people often make is that they tend to approach it by boosting the gain of frequencies rather than lowering them. It's not that there's anything actually wrong with boosting, but you can often achieve your tonal goal by negative equalization - by rolling off or reducing an area of frequencies rather than by boosting them.
	Sometimes the tone (timbre) you are wanting to hear already exists within what you are hearing. The best analogy for this is to consider a stone sculpture. Think about how an artist can take a large piece of stone and then begin carving away the areas of the rock that are not needed to create the image they desire from within. This same approach can apply to creating guitar tones as well.
	For example, if you find yourself boosting both the highs and lows, you can sometimes achieve the same (or better) results by scooping out the mids instead. Keep in mind that any time you can reduce, rather than boost frequencies, you are automatically helping to reduce noise as a beneficial side-effect. Second, if you can achieve the same results with one band of cut than two bands of boost, you have saved yourself an extra band for later use, and you are helping the original signal to sound more clear and clean.
	Take Note:   When adjusting an EQ's gain up/down, reductions will generally be larger than boosts. Why? Because of the way in which our ears hear a change in sound. Please keep in mind that boosting a band of frequencies will tend to increase the overall signal level significantly, and if overdone, this can result in unfavorable effects upon your tone. On the opposite side of this, cutting a band of frequencies doesn't change the overall signal level much at all.
Graphic Equalizers
	Graphic equalizers are certainly a great tonal tool for shaping sound spectrum, but they are not necessarily the best tool that could be used. As you can see from the image below, the center frequencies are set: 100Hz, 200Hz, 400Hz, 800Hz, 1.6kHz, 3.2kHz, and 6.4kHz. Please realize that with a graphic equalizer, you are not boosting/cutting just these specific frequencies alone, there are other frequencies on either side of these set frequencies which will be affected as well when you move the slider. In other words, these sliders have a fixed center frequency (which you see) and a fixed bandwidth (of which you don't really know - unless the manual tells you).
	Anyone who has spent time working with an EQ stomp box has probably experienced at least one frustrating moment when you thought, I can get real close to the sound I'm looking for, but it seems the frequency I'm needing is in between two of these sliders - such as 300Hz, or 650Hz, or 2.4kHz, etc. You get the idea...
	Even the Vetta's Graphic EQ center frequencies are set:  LOW = 80Hz LOW MID = 200HZ HIGH MID = 800HZ HIGH = 3kHz
	Take Note: Graphic EQs limit your choices to specific center frequencies and pre-determined bandwidths to boost or cut. But because of the fixed nature of these guitar graphic-eq stomp boxes, they can teach you a thing or two about which frequency areas you should be paying attention to. When you look at most every eq stomp box like this, the frequencies are typically the exact same (or very close) for a specific reason. Why? Because many years of experience has taught design engineers that these are the frequency areas that are most often needing to be adjusted by guitarists.
Parametric Equalizers
	When you want to get down to some serious frequency tweaking, the Vetta's 4 Band EQ is the tone tool of choice. Why? Because in addition to the Low & High Shelving EQs, there are two bands of fully parametric EQ. A parametric EQ is very powerful because it gives you complete control over the center frequency, the width = Q, and the amount of cut/boost. With a parametric EQ there are no longer limitations. You are no longer limited to the pre-determined center frequencies used in a Graphic EQ. So you can move them up and down the frequency range to dial into the specific frequencies (or area) that you want to.
	Still confused - don't worry, parametric equalizers always seem to confuse and intimidate users, but they're really not that difficult to use or understand. Probably the most confusing aspect of a parametric EQ is the "Q" parameter, which just happens to be the real power parameter behind this type of EQ.
	What is "Q"   The "Q" parameter is what sets the width of the band of frequencies that will be boosted or reduced. In other words, it determines the amount of frequencies on either side of the center frequency. The basic idea behind the "Q" parameter is this: why boost/cut frequencies that don't need to be. By being able to control the actual width of the band being boosted or cut, you have more flexibility to shape the EQ to fit your needs.
	Understanding Q Bandwidth The "Q" parameter is what sets the actual width of the band of frequencies you can work upon, anywhere from a very broad bandwidth, to a very narrow bandwidth. You can see this for yourself from the images below.
	Take Note: The images below are simply meant to provide you with a visual representation how the Q control of a parametric EQ allows you to widen or narrow the bandwidth of frequencies you boost or cut. So don't get hung up on looking at the Q numbers and bandwidths since the Vetta only allows for Q settings from 0.1 - 2.0, because these are more than enough to alter the audio spectrum of your guitar tone.   Looking at the Vetta Owner's Manual, many of you tech-heads may have noticed what appears to be a typographical error regarding the description of how the Q parameter works in the 4 Band EQ. I've checked with Line 6 about this, and the original manual described it wrong. To set the record straight:   The implementation of Q in Vetta (and Vetta II) follows the normal definition of Q, where the higher the number, the more narrow the bandwidth, and the lower the number, the more wider the bandwidth.
Finding The Sweet/Sour Spots
	One of the biggest frustrations with applying EQ is that we don't often know where to begin. Where exactly are the sweet spots that bring out the life in the tone? Where are those annoying sour spots that make us cringe? Should I boost or should I cut? Good questions - but very difficult to answer. Learning to use EQ can be an art all unto itself, and a very subjective one at that. But there is a fundamental method to help you get into the ballpark.
	Effective EQ involves finding those areas of the tone that may need to be increased or decreased in strength. When you think a tone sounds too muddy or too bright, those are typically areas of the tone which contain more energy than surrounding areas.
	Suggested Q Setting   In general, start with a "Q" setting of 0.7 - 1.0
	The simplest method to find sweet/sour spots is to turn up the gain (about +10dB on the section of the equalizer you are using) then slowly sweep the frequency control up and down. Try not to focus on the overall sound itself, learn to listen for how the equalizer is affecting just the frequency area you are sweeping across. Remember, you are trying to dial into an area of frequencies (similar to a radio station). When you hit that area of resonance - which is that naturally occurring peak in the sound that already stands out as good or bad to your ears, this area might benefit from being boosted or cut. After you think you've got the right tonal color, then re-adjust the Q parameter to try and trim the bandwidth to just the right width that you need.
	Less is more. The minimal amount of EQ is usually best.
	For instance, if your looking for the area that sounds "boxy" the boxiness will really jump out at you when you sweep across it. Once you've found the area you are looking for, then it's just a matter of deciding how much you want to get rid of. Return the gain to zero, then start adjusting the gain up/down until things sound like you want.
	Take Note: If you use the Vetta's 4 Band EQ, you can increase your accuracy in pinpointing the area of frequencies to be controlled by adjusting the "Q" control for a very narrow bandwidth.
Vetta Cab Models As EQ
	Patch programming habits may be hard to break. Many of us are probably guilty of taking the easy (lazy) way out by just using the default amp/cab settings which come up every time, but that doesn't necessarily mean they're the best combination of amp, cab, and parameter settings. With all the tonal versatility that exists within the Vetta, experimentation will be the key to tonal success. And possibly one of the most overlooked applications of EQ are the cab models themselves.
	Experiment with different cab models before reaching for the EQ
	How many of the Vetta's cab models do we scroll right past and overlook because our narrow thinking has pre-conditioned us to NOT even consider using them? Don't think of the Vetta's cab models in terms of their speaker size and configuration, instead, think of them as 28 (different) preset EQ templates which must be tried out with whichever amp model you are working with. You may come to find out that the "magical" tone you're striving for suddenly appears from using a particular cab model you would have never considered using previously.
General Guitar Equalization Tips
	While it is impossible to give EQ settings that would be guaranteed to give you the "specific" results you might desire, I've included the following "generalized" eq tips for guitar. There are no rules when trying to get the guitar sound you want. Boosting somewhere between 75 - 90Hz can help to bring out cabinet clunk. Keep in mind that since the low E-string of the guitar is around 82Hz, it's not necessarily a good idea to boost the low frequencies below this note unless it helps you get your sound. Cutting somewhere between 100 - 250Hz can help to resolve a boomy or boxy sound. Try a "Q" setting of 1.0 - 1.4 Cutting somewhere between 160 - 320Hz can help reduce a muddy tone. Try a "Q" setting of 0.7 - 1.0 Bite can be added somewhere between 2 - 6kHz. EQ Food for Thought I've noticed that quite a few Vetta patches seem to use the exact same amp model, cab, and parameter settings for both AMP1/2. And while this obviously will sound louder and more full to most, I've often noticed it can lean towards sounding less distinct and less clear when compared to using just one amp model with it's volume cranked via the Post Compressor.    Two absolutely "identical" amps are simply adding their frequency responses together, which can can lead to an over-abundance of frequencies in more than one area...and these areas can in turn start to push the speakers enough to where the tone starts to sound worse, not better.   Therefore, it might be a good idea to experiment with subtle equalization "cuts" in some frequency areas of one amp model so that it will still sound almost the same, but that it will compliment the final sound when combined with the second amp model. Take Note: It doesn't do too much good to add any really high-end boost, because a guitar speaker cannot produce frequencies much over 4kHz. But remember, rules were made to be broken, and if it works for you, then go for it.
Frequency Areas
	The following EQ lexicon in by no means comprehensive and authoritative. Equalization can be defined with a variety of frequency vocabulary. However, many of these terms are fairly universal in the way in which they help our minds to grasp the realm of frequencies. Remember, many of these regions can overlap and often do. Boomy - low lows, typically in the region of 40-60 Hz. Telephony - a concentration of frequencies around 1.5-2.5 kHz. Sparkle - extremely high brilliance almost beyond hearing, around 15-20 kHz. Fat - region just above boomy, about 60-150 Hz. Cutting/Biting - frequencies which cut through, about 2.5-4 kHz. Brightness - can be achieved by a global shelving boost of everything above 10 kHz. Woofy - in the region of 125-250 Hz. Presence - anywhere from 3-6 kHz can make a sound more present. Darkness - the opposite of brightness, a general lack of highs at 10 kHz and above. Puffy - about 250-500 Hz. Sibilance - the "s" sounds of vocals often found at 7-10 kHz. Muddiness - excessive low end and also low mids, woofy and puffy combined. Warm - often found between 200-400 Hz. Zizz - a pleasantly biting high-end or brightness, around 10-12 kHz. Thinness - the opposite of muddiness, a general lack of lows and low mid frequencies. Boxy - usually found between 500 hz and 1 kHz. Glass - very translucent but noticable brilliance around 12-15 kHz. Openness - a quality of having sufficient highs and lows. Remember that equalization is extremely subjective and totally dependent upon each and every guitar tone being worked on. Therefore, one rarely can say boost or cut "x" frequency by "x" number of dB and have it apply to every patch in a generalized (or specific) way.
Equalization Is A Fundamental Practice That Must Be Mastered

Source: vettaville.com / vettaville.nl

Tone Quest - Fletcher Munson Theory... 09-01-2005

This page is important if you want to understand why the patches you've created sound different to you when played at loud, normal or bedroom sound level.

This all has to do with the theory Fletcher Munson has givin us. Apart from that your sound will be influenced by roomsize, playing level, reflections, materials in the room etc... However it's good to understand why there's a difference and general tips can be givin on how to adjust this.

You will see lots of references to equal loudness curves or equal loudness contours. These are based on the work of Fletcher and Munson at Bell labs in the 30s, or perhaps refinements made more recently by Robinson and Dadson. These were made by asking people to judge when pure tones of two different frequencies were the same loudness. This is a very difficult judgement to make, and the curves are the average results from many subjects, so they should be considered general indicators rather than a prescription as to what a single individual might hear.

What in the world is a Fletcher-Munson equal loudness curve, and why should I care?

Humans don't hear all frequencies of sound at the same level. That is, our ears are more sensitive to some frequencies and less sensitive to other frequencies. Additionally, the sensitivity changes with the sound pressure level (SPL). Take a look at the chart below. You'll notice it's marked horizontally with a scale denoting the frequency of sound. Vertically it's marked in SPL.

On the chart are a number of curved lines, each with a number (loudness level) marked. First, notice the lowest solid line marked with a loudness level of 10 phons. (The loudness level in phons is a subjective sensation--this is the level at which we perceive the sound to be.)

From about 500Hz to roughly 1,500Hz the line is flat on the 10dB scale. This means that for us to perceive the sound at a loudness level (LL) of 10 phons, (the overall curved line), frequencies from 500Hz to 1,500 Hz must be 10dB. Next, look further into the higher frequencies to 5,000Hz.

Notice the line dips here--this indicates that we perceive 5,000Hz to be 10 phons when the source is actually only 6dB. To perceive 10,000Hz at the same level (10 phons), it would need to be about 20dB. From this we can clearly see the ear is more sensitive in the 2,000Hz to 5,000Hz range, yet not nearly as sensitive in the 6,000Hz and up range.

Look down at the lower frequencies to 100Hz. For us to perceive 100Hz as loud as we do 1,000Hz (when the source is at 10dB), the 100Hz source must be at 30dB–that's 20dB higher than the 1,000Hz signal! Looking even farther down, a 20Hz signal must be nearly 75dB (65dB higher than the 1,000Hz signal)! We can clearly see our ears are not very sensitive to the lower frequencies, even more so at lower SPL levels.

Why is this? A simply physical explanation is that resonance in the ear and ear canal amplifies frequencies typically between 2,500Hz and 4,000Hz. Why can’t we hear every frequency at the same level? One reason could be because most intelligibility is found in the 2,000Hz to 5,000Hz range. Our ears are designed to be more sensitive here. While our ears are capable of hearing the lower frequencies, our bodies feel them more than we actually hear them.

This is the reason why many people who are nearly or completely deaf can still enjoy music--they can still feel the low frequency content in their bodies. (This assumes the level is sufficient that they can feel it. Often such people will actually sit on a speaker so they're in direct contact with it and the vibrations of the speaker are conducted right into their body.)

Notice how as the overall loudness level increases that the low frequency curved lines flatten out. This is because at higher SPL's we are more sensitive to those lower frequencies. Also notice that as the SPL increases, our sensitivity decreases to the frequencies above 6,000Hz. This explains why soft music seems to sound less rich and full than louder music--the louder the music is, the more we perceive the lower frequencies, thus it sounds more full and rich. This is why many stereo systems have a loudness switch--when you're listening to the stereo at low volumes, you activate this switch that boosts the low and some of the high frequencies of the sound.

Typically people become uncomfortable with levels above 100dB. You will notice 100dB is needed to perceive a loudness level of 100 phons at 1,000Hz--only 90dB is required to give a percieved loudness level of 100 phons at 4,000Hz. Again, about 104dB is required to produce a percieved loudness level of 100 phons at 100Hz.

Why is all of this so important?

Simply put, it helps us understand why many subwoofers are required to produce a loudness level equal to those attained at higher frequencies. It shows us how much more sensitive our ears are to the higher frequencies which can become very piercing if too loud.
Many times it helps to use an equalizer to cut some of the frequencies around 2,000Hz to 5,000Hz a little if music is being played loudly. This action keeps the sound crisp sounding, but not distorted and piercing at higher SPL levels.

A decibel meter (or SPL meter) measures the amplitude of sound. Inexpensive meters react to all frequencies equally, resulting in what's called "flat response". More expensive SPL meters allow measurements to be taken with both "C-weighting" and "A-weighting".

A-weighting is more close to resembling the frequency response of our ears (the low end of the measurement device is rolled off, downward to simulate our lesser sensitivity to the low frequencies).

C-weighting takes more of the low frequencies into account, even though our ears don't hear them at the same level.

Thus, it's best to make measurements with an A-weighting setting to know how our ears are responding to the sound. At the same time, it's interesting to flip the switch to look at the C-weighted response as well--During heavy rock music or a Fourth-of-July fireworks celebration, the difference between the A-weighted measurements and C-weighted can be 10dB or more!

Fig 2. Equal loudness contours or Fletcher-Munson curves.

The numbers on each curve identify it in terms of phons, a unit of loudness that compensates for frequency effects. To find the phon value of an intensity measurement, find the db reading and frequency on the graph, then see which curve it lands on.

The interesting aspects of these curves are that it is difficult to hear low frequency of soft sounds, and that the ear is extra sensitive between 1 and 6 kilohertz.

Phon Explained

A unit used to describe the loudness levelof a given sound or noise. The system is based on Equal Loudness Countours, where 0 phons at 1,000 Hz is set at 0 decibels, the threshold of hearing at that frequency (see graph). The hearing threshold of 0 phons then lies along the lowest equal loudness contour. If the intensity level at 1,000 Hz is raised to 20 dB, the second curve is followed.

It will be noted, therefore, that the relationship between the decibel and phon scale at 1,000 Hz is exact, but because of the way the ear discriminates against or in favour of sounds of varying frequencies, the phon curve varies considerably. For instance, a very low 30 Hz rumble at 110 decibels is perceived as being only 90 phons (see graph);

Compare: Sound level, Volume.

It is important to realize that the phon is used only to describe sounds that are equally loud. It cannot be used to measure relationships between sounds of differing loudness. For instance, 40 phons is not twice as loud as 20 phons. In fact, an increase of 10 phons is sufficient to produce the impression that a sine tone is twice as loud.

For the purpose of measuring sounds of different loudness, the Sone scale of subjective loudness was invented. One sone is arbitrarily taken to be 40 phons at any frequency, i.e. at any point along the 40 phon curve on the graph. Two sones are twice as loud, e.g. 40 + 10 phons = 50 phons. Four sones are twice as loud again, e.g. 50 + 10 phons = 60 phons. The relationship between phons and sones is shown in the chart, and is expressed by the equation:

Phon = 40 + 10 log2 (Sone)

Equal loudness contours for pure tones and normal threshold of hearing for persons aged 18-25 years, using free-field hearing (from ISO recommendation R226).

If you wanna know more on this matter read this

A combination of table and formula is given in

D.W. Robinson and R.S. Dadson,
'A re-determination of the equal-loudness relations for pure tones',
British Journal of Applied Physics, 7, 1956, 166-181

These data are generally regarded as being more accurate than those of
Fletcher and Munson. Of course both sources apply only to pure tones in
otherwise silent free-field conditions, with a frontal plane wave etc
etc.

Source: vettaville.nl

Tone Quest - Gain Staging... 09-01-2005

Gain Staging

by DAR (Forum Member)

This page gives you insight on what Gain Staging is and how to use it.

"I decided to do this in another topic to sort of separate it from the philosophical take on gain staging.
There are several definitions that we need to start with in order to truly understand the topic of gain staging and what we're trying to accomplish when we do this.

1. Signal to Noise Ratio. That's the first step. Signal to Noise Ratio is, simply, the level of the signal with respect to the level of noise. It's also expressed as S/N.

3. The ambient environment, in a quiet studio is about 40dB SPL*. Most gear, because it's analog, has an S/N of 85dB, or higher, usually in a single piece of gear. That said, you're already left w/ a 45dB S/N as a result of the environment and a piece of gear IF you mic doesn't add any noise. Better audio equipment will run something like 92-95dB of S/N giving you a bit more S/N, overall.

S/N is an AVERAGE measurement. That means we take the signal power and average it over some period of time. That "levels" the peaks and "raises" the values to some relative value that represents the changes over time.

4. The next thing to understand is the concept of headroom. The maximum gain a device can provide to you, before distortion, is what we refer to as "peak power handling" or "peak power output". It is during these periods that the actual peak power, with respect to noise, is MUCH greater than the rated S/N (a good thing).

picture below shows the relationship between dynamic range, SNR and headroom:

The difference between the signal power average (S/N) and the peak power handing is called "headroom".

The more headroom you have (i.e. the signal power average, or S/N) the lower your S/N. The less headroom you have the less headroom you have and the more you have to control your overall signal levels.

Hopefully, that explanation is slightly clearer than mud!

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So... moving on, then. The idea of gain staging is to start with your most critical component in the signal chain and get it adjusted so that you have the best tradeoff between S/N and headroom. If you're playing classical music you have to create more headroom than S/N. If you're dealing with rock, country and modern music you can lean more toward S/N than headroom. By the time a CD reaches your ears on a car stereo there's less then 5dB of dynamic range left on it... 5dB doesn't require much meter travel.

Gain staging is, simply, this: The creation of the proper balance between S/N and headroom such that all musical peaks can be handled without changing the character of the original signal while maintaining the maximum S/N allowed by the equipment.

Additionally, the settings of each parameter of each device in the signal chain is set such the the original tone is altered only by that device and not at any other point in the chain, when the device is enabled or disabled. (unless, of course, that device is designed to modify the behavior of other devices in the chain - i.e. the use of a Tube Screamer to provide extra input gain but not distortion). To explain, when you turn on your compressor there should be no apparent level change (especially if measured) it should just smooth the peaks and bring up the volume of the lows. When you add distortion (via Ratt, or such) you add distortion, not volume... when you add Delay you don't increase the overall low-end level, or high-end level beyond what the amp is currently producing, you tone those down and "balance" the level of the delay, etc.

To achieve this on your Line 6 amp (orginally posted POD XT).

1) Choose an amp/cab/mic/room sound that you're going to want to work with. Set the channel volume, output volume, gain and tone controls such that when you play the guitar the loudest you're going to play it doesn't get overly compressed or "saggy"... and when you play the softest that you'll end up playing you still have plenty of good clean tone coming out of the thing. Tweak... tweak... tweak at this point because this becomes the fundamental building block of the entire tone you're creating.

2) If you need to "smooth" your playing out then you'll add a compressor to the signal chain. This can either be pre-amp or post-amp/cabinet modeling. So, first, remember (or measure) the output level of the amp/cab/mic model. Add the compressor. With the same signal input level, then, adjust the compressor gain, threshold, attack, compression and release controls such that when the compressor is on, or off, you end up w/ the same average signal level. Once you've done this, you've successfully gain-staged the compressor into the circuit and it's addition, or subtraction, will not produce wide variations in level...

3) Need some distortion... grab your distortion pedal and plug it in. Leave the compressor off, at this point. Tweak the gain and tone knobs to get the overall tone out of the thing that you want. Remember, distortion adds a TON of high frequency harmonics so make sure that you A/B the sounds to be sure you're not adding too much high end to the basic tone (unless, again, that's what you want). Then... once you've the overall tone that you want tweak that distortion level to be about the same with the distortion on, as with it off.

If you're using this for lead, etc. then you can tweak the level to be about 3dB hotter than the original signal level IF you have the 3dB of headroom in your original amp setup. If you don't, you're going to have to revisit the original amp tone and tweak that down 3dB to get some headroom.

4) Add some chorus... Chorus, unless it's really used to effect the sound, is a "subtle" thing. So... 1) Tweak the chorus to get the overall "tone" you're looking for. Then tweak the "level" on the chorus so that adding it to the effects loop doesn't noticably change your levels.

At this point, double check the entire chain. Start out with the amp/cab/mic model. Then add the effects one at a time. You should notice that as you add the effects the output level should not be changing NOR should your basic tone (unless, of course, the basic tone changes by design with the addition of pedals). If things are changing dramatically go back and check each item individually till you find the one that's wreaking the sonic havoc. Once tweaked check them all, again.

5) Add some delay. Delay does two things when added, if you're not careful. It, of course, repeats a bunch of stuff. So, all those low-end noises your guitar makes when you palm-mute... well, the delay loves to repeat them over and over and over again... making your tone sound like mud. Thus, use the delay levels judiciously. It's VERY easy to run out of headroom, here, with this very useful and painful effect. You may spend a good amount of time, right here, just trying to find that right balance. You can do it! - Happy Gilmour.

6) Add reverb... ahhhhh... the last thing in the signal chain. Lush, deep, loud reverb sounds so awesome in the headphones... even with a nice wide stereo field and some good speakers, while you're playing alone. Put that in a mix and instant "washout"! So... tweak up the reverb just a hint, just so you can start to hear it without having it muck with the overall tonality... reverbs have a nice way of accentuating high-frequencies... so tone those down if you need to turn it up, a bit (remember - you can't get more level out of the thing when you turn this on)...

There you have it... gain staging in a nutshell. It's not too terribly hard.

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Couple of things... in the real, rather than the fairy tale world of the Line 6 Amp (original text POD XT), effects actually create a great deal of noise. Here's a quick rule of thumb.

The "louder" the input signal to a device the better the S/N will be. Conversely, the less headroom you're going to have. In noisy environments (such as the one created by vintage gear - hums, hisses, buzzes, crackles, pops and various other treats) it's best to try to crank out the most volume you can w/o changing your basic tone and compromising the basic dynamic range of the music you're trying to perform. (remember to always wear your ear plugs when doing this).

The louder things get, the more you have to drop back on the extreme settings of the tone controls, especially the bass control. Everyone loves that knob at 10... however, remember, in the XT as in the real-world, the louder things get the louder, and more pronounced, all the bad things get. Just because you have headphones, or studio monitors, available and you're listending at a cool 85dB and protecting your hearing doesn't mean that your virtual Fender Twin isn't running at 125dB and suffering all of the evils of that intense volume level.

It still works the same in the XT as it does in the real world. Thus you really have to pay attention. The louder the volume gets, the more you have to back off on the tone controls, etc. to get the amp back into a resopnsive mode where it does what it's supposed to. There are some other posts where folks describe similar issues w/ real life amps, including Bogner, Fender and others. I, also, have experienced this first hand.

Have fun. Hope that helps...

Used with permission: Original text by Dar

Source: DAR / vettaville.nl

Tone Quest - How to get Loud... 09-01-2005

Source: Line 6 / vettaville.nl

Tone Quest - Post Compression... 09-01-2005

Post Compression

Post Compression by Jaded Faith (ION Forum Member)

This thread is for all of you who are struggling with the strange and mysterious world of post compression. I have a bit of knowledge in the compression department. If this is redundant for anyone, skip this post. For some, this might be a huge help.

As far as gain and saturation are concerned, let's leave these terms to explain processes in the amp models stage, not in the post effects for arguements sake.

So let's say you dial in "tone x". It is a high gain patch. Now you want to play with the post compression.

First is our threshold. This is the point where the compression kicks in. The bigger the negative number (example: -15 is a bigger negative than -10), the smaller the dynamic range between the softest and loudest notes you can play. To hear an extreme example of a small dynamic range, check out the "Trademark Clean" factory preset. Crank your Vetta. See just how little difference picking you ass off does to make the output louder? That is a high threshold combined with a very squashed ratio.

We will next look at the ratio. This is the amount of volume going into the compressor it takes to get a given output if the compressor was not on. This works hand in hand with the treshold setting, because the input must pass the threshold point before the signal is squashed by the ratio amount. For our purposes, think of 1:1 as no compression, 20:1 as super-squashed. Using my last example, the "Trademark Clean" patch has a 20:1 ratio, if I am not mistaken.

Last is our gain setting. With all this level reduction, you lose some volume. This setting is how you gain that back. It is also a common method by many of us to balance out one patch relative to another or just to make a patch louder. This is a great way to do this too, as you spend all that time setting the tone controls on the two amps you are using. This is a great tool for blending them.

You will not get the same type of volume as the uncompressed signal would. You have this squashed, focused signal and simply boost it. This is why you hear compressors are used by guitarists to increase sustain. Rather than having this screeching feedback (think a pair of Recto's, full volume and gain, no noise gate and you are standing 2 feet in front of your Vetta style screeching feedback), you can obtain a smooth, controlled and focused feedback/sustain we guitarists are quite fond of. See my "Paint it Black Solo" patch on here to hear a smoothed out lead patch with a 6:1 ratio to control notes on the high E and B strings. I do not really use alot of threshold, as I like to leave my dynamic range as open as possible so I can still get dynamics with my pick attack. You can also see how this process can help to tighten up a high gain rhythm patch.

All of these principles apply to clean tones too. A squashed clean tone is great for that funky Strat (Chilli Peppers?) tone.

All said, I hope this helped someone make heads and tales of this valuable and often misunderstood part of our toybox.

"It's about thinking completely outside the box all the time."

---------------------------------------------------------------------------------------------------------No pride

One question: the gain level on the post comp shouldn't have any influence on how the ratio and threshold interact with each other, correct? The only reason a gain level was added to the post comp was to compensate the volume reduction that compressing a signal causes, right? I just want to be sure that I haven't been leading Richedie on some wild goose chase!

----------------------------------------------------------------------------------
jtroska

The "gain" (aka "makeup gain") doesn't change the way the other controls work. It simply boosts the output to makeup for volume that was lost while being compressed.

If you had a ratio of 2:1, every 2db you sent to the compressor would come out as 1db. So hitting the compressor with 6db would output 3db. Now you could use a makeup gain of 3db to bring your max volume back up.

----------------------------

richedie

One thing I have noticed is that on some of my patches I'll have the rhythm patch at a gain boost of 2 with the compression at 1:3? Then my solo patch of the same settings will now have compression of 2:1, but the gain I will keep the same and the volume seems fine between the two. Wouldn't you think I'd need to raise the post comp gain?

----------------------

tj4kbms

Think of volume as a diagonal line of 45 degrees with a point that it flatens out at the top like this.
***___
**/
*/
/
The threshold is the point (or volume) where the angle would begin to flaten out. Setting the threshold less negative brings the point up on the diagonal line (or at a louder volume).

The ratio controls the angle of the upper section. A higher ratio would be more flat (not getting louder). at 1:1 the line would just continue up at the 45 degree angle(no change in volume) And at 20:1 the line would be mostly flat.

The gain control is just for makeing up the volume lost by compressing.

An example:

with the threshold set at -10db and the ratio at 3:1 what happens is this. when you play softer, under -10db the compressor is not doing anything, but make up gain. when the volume coming in the compressor exceeds -10db, it will only allow the volume coming out to raise by 1db for every 3db that comes in(hence 3:1). The make up gain doesnt do anything but make all of the output louder.

So if you play a chord that is -4db, with the above setting, out of the compressor it will be -8bd. this is because you played 6db above the threshold and the compressor compressed it so it only raised 2db. 6db - 2db is a 3:1 ratio. So you see you lost 4db thats why you need the make up gain.

Why would you do all this? mainly because it make your softer sound eaiser to hear. Like for funk music when you want the wah down, to be as loud as the wah up. or string scratch to be as loud as full chords. Or mabey you want more sustain on your feedback as was suggested above.

Everyone really confused now?

----------------------------------------------------

Jaded

Glad to see this helped some guys out. I actually owned the Boss Compression pedal 2 years before I ever had a Distortion pedal! I guess all those days spent trying to figure out what the most un-obvious pedal of them all did was worth it.

Jtroska nailed it to a tee with the term "make-up gain." That is exactly how you should see it. Almost imagine it as a Master volume before the master volume.

About your lead patch Rich. You would not have to necessarily use a higher gain on the post compressor just because you have a 2:1 ratio versus a 1:3 setting. The reasons behind this are many and include other settings, effects and whatever else could be different between the two patches. Take it from Yoda, use your ears as the force on that call!

You might benefit from printing this out and sitting right in front of your amp walking through things as you read too. I kinda dig the visual approach to learning myself and sometimes the hands on thing is the best way to go.

--------------------------

John Doe

The Pod XT uses an emulation of an LA2A compressor in the post comp way. The original LA2A has a fixed attack release and ratio setting, so all you can adjust is threshold and make-up gain, which is perfectly fine for most tasks.
Question is, is Vetta2's post comp an LA2A emulation as well ? If so, since it's possible to set up attack,release and ratio individually, which settings are used by the LA2A ?
If it's not the LA2A, which settings could I use to get the same sort of all round compressor without all the attack, release and ratio tweeking for ech sound ?

-------------------------------

Jaded

One of the most famous compressors of all time was the Teletronix LA2A. This was one of the very first optical compressors developed in the late 1940s. It worked by an electro-luminescent panel (originally designed to show up instruments in a dark environment) shining onto a simple cadmium sulphide opto-resistor, and relied on the odd attack and release characteristics of these components to get the sound. In its original form, all these characteristics were wildly non-linear; the compression ratio varied with programme material (and was to some extent selectable by a toggle switch), the release characteristic was a strange curve, with a fast initial release, then slowing in speed as the gain returned to normal. The attack was very slow by modern standards, and tended to increase in speed as the equipment warmed up! Nevertheless, in the hands of innovative record producers, the compressor sounded wonderful, enhanced by the original tube amplifiers.

I did a little more leg work for you John Doe and this should be what you need:

ATTACK TIME: @ 10 microseconds

RELEASE TIME: 0.06 seconds for 50% release; 0.5 to 5 seconds for complete release

RATIO: The major difference with an LA2A is that the compression ratio is gradually reduced at a distance above threshold, slowly allowing the level to go back to a ratio of 1:1. This allows the loudest parts of the signal, such as drum beats and other peaks, to pass without being compressed as much as the rest of the signal. This is why an LA2A can greatly enhance warmth and 'punch'. I would suggest a ratio above 1:1 but probably no more than 2:1. Perhaps around 1.6:1? Try settings in this range and see what gets you closest.

---------------------------------------------------

John Doe

About the ratio though, I think I've read before (in the BassPod manual and some other places) that it's fixed at 3.3:1 and if you use the Limiter switch on the original then it becomes 10:1.

I'll just try what sounds good !

One other stupid question, how long is 0.06 and 0.5 in milliseconds ?
 

-----------------------------

Jaded

If the POD manual states those numbers for the ratio I would go by that. Trying to research the higher end of the ratio was like looking for buried treasure! 10:1 is way out there and definitely close to limiting versus compressing. If they state 3.3:1, shoot for somewhere around 2:1 in the middle. It won't do the cool trick I explained earlier (returning to 1:1) as well, but will be close. I actually use around that ratio for a very liquid solo tone, so I know it's good.

There are 1000 milliseconds in a second, so 0.5 seconds is 500 milliseconds.

Let me know if you need any more help.

-----------------------------

RichZ

What is the purpose behind the attack and release? I know what they do, but why would I want to set them anywhere but attack = 1ms and release = what ever the minimum is. I've played with some of the 2.0 patches here and found that sometimes the attack is so slow, when I hit a loud chord, you get this huge volume burst, then the compressor clamps down and it becomes unnaturally quiet. Very disturbing. It seems that the compressor should reduce the volume the instant it crosses the threshold and release it the instant it goes back.

-----------------------------

Scarr

You have to set attack and release based on the input material. If you have things setup incorrectly, you can get "pumping" and "breathing" where you're basically hearing the compressor raising/lowering the noise floor in a very artificial manner.

Depending on what you're going for, you might want to let the attack through unaffected and just compress the tail or vice-versa. A lot of acoustic sounds you hear layered into mixes are reduced almost entirely to the sound of the pick strumming against the strings.

If you're going for a nice sustaining sound, you probably don't want a short release time, or it'll chop off the tails of your notes. If your attack is too short, you might unnaturally cut off the attack of the notes and your sound will lose punch and definition.

Jaded did an excellent job explaining how the compression works, but the theory behind applying it is something that can (and does) fill books.

----------------------------------------

Jaded

I agree with Scarr on that. It is 15 years of playing with compression that made me understand it. I actually had the blue Boss compressor pedal a full year before I ever owned a distortion stomp. So I had plenty of hands on with it and could probably write you a novel about it's uses and aplications, but that has been done before. Even then, it's application is a personal preference and can be different for all.

----------------------

If you wanna read the complete thread here's the link to it

Jaded's little Post Compression Thesis

Used with permission: original text by Jaded Faith.

Source: Jaded Faith / vettaville.nl

Tone Quest - Tone Zone Articles... 09-01-2004

All articles are found in the Tone Zone section on the Institute of Noise site and links on Vettaville.nl under the Tone Quest menu above.

Andy Z is a experienced and excellent guitarplayer in both gigging and recording situations. Andy has much knowledge on how to build, create and record guitaramp tones and FX. He shares his knowledge with us through these monthly Tone Zone articles. He also is the webmaster of Institute of Noise.

Each month a new article worth reading

Source: Andy Z / vettaville.nl

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