Couldn't get a good sound until..

[barryswanson]

So I have a Multi-Watt Dual Rectifier I run in stereo with a Mark V and I was thinking of selling the recto because it just sounded dull and it has new tubes. I have the effects loop running through a g major. Anyway after reading one of Screaming Daisy's posts I switched the loop to bypass and bam. That sound I love is back! What would cause the g major/fx loop to suck so much tone? I know there is no free lunch in audio but it was a major difference. Have I set the g major up wrong? Do any of you guys run one and how have you set it up?

 

[vick1000]

Gmajor=digital. Digital by it's very nature, sucks up all tone, processes it, and spits out the interpretation of the programmers idea of tone.

 

[barryswanson]

But even in bypass mode it sounded like poo

 

[Given To Fly]

Not all "Bypass Modes" are equal. "Relay bypass" hardwires the inputs to the outputs with no digital conversion present. There are some "drawbacks" but none of them are related to tone suck. The G Major may not have this option.

 

[afu]

I think by bypass mode, he means when the loop is active, but turned off. I've noticed a difference in sound between the loop active and bypassed on my original 3 Ch Recto. Since the signal is going through two more valve stages, it's going to affect the sound.

I'm really tired and ill; I don't want to do complex math, but I'll kind of explain the Send a little. The stages input passes along all frequencies down to .7 Hz. The grid is held at 21 V below the cathode to keep things moving at a very steady rate with some amplification of all frequencies in the signal. The stage is a buffer. When the signal comes out, it hits a filter created by a capacitor and the Send pot.

At the Noon, "Normal" setting, the filter begins to cut frequencies below 289 Hz, which might be desirable, because the bit of amplification of the lows will mush them a little bit. Turning the pot to 1:00 moves the knee of the cut to about 100 Hz. Turning it to 11:00 moves it to 410 Hz. Having that filter helps clean up the mush.

The design of this loop is pretty awesome, but there's always a trade off, and, as I said, amplifying the lows will make them lose definition. It isn't just any particular frequency, because the harmonics will also be changed. Play an E5 diad and (among other things) you will hear the root, the 5th, the harmonics of the individual notes, and the harmonics of the root + 5th. So (simplified and all in Hz):

82, 123, 164, 205, 246, 328*, and 410*, 492* all sound off from those two notes. (*3rd Harmonics may not be clearly audible, but support the other notes.)

 

[screamingdaisy]

Well, since you're already running two amps... the traditional way is to go wet/dry. Run one amp wet with all the effects going through it and the other amp dry with no effects.

In the case of a Recto/Mark V pairing I'd run the effects through the Mark V as I'd want the bigger/fuller sounding amp to be dry with the effects filling up the midrange. I'm pretty sure this is how Hetfield runs his setup (Triaxis (IIC+) with effects, Triaxis (Recto) Dry and Diesel Dry).

Kim Thayil is similar with a different rig. He uses the loop on the T-Verb (midrange) and the Electra-Dyne is dry (Fullness).

Beyond that, if you want to use an effects loop or you're set up for wet/wet then you have to accept a bit of compromise. Truthfully, I think the tonal changes are a bigger deal at home than in a live situation. I've never specifically tested it, but I'd be surprised if it was all that noticeable with the drums and bass filling in everything around you.

 

[Given To Fly]

I think by bypass mode, he means when the loop is active, but turned off. I've noticed a difference in sound between the loop active and bypassed on my original 3 Ch Recto. Since the signal is going through two more valve stages, it's going to affect the sound.

I'm really tired and ill; I don't want to do complex math, but I'll kind of explain the Send a little. The stages input passes along all frequencies down to .7 Hz. The grid is held at 21 V below the cathode to keep things moving at a very steady rate with some amplification of all frequencies in the signal. The stage is a buffer. When the signal comes out, it hits a filter created by a capacitor and the Send pot.

At the Noon, "Normal" setting, the filter begins to cut frequencies below 289 Hz, which might be desirable, because the bit of amplification of the lows will mush them a little bit. Turning the pot to 1:00 moves the knee of the cut to about 100 Hz. Turning it to 11:00 moves it to 410 Hz. Having that filter helps clean up the mush.

The design of this loop is pretty awesome, but there's always a trade off, and, as I said, amplifying the lows will make them lose definition. It isn't just any particular frequency, because the harmonics will also be changed. Play an E5 diad and (among other things) you will hear the root, the 5th, the harmonics of the individual notes, and the harmonics of the root + 5th. So (simplified and all in Hz):

82, 123, 164, 205, 246, 328*, and 410*, 492* all sound off from those two notes. (*3rd Harmonics may not be clearly audible, but support the other notes.)

I have a Roadster 2x12 Combo and it shares the same tonal improvements when the Loop is Bypassed as all (or most) Rectifier models. Its a wonderful sound! But I had never heard of the FX Loop Send pot doing anything other than adjusting levels. So I turned the Send pot back and forth a bit and and there was definitely a change in Level. It wasn't until I sat down in front of the amp that I noticed there was also a change in tone. I would describe the tone as sounding bigger and drier with some extra weight across all channels. The fundamental seemed to be more present but that could have been the upper harmonics "guiding" my ear to the fundamental or it could have been the Placebo Effect. Either way, it sounded better. Where was the Send pot set? Right around 1:00! I play a 7 string guitar and I think amplifying the low end using this FX Loop trick may benefit the overall tone. Is it in the manual by chance? I will have to listen again tomorrow with fresh ears.

By the way, I also live in Tucson, AZ. Are you our resident Mesa Boogie expert? :?: This is a actually a serious question. I know other Mesa owners who are knowledgable about their amps but not to level you have demonstrated. 8)

 

[elvis]

Gmajor=digital. Digital by it's very nature, sucks up all tone, processes it, and spits out the interpretation of the programmers idea of tone.

Sorry, but this is simply not true. Most of the "tone suck" in analog/digital equipment comes from the analog circuits and the interface.

Case in point, most Mesa loop sends can't drive a low-impedance circuit well. The FX Send output impedance is usually high (5-25k) and often variable based on the send level setting.

The G Major input impedance (like the G System and other studio-based effects units) is quite low (13k) in order to keep noise low and make the most transparent buffers for the A/D converters.

These two combined create a filter and an attenuator that gives signal loss and tone change (not "suck", it's an analog filter).

The solution is to put a good-quality buffer (low noise, input impedance at least 1Meg, higher is better, output impedance 1k or less, lower is better) between FX send and the G Major input. That usually solves the interface problem.

Another common cause is simply impatience. Most people hook up the processor and then go straight to settings with 3 or more effects, each of which often has filter settings. There is usually an "All Bypassed" preset. Use that when first connecting the processor to hear if you have any tone change. Then add one effect at a time and carefully A/B bypassed with not bypassed, listening for tone change.

Good luck!

 

[afu]

I have a Roadster 2x12 Combo and it shares the same tonal improvements when the Loop is Bypassed as all (or most) Rectifier models. Its a wonderful sound! But I had never heard of the FX Loop Send pot doing anything other than adjusting levels. So I turned the Send pot back and forth a bit and and there was definitely a change in Level. It wasn't until I sat down in front of the amp that I noticed there was also a change in tone. I would describe the tone as sounding bigger and drier with some extra weight across all channels. The fundamental seemed to be more present but that could have been the upper harmonics "guiding" my ear to the fundamental or it could have been the Placebo Effect. Either way, it sounded better. Where was the Send pot set? Right around 1:00! I play a 7 string guitar and I think amplifying the low end using this FX Loop trick may benefit the overall tone. Is it in the manual by chance? I will have to listen again tomorrow with fresh ears.

By the way, I also live in Tucson, AZ. Are you our resident Mesa Boogie expert? :?: This is a actually a serious question. I know other Mesa owners who are knowledgable about their amps but not to level you have demonstrated. 8)

I don't think of myself as an expert. I tinker. I know some bits of electronics and applications to music tech, because I went to school and used to build/modify guitars, work on amps, and build effects. Now that I'm disabled, I'm trying to apply some of that knowledge to this amp, because I love it. Before I began, I read a lot of things that just couldn't be true. The analyses are more about trying to break it down so that people have an easier time, myself included. I have been a wrong about things a couple of times over the last few years, so don't just take my word.

The Send being used as a tone shaping tool isn't in my manual. It's meant for balancing the difference between the loop's on and off states. That doesn't mean it can't be exploited. Just keep in mind that "louder = better" can apply as a deceptive effect. As with anything else, time will reveal if it is preferable.

We need to jam sometime.

 

[vick1000]

Gmajor=digital. Digital by it's very nature, sucks up all tone, processes it, and spits out the interpretation of the programmers idea of tone.

Sorry, but this is simply not true. Most of the "tone suck" in analog/digital equipment comes from the analog circuits and the interface.

Case in point, most Mesa loop sends can't drive a low-impedance circuit well. The FX Send output impedance is usually high (5-25k) and often variable based on the send level setting.

The G Major input impedance (like the G System and other studio-based effects units) is quite low (13k) in order to keep noise low and make the most transparent buffers for the A/D converters.

These two combined create a filter and an attenuator that gives signal loss and tone change (not "suck", it's an analog filter).

The solution is to put a good-quality buffer (low noise, input impedance at least 1Meg, higher is better, output impedance 1k or less, lower is better) between FX send and the G Major input. That usually solves the interface problem.

Another common cause is simply impatience. Most people hook up the processor and then go straight to settings with 3 or more effects, each of which often has filter settings. There is usually an "All Bypassed" preset. Use that when first connecting the processor to hear if you have any tone change. Then add one effect at a time and carefully A/B bypassed with not bypassed, listening for tone change.

Good luck!

Yeah, cause when you convert analog to digital, the "tone" is completely unaltered. [/sarcasm]

I never said analog circuits are immune, but any processor based effects are going to be far worse.

 

[elvis]

Yeah, cause when you convert analog to digital, the "tone" is completely unaltered. [/sarcasm]

I never said analog circuits are immune, but any processor based effects are going to be far worse.

Prove it. Modern digital acquisition systems often outperform analog, so many applications have gone to direct-conversion.

Incidentally, every recording you've heard in the last 20+ years was digital.

 

[afu]

I think the impedance of the Send would be a maximum of 6.25 K + source impedance. The voltage divider ends up making the pot's maximum output impedance equal to the pot value divided by 4 (25 k /4 = 6.25 k). It "sees" two resistors in parallel. The stage should be around 600 to 700 ohms output impedance. So 7 K would be the max, probably. Going either direction from halfway will reduce the output impedance. Of the two resistors, one is growing shorter in either case.

From experience, digital technology has come far enough to sound really **** good. This isn't 1996's Digitech RP3 pedals. The D/A converters and sampling do a great job. It doesn't perform like all-analog, but I don't think that's always the goal. Especially when many people are becoming more accustomed to good digital sounds and digital can stand on its own for the things it does well.

 

[barryswanson]

A bit off topic but I just bought the Sennheiser D1 digital wireless guitar system. If you hear how good that thing sounds you might be convinced digital has come a looong way.

 

[screamingdaisy]

FWIW, once I switched from using delays that converted the entire signal to digital to a delay that had an analog dry through that mixed the digital repeats in parallel I noticed an improvement in clarity and better separation between the dry and repeats.

With an all digital signal path I always seemed to be either too wet or too dry and could never get the repeats to sit right. I'm told it's because the dry and repeats are mixed internally and you're at the mercy of both the hardware and algorhythm for how your effected signal is output. With analog dry through the processor only affects the quality of the repeat, and if the repeated signal is less than a perfect recreation it doesn't stand out as much because it's only a fraction of your total sound (unless you're the Edge).

 

[afu]

FWIW, once I switched from using delays that converted the entire signal to digital to a delay that had an analog dry through that mixed the digital repeats in parallel I noticed an improvement in clarity and better separation between the dry and repeats.

With an all digital signal path I always seemed to be either too wet or too dry and could never get the repeats to sit right. I'm told it's because the dry and repeats are mixed internally and you're at the mercy of both the hardware and algorhythm for how your effected signal is output. With analog dry through the processor only affects the quality of the repeat, and if the repeated signal is less than a perfect recreation it doesn't stand out as much because it's only a fraction of your total sound (unless you're the Edge).

That's why I'm so impressed with TC Electronic. Their time based effects from the last several years aren't completely perfect, but gorgeous, still. At high mix settings, the sounds still have slight artifacts, but they are only really perceptible at Edge-like settings. I haven't used mine alongside our drums recently, but I can't imagine it being a problem in a band.

I sometimes have trouble with the reverbs being too wet. I can tool with them to make them perfect, but if I change the volume of the amp, then I have to change the 'verb settings, too. One of my settings is -18dB wet signal mixed in at only 5% to keep it feeling right. I think "lush" is the right adjective, including the definition that means an alcoholic.

 

[elvis]

FWIW, once I switched from using delays that converted the entire signal to digital to a delay that had an analog dry through that mixed the digital repeats in parallel I noticed an improvement in clarity and better separation between the dry and repeats.

With an all digital signal path I always seemed to be either too wet or too dry and could never get the repeats to sit right. I'm told it's because the dry and repeats are mixed internally and you're at the mercy of both the hardware and algorhythm for how your effected signal is output. With analog dry through the processor only affects the quality of the repeat, and if the repeated signal is less than a perfect recreation it doesn't stand out as much because it's only a fraction of your total sound (unless you're the Edge).

I've used my G System with 7 different amps. It sounds completely different with each one. For some it's too much effect, for others, difficult to get enough. On one or two it's perfect. To me that points to the amp design, in particular the loop, and the interface between the amp and the FX. I have used buffers that help a lot.

Ultimately I think the issue is more about what happens to the signal once it hits the FX return rather than what the FX box does. If the amp is adding harmonics after the loop, that could be why you hear a tonal shift with time-effects. I know that some guys go dry all the way through the amp, put a dummy load in place of a speaker and then run THAT signal through the FX, then to a stereo power amp w/ speaker cabs.

The only reasonable test of the digital argument is to set up the FX with all effects bypassed, so that the signal runs A/D/A with no processing. If the levels are set appropriately, you shouldn't be able to tell when the FX is in the loop or not. Generally I get a very small difference that I can trace to the interaction between the FX loop send circuit and the FX unit's analog front-end, but that's about it. The rest is whether you happen to like the particular FX algorithm or not.

 

[vick1000]

Sounds to me like you use FX all the time there Elvis. See, I only use a boost and/EQ out front of my tube amps. Nothing in the loop. Now I have used all sorts of digital devices, and on there own they do a great job today. But no matter how you look at it, once the signal is converted to digital, you lost all the original "tone". See that's the thing, how one defines the term "tone". For me, it includes the dynamic feel of an all tube amp, that response from my pick to the speaker, can't be passed through an ADAC. It has nothing to do with recording, to me "tone" is not what someone else has recorded.

 

[elvis]

Sounds to me like you use FX all the time there Elvis. See, I only use a boost and/EQ out front of my tube amps. Nothing in the loop. Now I have used all sorts of digital devices, and on there own they do a great job today. But no matter how you look at it, once the signal is converted to digital, you lost all the original "tone". See that's the thing, how one defines the term "tone". For me, it includes the dynamic feel of an all tube amp, that response from my pick to the speaker, can't be passed through an ADAC. It has nothing to do with recording, to me "tone" is not what someone else has recorded.

Nope. I play with and without FX. Since I have several amps I usually just plug straight in. I attach the FX when I gig, or when I feel like it at home. So I constantly A/B with and without FX.

Again, you make an assertion that you can't pass tone (and dynamics) through ADC and DACs.

Cite your source. :?:

Interestingly, there are whole industries that are based on high-speed, high-precision measurements through ADCs. Their signals are no less complicated than what's in your guitar amp, and often way MORE complicated and with much higher dynamic range.

This all seems to be based on your experience with various processors. Not knowing what they were, I can't speak to the quality of the processors or their algorithms. But in my experience most problems with digital systems are either analog interface problems (most often impedance mismatch) or processor speed limitations (trying to do too much with too little). The rest are endemic of low-cost products: poor analog circuits, cheap converter ICs, poor-quality programming, etc.

It's one thing to say "I have had bad experiences" or "Some processors seem to struggle". But to claim that ADCs and DACs just aren't up to the task by their very nature is extraordinarily misguided. Perhaps you can explain what the technical limitations are? I defy anyone to go to Stanford University and make the claim that ADCs can't capture tone and dynamics in the range of Human hearing.

 

[vick1000]

Again, that subjective term, "tone".

What is this term to you? You can put a signal through the best audio processors in the world, but once it's converted to ones and zeros, all the original "tone" is gone. Now with some great software and conversion algos, you can get quality output from a digital device. I am not disputing the quality of signal processing in todays devices, such as the Fractal and Kemper devices. What I am claiming, I don't need proof. It's simple physics, and a result of the conversion itself, not the device.

It's very simple if you know the difference between an analog and digital signal. Even the highest bitrate digital, is still limited to a fixed number of "steps" in any variable. The output is inherently tied to those limitations, meaning you can never get the same signal out that was sent in. This is apparent on an analog scope, and easily comparable. Even that aside, the dynamics of tubes in an audio circuit, can never be replicated by digital equipment. Tubes react to signal changes much faster than even the fastest transistor, because they are not just a gate opening or closing, then bieng amplified by external devices. A tube IS the amplifier of the circuit, and changes the electron flow on a massive scale. It's very inefficient, but also very dynamic due to those high current levels.

I'll leave it at this. If digital amplification was on par with analog tube amplification, there would be no more market for all tube amps. Even the best digital guitar processors, such as the AxeFX line, and the Kemper profilers, are known to only get close to emulation of full analog tube signals. They are fabulous studio devices, making recording a breeze, allowing you to downsize a stable to practically nothing. But too many musicians know their limitations, and still demand their all tube configurations.

 

[afu]

Coincidentally, I did turn off my effects earlier with the Nova System in my modded loop. Here's what I found:

1) The Nova makeup gain can bump the signal to the point of overdriving the return stage. It was worst for the cleans. Controlling level mostly from the Send was useful for keeping the sound punchy. This won't work for a person who completely switches effects out of the signal path with the Big Foot.

2) When removing the Nova from the loop, the amp was louder. At one point, I found I was clipping the Return again; the Send was at 3:00. It's hard to judge if it was "better", but it was, maybe, a couple dBs louder.

3) Reinserting the Nova, turning off all the effects, and keeping the Send a touch above Noon sounded great. I didn't hear a big difference in response or tone from #2. I couldn't get the volume to match perfectly and that did make it hard to compare the tones. I can say it didn't sound "digital".

----
Vick, digital processing to create distortion wasn't really what we were talking about. I don't think anyone could dispute the difference between tube distortion and a digital simulator of tube distortion. Whether it can faithfully pass the signal in a loop without degradation was kind of the point, I thought.

------
Also, I made an error earlier. A cathode follower is almost always below unity gain. I believe the gain would be about .8. Cathode followers easily compress signals, according to Merlin Blencowe. After that, the pot will dump a portion of the remaining signal to ground and have a filtering effect. Half of .8 of the input to V4a is the "normal" pot setting. It is re-amped at the Return to bring it back up. The attenuation and amplification is probably the source of the difference I hear with the loop active, but switched off.

If increasing the Send is making it more lively, it could be due in part to the overdrive at the Return.