Couldn't get a good sound until..

[elvis]

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.

Interesting that you keep asking about my definition of tone without providing your own. From what I have read, you seem to have the somewhat religious or philosophical idea that tone is a sort of unmeasurable spirit that gets lost is a digital representation. Well, at least until you can find a technical argument, at which point it becomes physics. Having a background in physics and audio engineering, I ask again that you cite your sources. I have had this same debate with people in the audio engineering world for three decades and it always goes the same way: "You can't measure it, but I can hear it." Get an Audio Precision measuring system. They're great. Tone and dynamics actually have accepted, useful definitions. There is no reasonable purpose for calling those definitions into question other than to distract.

I am not disputing the quality of signal processing in todays devices, such as the Fractal and Kemper devices.

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.

Thanks afu, well-written. Let's keep the focus on CONVERSION FOR REPRODUCTION, not PROCESSING, for the moment. I own tube amps just like you, and for the same reasons.

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.

Again I defer to afu. Lack of good math for emulation does not equal "digital is inherently flawed".

What I am claiming, I don't need proof. It's simple physics, and a result of the conversion itself, not the device.

The statement "I don't need proof" is incompatible with "It's simple physics."

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.

Your model of digital is the same one that has been thrown around since at least the 1980s. It has been used to teach the basic concepts of digital conversion and mathematical representation to students and the public. It is both incredibly simplistic and completely incorrect with respect to audio converters. I recommend you look at how delta-sigma converters work.

As for the steps, even for early 8-bit simple stepwise DACs, a low-pass filter gets rid of the steps very nicely. You can look at the output of an audio DAC with a scope all you like and you will not find a step. It's impossible due to the filtering. THAT is physics. Now make it 24 bits and the steps are so small that the analog noise is FAR more audible even with no filtering. Not visible on a scope by any means.

Don't take my word for it. Neil Young was one of digital's biggest detractors in the 1990s. But even his argument wasn't that digital was inherently flawed, it was that it wasn't acceptable YET. He claimed that higher sample rates and more bits per sample were needed.

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.

This argument is a bit misleading. Yes, we use certain tubes for RF, but tubes are not magical, and we run some transistors at 10's of GHz. Like transistors, different tubes have different speeds. IIRC, 12AX7 bandwidth in a circuit is a couple 10's of kHz. Your loudspeaker, by comparison, can't move at even 10's of kHz with any efficacy.

EDIT: In rereading, I think you may be referring to latency. The math doesn't happen in real-time, it takes some time, delaying the signal in time. This has nothing to do with the speed of tubes vs. transistors, it is about WHAT they are actually doing. Latency is probably the best argument regarding trouble with processors, but this has nothing to do with 1s and 0s and the physics of conversion. It is also not much of a barrier given the ridiculous speed that processors can run these days.

:?: :?: :?:

OK, so why did I go to this length? I'm actually a decent person and not one to argue needlessly. I certainly don't want to put myself on poor terms with the fine people on this forum, Vick included. I truly hope that this is all taken in the spirit of reasoned debate.

The answer is that while these arguments about "tone" may be interesting philosophically, they are harmful in a practical sense.

Given that in my experience, most people find digital processors to be "good enough", the problems that they have are almost always either pilot error or analog design flaws. I can give simple advice that helps MANY people get their amps working quite well with digital processors. If some of them never get the amp and processor to play well together, or if some find that the processor just can't do the job, fair enough. For the rest, they got what they needed.

How many people are helped by parroting "Digital just can't do audio"? None. And the more people who pass around the "digital tone suck" earworm, the more likely it will be that people simply give up when they might be able to solve their problem or gain the awesome benefits of a good processor easily enough.

I also hate the pseudo-science that comes with audio. It really is all physics and math. If you find yourself unable to support a technical argument, probably you don't understand it well enough. That audio is a crossroads for the technical AND the emotional makes for interesting philosophy, but frustrating debate.

 

[vick1000]

When I say I don't need proof, I am stating it should be obvious. Do I need to prove to you the 1+1=2? If you do indeed have a background in physics and audio engineering, which it seems everyone on the internets does in these situations, then it should be clear that once an analog signal is converted to binary, it is no longer the same signal. To anyone that has played extensively in live and studio environments, it is painfully obvious. The connection between your hands and the speaker is...severed, for lack of a better term, when there is a digital conversion in the chain. And I don't mean just with clipping, which are a seperate issue from tube dynamics and compression.

When I said "tone" is subjective, I mean that one persons idea of a good "tone" is usually completely different from someone else. While you may find the result of a digital device in your chain to be the same quality, someone else will notice something in your "tone" that they cannot live with in theirs. Your play style and hearing are most certainly different from everyone elses. So how can I describe to you my idea of "tone"? It would be like trying to describe the color blue to a blind man.

I know the definition in the audio world of the word "tone", which is a specific output of a single frequecy, such as a tone output generator, touch tone phone, etc...

Let me ask you this. If there is no difference in digital tonal conversion to all analog, why have they not been able to reproduce the clipping? The answer is simple, digital emulation can not reproduce a smooth sine wave. It may look smooth when looking at the large bandwidth scale, but zoom in and you see the steps. Thos steps are always there in ever emulated signal, even a pass through. So you take something that was once a smooth signal wave, chop it up, and put it back together in pieces with a little missing in between each piece. That's is the flaw of digital anything.

As far as latency, it's irrelevant when we are discussing a signal from point A to point B, A bieng my hands, B bieng my ears. If a latent part of the system effects the whole, it's slower than the rest of the system. The principle applies in the same way as with any processor, instructions per cyle and cycle speed combine with latency for the overall outptu capacity. It does not matter how fast a CPU is, if yu are waiting for it to perform (or it is waiting for data), the end result is the same, slower or delayed output, resulting from a bottleneck.

 

[elvis]

I apologize to barryswanson and the rest of you for a badly hijacked post.

If you must continue with my madness, go here:
http://forum.grailtone.com/viewtopic.php?f=13&t=70337

 

[barryswanson]

All good dude, it was enlightening

 

[elvis]

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.

Are you sure you weren't clipping the Nova Drive rather than the Return (I assume you mean the amp's FX return)?

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".

Since the system was louder with the Nova removed, I wonder if you were clipping the Nova front-end. It may have a limiter that clips analog so it doesn't clip the converter. I also wonder if the gain difference was from the loop headroom gain control.

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.

I don't have the schematic for your amp FX send, but this may help. A 12AX7 cathode follower is 600 Ohms output impedance. Gain will be a voltage divide between that and the load circuit. So 0.5 into 600 Ohms, almost 1 into 60k.

Unfortunately the user needs to be protected from the tube voltages (and the tubes from the user), so there is usually a resistor network between the very nice 600 Ohm follower and the send. It could be a simple series R, but is usually a voltage divide (to refer the output to GND for safety and to reduce the signal level) plus another R divide via a potentiometer to adjust send level. Or the pot is an adjustable shunt. In any case, the resistor network probably dominates the output impedance.

 

[afu]

It wasn't the Nova System. I'm using the line in and that skips the drive section. The remaining headroom was wide on the meter it uses for setting input, unity gain. The clean channel clipped the same when the Nova's gain was up or if the loop was bare and the Send was turned way up. I'm pretty sure the clipping was the amp's return stage, but the PI is possible. It just didn't sound like the output clipping or clipping I usually hear at normal loop levels. Someone else can try it and report if they want. I can't play guitar today and maybe not for several more days.

The Send pot is by itself as a voltage divider in my amp. It'll divide the 25 k pot value into 4 at the halfway mark and be a maximum of 6.25 k (two resistors of half the value in parallel). Add in the output impedance from the Send stage and it's below 7 k. That's pretty low. I'm not going to dispute your experience with loops, because it isn't necessary.

 

[elvis]

Actually 7k is quite high impedance for an audio output. Add to that the input impedance of the Nova system, which is 13k for the unbalanced line in, and you've got a significant impedance mismatch. You'll get at least 3dB signal loss, plus the filtering from 7k into whatever the input capacitance of the Nova is. I'd put an audio buffer in between. I have to buffer that connection with my G System for the same reason.

 

[jeff5]

For a little bit I thought my Dual (have the Multi Watt) seemed to have more thump and just sound better with the loop bypassed. However, as I tweaked the EQ, FX send and G Major 2 levels, I have been finding that it makes little difference whether its bypassed or not. (Using the bypass on the G Major, bypass on the recto, or disconnecting anything from the loop) I think the new Multi's have a really good FX loop.