The AM Forum

Hi all,

By profession I am a recording/live sound engineer. One of the things we do in the studio is record a guitar directly into a console and then use a technique called "re-amping". While it is normal to record an electric guitar via a microphone on a guitar amplifier to get the overdrive and "crunch", we also record separately the direct signal of the guitar on another track to re-amp later. This mean we take the previously recorded "clean" signal and on playback we can then send that signal from the mixing console to other guitar amplifiers during mixdown and pick what we think works best for the given song. Guitar amplifiers have very high impedance inputs. This method allows us to record the same musician performance thru different guitar amplifiers without the need for the guitar player to be at the studio. We can get different tone, different mic placements, different overdrives from the same performance.

To do this we use devices called simply "re-amp boxes". These devices covert a standard balanced line level signal to the unbalanced high-impedance output required by a guitar amplifier. As with any technology, we used to simply use impedance converting transformers to accomplish this, (homebrew devices in most cases) but most of these homebrew devices were less than desirable from a performance point of view.

While it may be difficult for some of you Elmers to understand, in rock music with over-driven guitars musicians invest a great deal of time and money developing the tonal characteristics and timbre of their distortions. Investing ten of thousands of dollars on guitars and amplifiers just to achieve minutia levels of timbre character. So preserving the clean signal of a $30,000 57' Gibson Les Paul guitar is mission critical for us engineers. These reamping devices therefore must have perfect fidelity, and preservation of what goes in and when it  comes out. (Incidentially, it was Les Paul who first used this technique in recording).

I was reading the post about the restoration of a Globe King 500 and read how the gentleman was using a Behringer mixer and a Marshall MXL condenser microphone. I suspect he is using an inexpensive impedance device on the output of the mixer. All this great audio gear and the last stage is going through a poor quality device. I'm guessing but I'll bet that's the case.

A company called Radial Engineering makes impedance matching devices and various interfaces for pro-audio applications. Its product line quality is outstanding, and since RFI/EMI interference issues are of the utmost concern in recording applications, they are designed to reduce these issues even in a high RF environment. They make a device simply called X-AMP. It is a device used for this re-amping technique I described above. The device will take a balanced 600ohm line level signal and covert it into two hi-impedance outputs, with the outputs having level, phase reversal. The two outputs are isolated from each other via a high quality Jensen 1:1 xfrmr to eliminate ground loop issues between the two devices being driven. The input also has a ground lift switch. The audio circuitry is Class-A. Here's a link for technical specifications and pictures:

http://www.radialeng.com/di-xamp-features.htm

This device would allow a Ham to use any commercially available pro-audio mic, mic pre-amp or mixer, and drive the audio input of two vintage BA transmitters directly and properly, with the highest fidelity possible. I plan to use some of my very high-end audio devices on my station. I have a Telefunken U-47, and a Neumann U-87 condenser microphones, and Neumann BCM-705 BC dynamic microphone. I also have a Manley Vox-Box, and a Avalon tube mic pre-amplifiers. Both preamps will be connected to my transmitters via a Radial X-Amp.

They are available from most major musical retailers like Sam Ash or Guitar Center or online from music retailers like Sweetwater. Do a Google search or follow this link to Sweetwater.

http://www.sweetwater.com/store/search.php?s=radial+x-amp&go=Go!

Now I know alot of us Amateurs are always looking for the cheap way out. (This device costs about $200 street price new). But sometimes it also better to simply let the experts do the designing on things other than RF circuits and just pay the money. (I don't see many of us building microphone elements from scratch, for instance). This is a simple off the shelf solution to what is always the weakest link in most of our AM stations. If you are one of us who is looking for the best sound possible, this is the best solution IMO to one of those nagging little "weakest link" issues. As we always say in the audio world, "Garbage in, garbage out".

See ya,

John
W2WDX

Us cheap asses in the ham world would just take a triode or FET and build a phase splitter.

That's a rather patronizing thing to say to a group of people you don't know, isn't it? Talking down a group of people you've only just met (and whose work experience you don't know beyond making assumptions) is not a good way to make a first impression.


I've been a guitar builder and player longer than I've been a ham. Anyone who spends 30 grand on a Les Paul (or any guitar) is a moron.


Les Paul also preferred low-Z balanced pickups and output circuits to hi-Z unbalanced, to get around re-amping. Nobody listened to what Les has been trying to tell us for decades, only to edited snippets.

These re-amping devices are only as good as the audio output transformer being used. By their very nature, transformers are restrictive in both amplitude and bandwidth, and the ones I see being used are both. They seriously lack in core material, and the low end suffers as a result. The only thing the designers ever seem to care about is getting the impedance ratio correct, then they just hope for the best.

These re-amping devices don't have "perfect fidelity", just good enough to pump guitar audio through.


I have generally avoided gear made for musicians in a radio transmitting environment, both amateur and broadcast. The design criteria are different. You are talking about a device made to re-amp a guitar, which has a very symmetrical waveform. The human voice does not, and those devices tend to fall short in those applications. There are much better solutions that can easily be home-brewed. A cathode-follower comes to mind, no need for a transformer there.

Notice I didn't say anything about cost? High cost does not equal high quality.


You should have read more before posting, then. There are quite a few guys on here homebrewing microphone elements from scratch. Gotta be careful with those assumptions.

If I can build something better than the so-called "experts", then I will. I'm none too impressed with much of what the "experts" have to offer these days.


The quest for "the best sound possible" leads to audiophoolery. That's when people start spending 30 grand on a Les Paul because they think it will give them talent or make them sound good. It never does. That's when people start buying oxygen-free power cables because they think it will give their amp more current to work with, as if electric power just conceives itself at the wall socket. That never works, either.

As a musician, ham, and engineer; I find most stuff being made for the music industry these days is horribly overpriced and drastically underperforms. You can't buy a lousy 12AX7 or a 6L6 for anything less than a king's ransom these days because that's what the musicians use in their amps. They aren't worth it in reality, only to people who measure audio quality by the numer of digits following the dollar sign. That sums up most of the stuff made for the music industry now.

Most AMers can (and usually do) build better equipment out of their junkboxes for far less money. Sorry, but that's a fact.

--Thom
Killer Album One Zappa's Greatest Compositions

I fully understand what u are saying BUT.......when it comes to the sound the guitar player wants, that is usually a function of his guitar AND the distortion he wants from his amp. If u go to the board "direct" as with a direct box....the distortion from his amp is gone.
Direct boxes come in basically two categories......passive and active and its up to the musician and the sound guy to decide which one is the best for the instrument or musician.
I usually run the bass thru a direct box and straight into the board because most bass players dont want any distortion. Doing that allows the board op to apply only limiting or compression to keep the bass "out in front" to maintain the drive.
Again this applies   to  rock..........not elevator music!

JMO,
Bill   

A UTC LS-10 or LS-10X 500Ω to 50K input transformer or its equivalent would do the job, if you can find one.  Audiophools have sucked up most of them.  I probably paid a buck apiece for mine at a hamfest many more years ago than I like to ponder..

Thom, I'm sorry you took offense or thought I was being patronizing. But the truth is it's counter-intuitive for some who do not like rock or not musical in general, as I'm sure many here do not or are not, to understand the issues musicians deal with to get "distortion", something most people consider "bad". As far as paying that much for a guitar, some musicians at the level I work in pay that much because they can afford it. Not because it's smart, but because it is the prevailing rate for guitars in excellent condition of that vintage. Sorry you don't understand that. To claim anybody who pays that much for a guitar is, as you say it, "a moron" is more patronizing then anything I said. In fact I find it insulting to me and my clients. It's all relative to ones economics and talent. Simply because you can't afford it or justify it? What's going on there? Many of my clients can afford it with ease, and are looking for an instrument of quality, not one simply because it costs alot. That was the point.

As far as the negative vs positive peaks on voice you are incorrect. On the output of a transmitter yes, but not in the balanced line level audio world. I spend my days looking at waveforms and I can tell you with all certainty (with the exception of bad cables in a balanced signal situation and some tube processors) the positive and negative peaks are equal.

As far as Les Paul is concerned, he preferred low-imp pick-up scheme as a consequence of reamping. Les is an old and current client of mine and we have had this discussion ad-nauseum. He made the claim about being the originator of "reamping", not me. Oh and incidentally, Les must be one of your morons, since he just bought one of the guitars he built 50 or so years ago for $36,000. Will one of your guitars ever reach that value ... hmmm?

The device in question uses Jensens, about as good as it gets in audio transformers. And the rest of the audio chain throughout is Class-A. This device is not only used for guitars.

And I'm not talking about gear for "musicians" as you say. I'm talking about stuff used by engineers, real pro-audio gear. Every radio station I see uses this kind of stuff, along with every world-class studio I've worked in around the world. You seem to make many assumptions about me here, which you are quick to point out about my comments. Which is not the case. I assume nothing. I said "most" guys don't make their own mic elements, and that is true! Most don't, a few do! I did read it! I was talking to the guys who don't.

The Radial device I suggested has a relatively flat frequency and phase response 15Hz-22k ⁺/_- 1db. Ah ... I'd say that's pretty good fidelity in my book. Incidentally, that's measured, not manufacturer published.

Audiophoolery? Is that a technical term? Well ... geez ... I had better quit my job, and go work work in a McDonalds. And toss out my two Grammy's and voting member status in AES while I'm at it. Jeepers creepers!

I'm sorry I took your reply so personally, but you're just so patronizing and made way to many assumptions about me personally and professionally when I was just trying to offer a simple suggestion that works well, regardless of your incorrect preconceptions.

So anyways ..... For the rest of you, this device would offer a good high-fidelity plug-and-play solution for those of you who don't want to build something.

John
W2WDX

Bill you're misunderstanding. The overall guitar sound is the player, guitar and amplifier as a system in total working together, of course. In recording is is common to record the guitar both with a mic on an amp (or several) on one track and the direct sound on another track. This way one can "re-amp", taking the direct guitar track during mixdown and sending it to another amp, at a later date. The direct sound is not usually used in the final mix, except in certain situations which I won't get into. Amplifiers makes a radical difference in the overall sound of a guitar. A Fender Twin will not make the same distortions as a Mesa DualRectifier, for instance. Nor will it have the same tonal characteristics. Also if the tracking on the mic'ed amp wasn't good or technically flawed in some way, or the sound of the amplifier recorded just doesn't work in the final mix, re-amping allows the engineer to re-record at a latter date via the reamping technique. All this without having to call the musician back in and also preserving the original performance (notes and styling played) which the musician might not be able to reproduce as well anyway.

In a recent session with a Metal band called God Forbid, we had one direct track, one live amp track recorded during the session, and eight reamp tracks with four different guitar amplifiers and two different microphone schemes. The reamp tracks were recorded on a different date without the artist present (they knew about this of course). Some of these reamp track weren't used and others were mixed together for the final mix. And this was for each guitarist (there are two). The direct guitar track was not used in the final mix, only for reamping. This technique allows the producer and engineer to have many options at mixdown time.

The line level signal out of a DAW interface or mixing console is not handled correctly by the guitar amp and the signal must be converted back into something that resembles a high impedance guitar pick-up signal. Think of it as an active anti-DI. It does the opposite of a regular DI. It takes a line level signal and turns into a "guitar signal". The trick here is using a DI for tracking and a reamp box that does not change the non amplified sound of the guitar. That is why high quality DI's and conversely reamp devices are mission critical for engineers. You want that guitar sound to go through the process as unchanged as practically possible.

Not to sound condecending, or anything, but this technique is used very often and I'm surprised you haven't heard of it. Look it up online. It's really a cool thing to do. Having the direct guitar as a separate track also allows the use of DSP based plug-in things like Guitar Rig in DAW's like ProTools.

Sorry about the off topic conversation guys.....

John

Hi John

I would like to personally welcome you to the site. I see Thom has welcomed you in his standard fashion. I found it interesting that he was on you not knowing who you are either. God forbid you have an opinion John!!

You will find you have company here. Many of our members are in the broadcast, recording,  sound reinforcement, and engineering industries. And you will learn we have many experts in here that talk the talk but don't walk the walk.

As you can appreciate, if you get 10 people in a room and ask each person how they would do something all ten have the perfect way.

From my years of experience  and my familiarity with consoles, processing and sound in general I found you information to be factual.

Anyway I am interested in converting the INR studios from a radio station into a home recording studio and would like to be able to lean on you for some input from time to time.

Welcome aboard and please feel free to educate the pro's in here John.

Gary/W2INR
Administrator and Chief Bottle Washer - AMfone

Thank you Gary. I was here many years ago when it was a separate BBS. I'm the guy who drew up the K1JJ 813 schematic. So I know most of you by rep and past conversations.

The code thing and my very busy work schedule has kept me away from Amateur pursuits for some time, so I may be unfamiliar to most of you. I tend to always come out of the gate a little strong with people that don't know me. It's a character flaw and I apologize. It always get me into trouble. Ha!

As far a Thom is concerned ... ah ... no .... I'm not gonna go there. I said all I want to say on that.

Gary ... any opinions you want or need on your audio setup, just get in touch and I would be happy to give any assistance I can and you may need.

John

Makes perfect sense.  If you want to try different "distortions" on a signal you need to have a copy of the undistorted signal.  Converting from one distorted master to a different one, I suspect, would be very difficult if not impossible.

Just an idiot's observation here but the way I see it is if you don't reproduce it at the transmitter OR it doesn't get reproduced at the receiver, (which is usually the case), then it is all for naught. 10~10Khz reproduction is awesome but making a transmitter to do that is pretty gut-wrenching. Finding a receiver to reproduce it is like finding hens teeth.

Good point Mike,

How many true good fidelity receivers are out there? I would bet less than the good fidelity transmitters.

G

John ,

Glad you made it back here. We are a diverse group and honestly everyone here has something to offer. 

I will pick you brain on the studio as the project continues.

Now back to our regularly scheduled thread.

G

Yup ... that's the problem. It is impossible. You'd be distorting an already distorted signal.

The reason I suggested this device is because it was design to be as neutral as possible. Before measuriing its characteristics I used a simple test. I ran a direct recorded guitar track and ran it into this device. Then I patched into a channel of an SSL console and patched the same direct track to another channel and reversed the phase 180 degrees. When the two channels were brought up to the same level you heard nothing, total cancellation. This implies the unit has very neutral frequency and phase response.

In this application, you will know you will get what you put through it and not some "color", any slight mismatch between the X-Amp and your transmitter notwithstanding. I have tried this (off-air) with several of my transmitters and it works quite well. It also seems immune to RFI/EMI and even caught some RF which was flowing on my audio cables and it didn't pass it or rectify it.

Also, in reply to W3SLK, the idea is not passing that bandwidth but having it capable and stable. There are consequences in frequency and phase response from having bandwidth limited circuitry. It's one of the reasons many audio amplifiers can produce very wide frequency response way beyond the range of hearing. Frequencies not handled well below 20Hz can have an impact on the working frequencies involved. If you don't follow, think of it like spurious emissions. If a circuit is unstable at certain frequencies it may, by things like IMD, effect other frequencies. So being stable at 20k means the harmonics below that, like 5k are not effected by these artifacts from instability at higher frequencies. Conversely, the same is true at the lowest frequencies. There is a term for this design criteria which eludes me right now.

Also neutrality, lack of color, is a factor here. And you would most likely be performing some equalization with some sharp high and low shelving filters in the audio chain before the X-Amp to limit audio bandwidth before feeding the transmitter. Fidelity isn't only in the overall bandwidth, but the quality and linearity of the passband that is used. Again, get the audio as good as it can be, the transmitter is gonna do what it does anyway. But remember garbage in the chain is additive in nature, so why not get the audio as best as it can be beforehand?

John  

   

I agree.... Most Ham rx's don't have the ability to reproduce HI-FI....That being said, I still like playing with any number of broadcast, musican, and homebrew audio boxes on my Transmitters...Thanks for your input John....Steve

Well I agree ... most receivers are "fidelity challenged". Ok ... but what about spurious emissions caused by excessive input bandwidth to a transmitter in the audio chain. What about the additive effects of distortions in the audio chain wasting power for instance. I don't know ... I'm just trying to maximize performance applying stuff I know well.  :'(

Come on guys, we like AM cause it sounds good. Why not make sure what you put into the transmitter is great. This way if you do have an issue with your received audio, you know it's not your audio chain. I thought this was all about the audio? Hmm ... maybe I'll just go and suggest this to the slopbucket crew, they "sound" like they could use the help. Whoa ... hahaha!!!!

John

Yup, the audio applied to the final plate circuit of a plate modulated transmitter (or audio input to a class E rig) must be as squeaky clean as possible, so as to maximize the on-air quality. This is, of course, pretty obvious.

Bob Orban, an audio engineer of the highest pedigree in my opinion, states the same thing. He makes the keen observation that some of the best sounding AM broadcast stations he has heard employ minimal amounts of equipment between the station console and the transmitter; i.e., the less the equipment within the signal path, the better. And these stations purchase the best possible equipment for the job. He also makes the observation that many of the larger-market stations, with bigger engineering budgets, have some of the worst sounding audio, as they have the $$$ to spend on too much processing equipment within the signal path, with the attendent result of excessive distortion, etc. This statement does not include the results of good quality equipment that is misused or improper for the application, or poorly maintained, etc.

I come from the audio school of thought that states, generally speaking, less is more. No more or no less to properly do the job at hand, and by virtue of this, minimal introduction of any undesirable sonic artifacts.

I agree with the comments posted here about the generally poor audio quality of the receivers most hams use for AM reception. Most of this distortion takes place within the IF, detector, and audio stages. It is actually fairly difficult to design a receiver to a competitive cost point so that it provides high-fidelity reproduction of an AM signal. This is another discussion for another time.

John/WDX: I am in Commack, just north of you. If you need any help with that homebrew 813s by 813s, let me know. I run a homebrew pair of 4-400As plate modulated by a pair of 833As, with all broadcast iron in the audio signal path.

Interesting thread so far.

73,

Bruce

Thanks Bruce,

I have yet to get my version of Tom's transmitter on the air, as well as others I have. My version has a little bit more beefy PSU with more current capability and capacity. On a load it sound good but until I can say AG after my callsign it's just a room heater at this point. I built it about five years ago, and I hope I will not be hearing any cracking when I put it on a real world slightly reactive load (antenna). I'm waiting on my VE exam for the General, which I should pass easily next month at the GSBARC. I had a long term problem managing the code, which I will not get into. Its not an issue anymore and I will be off the repeaters and 50 and up world soon.

Just to beat a dead horse on the positive/negative peak asymmetry thing, I think some of you who think this myth is true should look at this page.

http://amfone.net/ECSound/oscillog.htm

I don't know but this guys voice seems pretty symmetrical to me. Are you guys confusing frequency response with amplitude symmetry? A male voice does peak higher at lower frequencies which this scope reading shows.

Anyway thanks Bruce

John

My voice is asymmetrical. See the scope photo below. This is also covered in older literature, so it's not a myth.

Thanks for looking out for us. What would we do without you?  ::)

As nearly all of us on this board will attest, Steve is correct in his observation, as are other radio and broadcast engineers associated with this art for the last 75+ years.

The energy contained in most male voices is asymetrical. Female voices are generally more symetrical in nature.

I can see this everytime I fire up my rig. When I modulate the transmitter, I can see the positive-going peaks on the modulation monitor going up to and occasionally beyond 125%, whereas the negative-going peaks will occasionally hit 95%. The idea here, of course, is that you do not want this to be the other way around! That is the concept of phasing the audio correctly to an AM transmitter.

When I ask Janet to "step up to the microphone, sweetie", and watch her modulate the transmitter, the positive and negative-going peaks essentially track each other, clearly indicative of the more symetrical nature of her voice. And this is not a result of any kind of audio processing (i.e phase rotation, etc.) here at W2XR.

Perhaps John is confusing the subject at hand with something else. He is, no doubt, aware of the concept of correct phase of audio within a recording facility, etc.

With reference to audio as applied to an AM transmitter, we all know this to be the correct phase of the audio as it is applied to the modulated stage within the transmitter.

Just my observation.

73,

Bruce

Pffft! You know me better than that. I just tell it like it is. You're on your own.

Besides, from the looks of the last few posts, it looks like Deja Vu will be setting in for the rest of you pretty soon, too. Enjoy!

It really isn't all that hard to do either - but, for whatever reason, most folks choose to go the hard way on the transmitter side, and many just leave the receivers "as is" (or do scant modifications).

On the transmitter side, DC (or just about DC if you wish) to whatever you want (let's stop at 7kHz for the moment) is not overly difficult - but it is definitely different than using a piece of iron to couple audio to the transmitter.  Pulse Width Modulation has been the broadcast transmitter method of choice for about 4 decades.  I have personally used the method for over 35 years.  It is definitely less expensive, and usually easier to design and build a PWM system than to try to come up with a transformer coupled modulator of equal quality.

The receiver side is really not a big deal, but you generally _do_ have to modify (or replace) the AGC and detector, and use an external audio amplifier system.

I definitely believe in transmitting the best possible signal with the best frequency response and lowest distortion possible.  For anyone with a good receiver, the audio will sound as it should, giving some payout for all the effort expended in making the receiver sould good in the first place  8)

Regards,

Steve

I could very well be wrong Bruce. But again I'm actually referring to audio before it gets into the transmitter. Which is symmetrical with a proper microphone and in a balanced audio situation. I can't comment on things like crystal mics and their symmetry, and other such things. Now once audio gets into a transmitter this condition may change for a variety of reasons. Admittedly, this is not my field of expertise. But pro audio applications and acoustics are. And here symmetry is real, why it would change in RF circuits is something I probably shouldn't have made comment on, since I cannot speak on this with any authority. So I will stand back and step aside on that issue. So I'll just leave that out there for what it is and maybe someone can educate me. I would like to learn what would do this?

Tom ... you know me? KB2TQW is indeed my old call. You should speak plainly and say what you mean, instead of making inference. Preferably off the board. Obviously you have some old (very old) issue with me, but we probably shouldn't bore everyone else discussing it here. Personally, I have no idea what you are referring to. A past transgression of some kind on my part in the past no doubt. It was probably me shooting my mouth off at something I didn't know what I was talking about, which we all do from time to time. That's the reason for discussion boards like this. To learn through conversation, and the occasional verbal faux-pax. Be plain and say what you mean Tom, I'm a big boy. That's all I have to say.

John

I have a series of United Transformer Co. catalogues that include their highest grade transformers, going back to the mid 30's.  From the very beginning, UTC has always recommended an undistorted flat frequency response going up and down at least one octave above and below the intended frequency range of the audio the amplifier is designed to handle.  For example, if you are aiming for a transmitter with a 100-5000 Hz audio frequency range, the audio chain should be flat at least from 50 Hz to 10 kHz.

Not necessarily.  Sometimes the distortion in one stage partially cancels distortion in another stage.  If one stage in the audio chain, for example, has 0.3% THD, and a following stage has 0.25%, the combined THD of the system may end up less than exactly 0.55%, since non-linearities may be subtractive as well as additive.  This is exactly what happens in a push-pull class-A amplifier stage.  Each side inevitably has a degree of non-linearity, but the non-linearity in each side that would tend to generate even harmonic distortion should largely cancel the  corresponding non-linearity at the opposite side. That is why a push-pull amplifier stage has lower distortion than a single-ended stage using otherwise the same design parameters, despite claims made by the audiophool community that a single ended output stage is superior and the big bucks they spend on single ended output transformers.

That's right on, Don. Thank you for mentioning that. What I meant was more generally related to tonal characteristics and audio quality, and not as specific as your comment.

I'm still very interested in hearing about asymmetry in vocal signals as seen on the outputs of transmitters. A few questions. Is this related indirectly to what Don mentioned above? Would a push-pull configuration cause asymmetry in amplitude if an inbalance of some kind exists, either in the modulator or the final or both? Is it the nature of modulation transformation? Could it be something as simple as different gain within the tubes in a push-pull? What about supply voltage differences or capacity variances of such. Would a pure sine wave show as asymmetrical on a transmitter showing this asymmetry on male voices? I'm very curious about this. I am quite positive about my assertion that audio of a vocal is symmetrical prior to going through a transmitter, so I would like to know what could cause it to change. I'm not seeing this effect in my K1JJ 813 style transmitter. So I'm a little confused and even a little concerned. What am I missing?

It's getting off the original topic, but .......

Even harmonic content tends to produce asymmetrical waveform.  I can look at my own voice with an oscilloscope connected directly to a microphone, and see a certain amount of asymmetry.  I see a very similar asymmetrical waveform regardless of the microphone I use, except for a very poor quality one, like for example a WW2-era military surplus carbon mic.

Remember, capacitors and transformer cannot pass DC, so the area under the curve, which represents total energy, on the positive half of the audio cycle must equal that of the negative half unless the stages are direct coupled.  If a transformer or coupling capacitor is used, the half of the waveform with the higher amplitude ends up of shorter duration, while the lower amplitude half remains above or below the curve for a longer period of time.

Distorting the waveform, deliberately or unintentionally, so that there is a high degree of even order harmonics will allow an asymmetrical  waveform to pass through a transformer or coupling capacitor, and a good modulation transformer will pass the natural asymmetry of a voice waveform .  But if that asymmetrical wave is used to modulate a transmitter, there will still be zero carrier shift.  OTOH, with schemes like ultramodulation and negative cycle loading, rectification occurs, and the resultant DC adds to the DC from the power supply, so there is a positive carrier shift.  But the rectifier circuit must be placed between the modualtion transformer and final PA stage, or else the audio chain must be DC coupled all the way to the final to maintain the carrier shift.

If the waveform is hard clipped, extremely good low frequency response must be maintained or else direct coupling must be used, to avoid tilting the clipped waveform.

Okay, just to show what a good sport I am, I'll bite on this one.


It's seen on the inputs of transmitters, too. It's seen on all male voice audio.

Take a look at the scope shot that WB3HUZ posted. It's fairly typical of a male voice.

Look closely at the waveform. Pay particular attention to where the zero line is. Notice how every nth peak shoots way up above the zero line, while the others stay mostly on the negative side? That's the asymmetry of the male human voice.

Now plug one of your mics into a scope and speak into it. No processing, just the mike. You'll see exactly the same thing.

The fact of balanced vs. unbalanced is irrelevant. The only difference there is one conductor carrying one waveform vs. two conductors carrying one waveform in two different polarities (notice I didn't say "phases").

The male voice consists of low-frequency pulses and high-frequency semi-sines mixed together. Because those pulses are always in the same direction, you get asymmetry in the waveform. To make up for this, most of the lower-amplitude waves spend more time in the negative, so there is no DC component.

If you get your mic wired backwards, those pulses will travel negative, and the tube or transistor will get slammed shut, causing splatter. That's why we care about the phasing of our mics.

Forget about everything else you asked about. You're overthinking the issue. The waveform of the male voice is asymmetrical. Period. This has been known since humanity first developed the means to observe these things. It's not something a transmitter or processor does from bad design, it's something our voices do because they work as designed.

  And if you really look at waveforms from other sources as well you will find that most sonic waveforms are non symetrical. That is one of the main characteristics
that give an acoustic instrument or even a synthesizer for example, it's sonic signature since the assymetry effects harmonic content of the sound.
  Even something like percussion instruments have a greater positive than negative going initial waveform.
  This critical detail is usually lost due to excessive, or even just normal negative circuit feedback. This is why single ended 0 feedback amplifiers claim to sound beter even tho the overall performance specs are poor.
  But even the best low to high matching device will still be subject to the following amplifiers response characteristics that in vintage amateur transmitters usually needs serious modification to equal the input sources quality.
  Just my 2 cents saved up from 35 years of audio engineering work.

Bill,
KB3DKS in 1 Land

Ok ... more questions.


Is the asymmetry you see on your scope possibly being caused by the compensation on your probe? Don't be offended by that question ... please. I know that in the case of what I see at studios, on microphones like vintage Telefunken U-47 in good shape and new Neumanns like the U-87, the waveforms are very symmetrical in terms of amplitude. But also I have seen this type of asymmetry coming out of some tube microphone preamps, like some of those made by Manely and Avalon. Some of these preamps use a single ended designs (more common) and others use push-pull (less common), but both types show similar asymmetrical output. I'm not sure if you are familiar with modern recording systems, like digital recording on computers using software like ProTools, but you can see the waveforms as you record them, which is a great aid for engineers to see at least symmetry.

So I guess my question is ... is this asymmetry a factor of the electronics and how it works or an analog of the acoustic wavefront presented to the mic diaphram. I can't answer the latter, but the former is something I have yet to observe in practice unless there is something inconsistent in the microphones or cables.

This goes to you Tom as well. I have not observed this asymmetry in the studio on male voices, except under the circumstances I already mentioned. Now don't get me wrong, simply because I haven't seen it doesn't mean it doesn't exist. I'm open on this at this point, and after hearing more points of view. I am aware of the glottal force differences between male and female voices, and also phase rotators in broadcast applications (which I always assumed were used to reduce the peak-to-average ratio of voices by 3 or 4db). Are they actually being use to make voice waveforms more symmetrical? It's seems like something they could be doing since they are simply a chain of filters (allpass I believe) whose group delay is very non-consistant as a function of frequency. Ok ... now you guys have me thinking ... Am I wrong about this?

Now mind you, many of the vocalists I work with are very trained and experienced, so their glottal volume velocity waveform characteristics may be different due to that training. Have I tricked myself and made an incorrect assumption? I have to look into this now.

Thanks guys,

John

congrats John ...you have stirred the pot with a superior hand ...got folks talkin !! ... on microphone symetry ...satisfy yourself with a simple test ... I hooked up a Sennheiser 421 directly to oscope input ... when you talk into the mike the amplitude assymetry is immediately obvious .... may not be a condenser but is reasonable quality ... hey Thom ... this seems to have been said before ? .... my dega Vu moment ....javascript:void(0);

You are in good company John. I've heard this issue argued by several seasoned engineers over the years. In the end it gets proven on a blackboard.

John,

The phase rotator circuitry used in AM broadcast audio speech processing equipment is utilized for purposes of making the audio waveform more symetrical. The effect is significant; I can see it every time I use my Orban 9100A AM processor with my transmitter. The modulation characteristics as viewed on my modulation monitor are very much more symetrical in nature when the phase rotation is switched in. You are correct in that the circuit is really nothing more than an all-pass filter, but I'm sure it is also optimized for controlled group delay characteristics, overshoot, etc.

I suggest that you take a look at the Orban website. Bob Orban has written extensively on this very subject, and if anyone is qualified to discuss it on an engineering basis, it is he. I believe the subject at hand was covered in one of his technical white papers as posted on his site. Lots of other good technical information available there as well.

73,

Bruce

John, did you look at the scope photo I posted. It is my voice directly from a 1" diameter, high quality studio condensor mic, through a preamp into the scope. That's it. The waveform is obviously asymmetrical.




Bruce, this is nitpicky but important. The energy is usually (maybe always) NOT asymmetrical. The voltage waveform is asymmetrical but the actual energy of the two sides (positive and negative) will be equal (integrate the positive portion and the negative portion and the result will be equal).

Interesting thread indeed. We're not a group of old cranks, but look at audio and the nice sound of A.M. with practicality and simplicity. We're interested in an audio range of 50 or 60 HZ to about 8K. Nice full clean sound for A.M. The Hi-Fi bassy  sound does not propagate well in poor conditions. Too many people lsitening on a narrow banded receiver or on LSB on their SSB transceiver.
I dabble with broadcast with a radio show on a shortwave station and love the A.M. mode in Ham radio, and have often had to reverse my balanced audio going into the transmitter to get the assymetry in my voice to make pos peaks out of the transmitter.
And I have an on-going problem with my radio show. It all depends on what CD player they use or what computer is playing the MP3's. Apparently there's inconsistencies with hardwiring these devices to the audio console at the station, or something weird in my recording computer in my studio BUT I have to reverse the polarity of the audio from my mic to create the pos peaks from my voice. And their audio processing to the transmitter must not automatically shift phase. (I found that annoying, listening closely to an A.M. station playing music, I could hear the phase shifting from the processor) Music is constantly shifting in phase.
By viewing the modulation enevelope of my Ham station or the broadcast station, when they play my CD, I can see when the phasing is proper, when I open my mic.
When the phase is wrong, there is severe negative limiting and there will be reduced modulation of my voice. Ugly pos. peaks

As for that special sound from guitars, I thought there was a "dummy Load" that the musician connects his/hers amp to to get that "overloaded sound" to the audio console to preserve the sound from the speaker and cabinet. Still miking the cabinet. It seems like "Re-amping" creates 10 more steps to record a certain sound from a guitar amp. Computers, human interpretation of "is this the sound we want? or "does it sound like what comes out of the cabinet when it's cranked?" Seems like more labor.

Fred

Yes Steve, you are correct. I am speaking in strictly general terms here.

Thank you for the clarification!

73,

Bruce

That's quite a message. (Really)  Certain math relationships are quite revealing.

I see the effects of asymmetrical audio in my voice when using a clean AM transmitter that will pass below ~ 140 hz to DC faithfully. (A balanced modulator or class E rig)   A switch of the audio phase creates either a high pos peak AM modulated "shark fins" pattern (correct phase) or a more symmetrical looking waveform that hits negative 100% prematurely and has lower pos peaks.  I get some "false" but effective low end by boosting the deep lows using the EQ when in the correct phase.

Interesting thing is IF the transmitter will not pass DC-140hz cleanly, (using modulation iron for example) then the opposite phase gives higher positive peaks cuz the low frequency "shark fins" pattern become distorted and limited in positive amplitude.

I notice very little asymmetry in my voice at higher freqs above say, 800 hz. In fact it appears to be of opposite phase to my low-end voice freqs.

BTW, hello to John. Your 813 schematic has gotten lots of use - tnx again.

T

I disagree to a point. The fcc allows broadcast stations to modulate to 125% percent. Of course this is in the positive direction. A good AM compressor limiter will ignore positive going peaks untill they reach 125% BUT will limit negative peaks at the base line. There is no other way to reach 125% modulation without splatter. Im sure u realize 125% modulation in the negative direction doesnt exist...at least in normal AM modulation.

JMHO,

Bill   

John,

Better look again on a scope at the envelope of a typical voice that you record, BEFORE you run it through some processor. It is going to be asymmetrical.

I think a whole lot of us know about Jensen transformers.

You can do the same trick using high quality opamps and convert balanced to single ended. Your guitar guys like the transformer for various other reasons - not "fidelity."

I have yet to run you in QRZ.com so I don't know your last name. Or your "level" of "fame."

Many on here either know what you know and more, or know about or are in the broadcast and/or recording industry.

Fact is that you are actually talking about converting BALANCED low Z to Single Ended high Z. You don't actually "need" anything for a low Z line to drive a high Z input - you can just throw away one phase if you want. Noise and hum from a long run, Common Mode of course will not be cancelled out if you do that. But in a ham radio situation, that may or may not be an issue. And besides, no one had best be running an AM transmitter with response out to 20kHz that he USES as a matter of course.

You do know why that is?  :o

So what this means is that an old UTC A series transformer will do the job just fine. Or for that matter a whole range of other transformers that have less prestige or pedegree...

My comments come after reading only your second post... maybe all this has been covered already.

               _-_-WBear2GCR

Edit: Let's talk acoustics and transducers for a minute. Technically none are symmetrical. Not one. Nor is the waveform produced technically symmetrical, that is because air is asymmetrical even when driven "without distortion by a sine wave". Compression and expansion are not equal. We can say that it is insignificant, but it is so far virtually impossible to make anything that moves air be truly symmetrical. This includes the Neumann and Telefunkens...

Ok,

Just got back from the studio today. I did some tests, using several microphones and a nice recently calibrated Agilent scope. I also called several top engineers, both from the audio world and BCE's.

Asymmetry indeed! I went through quite a bit of ribbing today. Comments like, "Aw .. come on John, your kidding this is fundemental. YOU didn't know this?" and various similar comments from my engineer friends. My only defense is that it is not so noticeable in the waveforms in Protools, but on the scope it was prevalent. Not an excuse. It has been so many years since I bothered to use a scope in that environment.

I went on at that point to start reading the information available about why that is. It turns out, as it is obvious many of you know this already but I'll mention it anyway, it is in the nature of the mechanical folds in the larynx and how they effect airflow and subsequent wavefront formation in the mouth. Not to mention why there are differences in the larynx, especially when you compare male and females. Incidentally, there is some data out there that talks about the differences between trained and untrained voices, due to the stronger and more developed muscles in the trained voice, both in the larynx and diaphram. As it was explained to me by Alan Parsons's this afternoon, glottal volume velocity waveform characteristics are such that the acoustic wavefront produced by the stronger forces in the male voice produce stronger positive forces in the air in the acoustic wavefront leaving the mouth. These are transduced accurately by the microphone as stronger electrical positive peaks or negative peaks, depending on the phasing of the microphone. As Thom mentioned, and confirmed by my BCE buddy's, this is why phasing of microphones is an important aspect at a commercial AM radio stations, as one example.

I am humbled and yet excited about learning something new, at least for me. This is fascinating stuff. That's what great about forums like this.

I did something like this before on this board, didn't I? Thom is that what you were referring to? LOL!!!! That's funny. I'll probably do it again!!!   :-[

John  

Eureka! Give that man a medal!

Since you asked (and he did ask, folks): yes, John. A few times, actually. You rubbed quite a few of us the wrong way when you did it, too. That's okay, though... admission is the first step to recovery, as they say in those insipid twelve-step programs.

Part of it was that, but part of it was the way you blew your own horn as loudly as you could, not really knowing who your audience was. In the process, you were also showing that you actually didn't understand certain fundamentals of audio. It's impossible to do both without coming off as fraudulent.

Obviously, you didn't realize you were doing it, but that also rubbed a few of us the wrong way all those years ago.

While some don't approve of my tactics, your wheels started turning pretty quickly when you discovered I was doing it deliberately. Took about 24 hours, right? That's quicker than 10 years. We got nowhere with you being nice guys the last time around.

A word of advice: you and I are about the same age, and much younger (and far less experienced) than most on this board. Listen to them, and don't make any assumptions, pro or con, of your knowledge base vs. theirs. As we say on usenet: read, learn, evolve.

There, now that's settled. Welcome back, John, and don't be afraid to have any preconceived notions you walked in with uprooted. It's good for the soul.

Now, back to your regularly scheduled thread about audio.

--Thom
Kilowatt Amplifier One Zero Grid Current
See, guys? Now, was that so hard?

Nice results can be had by operating all the audio iron at 1/2 to 1/3 of its ratings. overbuild, and loaf along. Crank up when necessary.. I use the typical old iron for converting those levels from line to 600, etc.. no active components.

The only thing about the product I want to disagree with is the claim that the board is mil-spec. It does not look like a glass expoxy board - looks white on the end/side, like the consumer grade material, instead of the thick glass epoxy boards I usually see in the fighting stuff.

White? Wouldn't that be G-10? I thought that was compliant.

Then again, I tought they were slipping so bad as to let some of that RoHS garbage into the system.

John,

Do you wish to comment on my post, especially the part about the balanced to single ended conversion??

             _-_-WBear2GCR

Maybe. The picture is too poor. It reminds me of the whitish phenolic-like material instead.


On the question of symmetry, more exactly the symmetrical action of using a push-pull modulator to plate-modulate a class C RF stage, doesn't one modulator tube work harder that the other? Maybe it is my imagination.

and for phase rotators, nothing beats a symmetra-peak! for age and weight anyway.

Yep. Even when I still had a well-matched pair of RCA 811s in my modulator, the positive-peak tube showed a bit more color on the anode than the negative-peak tube.

The difference isn't as great with a pair in class AB_whatever compared to a pair in class B (as the vector sums are closer in AB), but it's still noticeable.

I see nothing new or extraordinary about this circuitry.

The input is balanced by a solid state idifferential amp circuit followed by a Jensen output transformer which does have good phase specs.

IMHO, if you look at the Jensen app. circuits, it appears this amp is a combination of various app. circuits.

Phil - AC0OB

Hi John!

Welcome to the board!

The Orban phase shifter is an all-pass phase rotator (using caps and an op-amp) that shifts mainly the low frequencies. So the low frequency peaks get tilted (err rotated) and end up being more symmetrical than the input (if they were assymetrical). This circuit does affect the sound a little and many people on this board end up using a mic polarity switch, which is more faithful.

You would not see the assymetry in a vocal track in Pro-Tools unless you expand the horizontal time way in to where part of a word fills the screen (so you see maybe 10 cycles of the waveform). Maybe someone singing will tend to be more symmetrical that spoken word??

As mentioned here, some modulators can be VERY linear..tube or solid state, and thus are able to impose the exact waveform on the RF carrier that is fed into it. Steve's two modulators...PDM and linear are flat within the audio range of near DC to 7kc or higher...able to pass a square wave pretty accurately. Those tube modulators with just a modulation transformer (no grid transformer) can get close to this.

As stated earlier here, for ham AM, the goal is to get the highest peaks of a voice going in the positive direction and not negative so that the carrier is not cut off by the extended peaks. Some people use analog soft limiters and/or clippers to limit the negative modulator excursions.

Please continue to chime in, ask questions and tell us of your radio exploits.

I have learned a lot from this board and enjoy reading all the opinions.

73,
Dan
W1DAN
(A broadcast engineer)

Here is a Bogen passive that's not expensive. It does not have all the high spec for frequency response, but looks OK for comunication use.

http://lib.store.yahoo.net/lib/cam-electronics/WMT1As.pdf

YUP OPCOM,
I love those WMT-1's from Bogen. My former employer always had many, many of those. Notice HAD!!!  Great z match. Made a big differernce in the TX audio in the K7DDY Class D transmitter.

Fred

As mentioned here, some modulators can be VERY linear..tube or solid state, and thus are able to impose the exact waveform on the RF carrier that is fed into it. Steve's two modulators...PDM and linear are flat within the audio range of near DC to 7kc or higher...able to pass a square wave pretty accurately. 

[/quote]

Hi Dan,

Can you clarify one thing for me with regard to your post?

Concerning the statement, "Those tube modulators with just a modulation transformer (no grid transformer) can get close to this. " When you refer to (no grid transformer, are you referring to the lack of a driver xfmr to the push-pull modulator grids, or the audio input xfmr that is generally utilized to split the phase to the audio driver?

Perhaps you are referring to the later Gates rigs with cathode follower drivers to the push-pull class B modulator grids; this circuit eliminated the need for the audio driver xfmr, but I believe still required an audio input xfmr to split the phase at the input to the driver/modulator audio deck.

How would the elimination of one or both xfmrs improve the asymetrical modulation characteristics of the xmtr? In terms of square pulse response, I can clearly see how this would improve in terms of transient responce and a reduction in overshoot and ringing, but not necessarily the ability of the rig to faithfully pass an asymetrical waveform.

Thanks & 73,

Bruce

Reposted due an error at my end in the "quote" function! Let me ask the question once again:

"As mentioned here, some modulators can be VERY linear..tube or solid state, and thus are able to impose the exact waveform on the RF carrier that is fed into it. Steve's two modulators...PDM and linear are flat within the audio range of near DC to 7kc or higher...able to pass a square wave pretty accurately. Those tube modulators with just a modulation transformer (no grid transformer) can get close to this. "
[/quote]

Hi Dan,

Can you clarify one thing for me with regard to your post?

Concerning the statement, "Those tube modulators with just a modulation transformer (no grid transformer) can get close to this. " When you refer to no grid transformer, are you referring to the lack of a driver xfmr to the push-pull modulator grids, or the audio input xfmr that is generally utilized to split the phase to the audio driver?

Perhaps you are referring to the later Gates rigs with cathode follower drivers to the push-pull class B modulator grids; this circuit eliminated the need for the audio driver xfmr, but I believe still required an audio input xfmr to split the phase at the input to the driver/modulator audio deck.

How would the elimination of one or both xfmrs improve the asymetrical modulation characteristics of the xmtr? In terms of square pulse response, I can clearly see how this would improve in terms of transient response and a reduction in overshoot and ringing, but not necessarily the ability of the rig to faithfully pass an asymetrical waveform.

Thanks & 73,

Bruce

Hi Bruce:

I was thinking in terms of phase shift over the frequency range of interest (and beyond) changing the phase of an asymetrical (or symmetrical) signal at the frequency extremes.

With more than  one transformer, it is easy to get up to 180 degrees of phase shift at the extremes of the audio passband...resulting in a conversion of normal negative feedback to positive feedback at the high and low end.

If you feed audio that has been diode clipped at the negative excursions into a modulator that has phase shift, the areas that are supposed to be flat will get tilted by the phase shift.

Normal audio's phase will also be changed at these extremes, but it is a dice toss as to whether it will be good or bad (i.e. make negative going spikes where you do not want them).

The least amount to phase shift can be had with a modulator (like a Gates BC1T) that has a cathode follower driver and a mod transformer. I'd do an active phase splitter at the modulator input as well. Reactive elements are still reactive and in my mind should be minimized.

This can be demonstrated by running signals through a Viking II or Valiant, and then changing the driver circuit to an active splitter...much wider frequency response can be had. The Modulation transformer still is the limiting item (next to the power supply).

Asymetrical modulation normally requires a modulator that can provide 125% or greater positive swing (and thus power). This obviously means a beefier power supply and modulator tubes.

In the end, the closer you can pass a square wave with enough amplitude, the more fidelity you will pass through the modulator. This is usually very hard to do with transformers.

Hope this helps...

73,
Dan
W1DAN

Hi Dan,

Many thanks for the usual detailed and thoughtful reply.

You are correct; I neglected to consider the issue of phase shift thru the magnetics within the audio path. Changing or shifting the phase relationship thru the audio path will plainly cause a shift in symetry. I can see this also when I make a significant change in the equalization of my audio; if certain frequency bands are boosted, the symetry (or asymetry) of the modulation characteristic will change.

Thanks again!

73,

Bruce

Bruce:

yes the same physics apply. I was gonna suggest your sending a square wave or voice through an EQ and look at the results when you hi pass and low pass, change frequencies, etc.

Let us know what you end up doing on your modulator!

73,
Dan
W1DAN

this conversation's really as broad as it is deep. Behringer since it is always mentioned, makes something called a Behringer US600. When properly applied to the microphone signal, it seems to fill out the speech bandwidth and increase clarity intelligibility a little, using the slightest amount of frequency shifing and mixing it with the straight signal. It seems to add more waves to the voice wafeform filling up some empty amplitude space within the half cycle. It was designed for guitars, but has some interesting effects on the voice and since it is DSP there's no having to hit the gate hard to make it work. It's sort of like having two voices at once, exactly in time, but the benefit is lost or ruined if too much shift or effect is used. Needs 10 turn pot. This thing is cheap too <$30. I just got it to play with, Not sure it would do anything good on the air.