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Author Topic: A high-fidelity interface for coverting line level audio (600ohm) to high-imp  (Read 30042 times)
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W2WDX
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« on: April 19, 2009, 02:14:58 PM »

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
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WA1GFZ
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« Reply #1 on: April 19, 2009, 03:58:57 PM »

Us cheap asses in the ham world would just take a triode or FET and build a phase splitter.
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KA1ZGC
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« Reply #2 on: April 19, 2009, 04:01:02 PM »

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.

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.

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.

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.

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).

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.

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.

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.

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).

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.

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".

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
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kc2ifr
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« Reply #3 on: April 19, 2009, 04:19:47 PM »

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   
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k4kyv
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« Reply #4 on: April 19, 2009, 04:38:43 PM »

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..
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W2WDX
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« Reply #5 on: April 19, 2009, 07:06:26 PM »

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
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W2WDX
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« Reply #6 on: April 19, 2009, 07:58:31 PM »

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!   

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
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« Reply #7 on: April 19, 2009, 08:14:01 PM »

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
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G - The INR


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« Reply #8 on: April 19, 2009, 08:23:57 PM »

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
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WB2YGF
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« Reply #9 on: April 19, 2009, 08:26:21 PM »

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.
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.
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« Reply #10 on: April 19, 2009, 08:50:19 PM »

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.
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« Reply #11 on: April 19, 2009, 09:00:47 PM »

Good point Mike,

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

G
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« Reply #12 on: April 19, 2009, 09:05:47 PM »

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
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W2WDX
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« Reply #13 on: April 19, 2009, 09:06:50 PM »

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  
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« Reply #14 on: April 19, 2009, 09:14:52 PM »

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.
   

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
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W2WDX
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« Reply #15 on: April 19, 2009, 09:45:50 PM »

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.  Cry

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
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« Reply #16 on: April 19, 2009, 10:09:37 PM »

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
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Real transmitters are homebrewed with a ratchet wrench, and you have to stand up to tune them!

Arthur C. Clarke's Third Law: "Any sufficiently advanced technology is indistinguishable from magic".
W2WDX
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« Reply #17 on: April 19, 2009, 10:43:53 PM »

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
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Steve - WB3HUZ
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« Reply #18 on: April 19, 2009, 11:05:50 PM »

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




* pos_audio.jpg (92.89 KB, 800x600 - viewed 480 times.)
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Steve - WB3HUZ
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« Reply #19 on: April 19, 2009, 11:12:32 PM »

Thanks for looking out for us. What would we do without you?  Roll Eyes
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« Reply #20 on: April 19, 2009, 11:20:14 PM »

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




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
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« Reply #21 on: April 19, 2009, 11:36:12 PM »

Thanks for looking out for us. What would we do without you?  Roll Eyes

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!
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steve_qix
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« Reply #22 on: April 19, 2009, 11:50:47 PM »

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.

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  Cool

Regards,

Steve
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« Reply #23 on: April 19, 2009, 11:57:09 PM »

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
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k4kyv
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« Reply #24 on: April 20, 2009, 12:26:39 AM »

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.
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.

Quote
But remember garbage in the chain is additive in nature, so why not get the audio as best as it can be beforehand?
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.

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Don, K4KYV                                       AMI#5
Licensed since 1959 and not happy to be back on AM...    Never got off AM in the first place.

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