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NU9N eSSB Hi-fi Audio
Amateur Radio Extended SSB Hi-fi Mid-fi Lo-fi Audio

Transmit Audio Setup


Setting Up Your Transmitter Audio

Nady SCM-1000 Sudio Condenser Microphone

This page was designed to step you through the several different aspects of setting up your transmit audio from component ordering to individual processor settings necessary in achieving that ultimate eSSB audio that you have been looking for.

First, let me set the record straight; If you are looking for high quality and high fidelity audio for SSB, then this page was designed for you! If you are looking for something else, I suggest checking out some of the other web pages available for SSB audio that would steer you more toward an articulate upper midrange type of sound. (See the "
Related Sites" page for some of these web sites.)

AM broadcast audio has, by nature of its 9 kHz audio bandwidth, a very wide open, more pleasing sound than the typical Amateur SSB station that has most of its emphasis on the upper midrange frequencies from 800Hz ~ 3kHz. Another attribute with typical SSB audio is a severely rolled off bottom end below 300Hz and a tight roll-off above 2.7kHz. While this type of audio has been accepted for some time, and while being very punchy and penetrating regarding weak signal communications, it can be quite fatiguing to listen to after a while.

When the Amateur community embraced SSB for its power efficientcy and found that it worked good under poor signal conditions, Amateur AM, for all practical purposes, was abandoned. Fortunately, for those who love the AM mode, it's still around. However, there has been a resurgence for "Quality" modulation in the Single Sideband domain while still remaining relatively narrow as compared to its full AM modulated counterpart.

Amateur Radio SSB bandwidths can vary depending on the transceiver being used. As a bare minimum, I first of all recommend acquiring a transceiver that is capable of at least 4kHz of AF/RF bandwidth right out of the box before any EQing.

If your transceiver cannot exceed 3kHz on transmit, then forget it! You will not have the necessary bandwidth to support any appriciable fidelity!

I'm not suggesting that anything less than a 3kHz bandwidth will necessarily sound real bad. 3kHz and beyond seems to be the dividing line between sounding "Canned" vs. "Open". (Notice I did NOT say "Natural") If you have ever listened to AM broadcast, you probably noticed that the engineers have made the announcers sound very rich and powerful. I have listened carefully to female announcers and have noticed that they sound very rich and full, certainly nothing like they would sound like in person! Why? Marketing! They know what sells and how to keep their market share.

What is it that makes AM broadcast sound like it does? Let's take a look at an AM broadcast audio spectrum vs. a typical SSB signal.

AM broadcast vs. typical SSB

The broadcast AM station in white (WBBM AM in Chicago) above is producing 9.5kHz of audio via a 19kHz RF bandwidth. (9.5kHz LSB + 9.5kHz USB plus carrier) This bandwidth supports excellent vocal and fair music reproduction.

As you can see, the typical SSB signal is very narrow but unfortunately, very "Canned" sounding as well. Now take a look at another SSB amateur station who has done some work on his audio via processing.

AM broadcast vs. Mid-fi SSB vs. Typical SSB

The SSB "Mid-fi" station (Cyan) sounded excellent! Not exactly like AM broadcast, but closer than the typical SSB audio without being a full 6kHz wide. In fact, this station was about 3.5kHz wide. More importantly, the station was clean with excellent carrier suppression and extremely low I.M.D. (Inter-Modulation Distortion) products. This actually contributes to less bandwidth overall than some stations running a 2.4kHz bandwidth with poor I.M.D., sometimes making them as much as 10kHz wide!

The next graph displays an SSB Hi-fi station with a transmit passband of 6kHz. This is truly approaching AM broadcast quality sound with only half of the necessary RF bandwidth required compared to AM since an AM station would have to occupy a 12kHz RF bandwidth to produce 6kHz of audio bandwidth. For more on Extended SSB Hi-fi audio, click here to see the Extended SSB page.

AM broadcast vs. Extended SSB Hi-fi audio

So, how do we improve our audio from GRAPH 1 to GRAPH 2 or GRAPH 3... Read on!

Connecting It All Together & Interfacing the Rack to the Rig

There are several possible connection schemes when connecting your rack audio together and interfacing it to the transmitter. I will discuss the pros and cons of each below.

Audio Rack Connections:

The NU9N Rack and Transmitter The best way to connect your audio gear together is via the Balanced XLR or 1/4" TRS connectors on your gear, using premium Starquad cable. (
See below for Starquad wiring) This is not always possible however because some gear does not provide Balanced input and/or output connectors.

You could also use audio isolation transformers between each piece of audio equipment wiring them for balanced on each side. This would provide maximum ground-loop isolation preventing annoying hum and buzz while also providing great RFI immunity. However, this can be costly when using premium grade transformers like the ones made by Jensen.
See this link:

- The next best way is to use Audio Transformers that can convert from Balanced to Unbalanced or vice-versa as mentioned above, in a system that cannot inherently be completely Balanced. Once again, transformers are a great way to convert these signals as well as providing for maximum isolation between devices.

- In some systems, it may be advantageous to just use an unbalanced wiring configuration throughout the entire rack. The advantage to this, if not all the equipment has Balanced connectors, is that you don't have to worry about any Bal to Unbal (or vice-versa) conversion problems. However, a completely unbalanced rack will not have the Hum and RFI rejection advantages of a completely Balanced wiring scheme!

- As a last resort, you can just simply wire for a Pseudo- Balanced connection as shown below in figure 2. While this is not the most eloquent way to unbalance a balanced signal, and there are some technical drawbacks by doing this, it will satisfy the basic requirements for an unbalanced signal.

Audio Cables / Wiring

It is important to use good quality cabling and use the proper wiring techniques throughout your transmit and receive systems to keep hum/buzz and RFI to a minimum. If possible, you should use a twisted, 4-conductor, 95% shielded (or better) cable in a balanced configuration throughout your transmitter audio rack. This type of cable is known as "Starquad" and is available at many musical outlets for about 50 cents per foot.

To properly wire a balanced to balanced cable using 4-conductor Starquad, assuming the wire colors are White, Blue and Silver, wire the cable as follows combining the same colors on one end with the same on the opposite end. If your cable colors are different, just use the same colors on both ends of the cable as shown in the following example:

StarQuad Balanced Wiring
Starquad Wiring for Balanced XLR to XLR Connections
Pin 1 (Shield - Chassis Ground)
Pin 2 (Plus Signal) White wires combined
Pin 3 (Minus Signal) Blue wires combined

Starquad Wiring for Unbalanced XLR to XLR Connections
Pin 1 (Shield - Chassis Ground) (Output Send Side ONLY!)
Pin 2 (Plus Signal) White wires combined
Pin 3 (Minus Signal)
Output Send Side: Blue wires combined with Shield on Pin 1
Receiving Input Side: Blue Wires on Pin 3 ONLY

Starquad Wiring for Balanced 1/4" or 1/8" Connections
1/4" or 1/8" TRS BALANCED
Sleeve (Shield - Chassis Ground)
Tip (Plus Signal) White wires combined
Ring (Minus Signal) Blue wires combined

Starquad Wiring for Unbalanced 1/4" or 1/8" Connections
1/4" or 1/8" TS UNBALANCED
Sleeve (Shield - Minus Signal)
Tip (Plus Signal)
White and Blue wires all combined

Balanced vs. Unbalanced

Converting Balanced to Unbalanced or vice-versa is no light matter. Sure, it can be done with wiring configurations, but there are technical problems with this unless you plan on using several good audio transformers to make the proper conversions. This is why I recommend either a rack that is completely Balanced or completely Unbalanced and not mixing the two schemes.

However, if you must convert a balanced signal to an unbalanced signal without the use of a transformer, here is the proper wiring configuration:

Cable Conversion from Balanced to Unbalanced or Vice-Versa
UNBALANCED 1/4" or 1/8" Mini IN
Pin 1 (Chassis Gnd) Sleeve (GND - Minus Signal)
Pin 2 (Plus Signal) Tip (Plus Signal)
Pin 3 (Minus Signal) Sleeve (GND - Minus Signal)

(Figure 1)

XLR Wiring Illustration

(Figure 2)

Balanced XLR to 1/4" Unbalanced

If a TRS 1/4" or 1/8" Mini is used as Unbal, leave the Ring unconnected.
Note: When shorting the Minus and GND to convert to unbalanced, always make the short on the sending side... See example above.

UNBALANCED 1/4" or 1/8" Mini
Tip (Plus Signal) Tip (Plus Signal)
Ring (Minus Signal) Sleeve (GND - Minus Signal)
Sleeve (Chassis GND) Sleeve (GND - Minus Signal)
If a TRS 1/4" or 1/8" TRS Mini is used as Unbal, leave the Ring unconnected

Interfacing the Rack to the Transmitter:
One of the most important aspects of the wiring scheme is how to interface the audio processing rack to the transmitter. There are several ways to accomplish this. One important thing to remember is that you must pad the Line-Level signal down to a Mic-Level signal if you will be feeding the transmitters mic input jack on the front panel. A Line-Level signal fed into a Mic-Level input will result in distortion! You can use a 12:1 transformer to accomplish padding or a resistor network pad of some kind... But you MUST pad down the signal if using the mic input connector on your rig! If you will be feeding your rack audio to either a Balanced Modulator input or rear panel Line-Level Accessory input, then padding will not usually be necessary. Described below are some interfacing options...

As in wiring the rack pieces to each other, the best way to interface your audio rack to your rig is accomplished by using an audio isolation transformer. This is true whether feeding your rig's balanced modulator, line input, or mic input.

The most direct approach is the best! If you are using an analog transmitter, then feeding the Balanced modulator or line input (bypassing the noisy and frequency restricted mic amp) via a 1:1 transformer is the best way. If you are using one of the more recent SDR transmitters, then feeding the audio to the line input section in the transmitter via a 1:1 transformer is the best way to go for the same reasons mentioned above.


If your transmitter is equipped with a "line-level" input, I recommend the pre-made Jensen 1:1 PI-XX series of the "IsoMax" transformer box specifically for this application. This box receives a balanced line-level signal and sends a balanced or unbalanced line level signal out to the transmitter's line-level input and can be configured with a variety of input/output connectors. ($150 direct from Jensen) I use the Jensen PI-XR transformer (XLR bal in to RCA unbal out) and it's perfect for interfacing my audio rack to my ANAN's rear-panel RCA line input connector.

Jensen ISO-MAX Media-1

If you want to save
a afew dollars, you could build your own box using a Jensen 1:1 Line-Level to Line-Levell audio transformer model JT-11P-1. (About $70 direct from Jensen)

If your transmitter does not come equipped with a "Line-Level" input, and you must use a mic-level input, I recommend the excellent Radial Engineering J-ISO Jensen Transformer direct-box specifically for this application. This box receives a balanced line-level signal and converts it to a balanced or unbalanced mic-level out. (About $250 from Amazon)

Jensen ISO-MAX Media-1

If you want to save
a afew dollars, you could build your own box using a Jensen 12:1 Line-Level to Mic-Level audio transformer model JT-DB-EPC. (About $73 direct from Jensen) See figures 6 and 7 in the "Transformers" section below for Jensen line to Mic level transformer wiring.

See figures 6 and 7 in the "Transformers" section below for Jensen line to Mic level transformer wiring.

The benefits of transformer usage in this application are threefold; Hum rejection, RFI rejection and Line-to-Mic-Level Padding. This is a very elegant solution for feeding the transmitter's mic input connector!

As a last resort, you could simply wire your audio rack's balanced or unbalanced output directly to the mic connector as seen in figures 3, 4, and 5 below, using a simple resistive "Pad" circuit to accommodate your transmitter's mic input level.

(Note 1)
Some hum may be present in some transceiver configurations using the interface schemes of figures 3, 4, and 5 below since there is no isolation occurring, but I have generally found these to work satisfactory in most cases.

(Note 2) Yaesu does not provide a way to send "Balanced" audio to it's mic connector. For this reason, Figure 4 below shows an "Unbalanced T-Pad" instead of a "Balanced H-Pad".

(Note 3) The resistor values indicated in the "T-Pad" and "H-Pad" networks are based on 600 ohm inputs and outputs, where -40dB of attenuation is needed (Line-Level to Mic-Level conversion). If different impedance and/or attenuation values are desired, see the "T-Pad Calculator" page and enter your desired values.

(Tip) Use 1/4 watt resistors and wire them inside of XLR connector.

(Figure 3)

XLR Balanced Output to Kenwood 8 Pin Mic Plug
with -40dB Balanced Resistor "H-Pad" Attenuator
(For direct rack-to-rig, line to mic interfacing)

XLR Balanced Output to Kenwood 8 Pin Mic Plug

(Figure 4)

XLR Unbalanced Output to Yaesu 8 Pin Mic Plug

with -40dB Unbalanced Resistor "T-Pad" Attenuator
(For direct rack-to-rig, line to mic interfacing)

XLR Balanced Output to Yaesu 8 Pin Mic Plug

(Figure 5)

XLR Balanced Output to Icom 8 Pin Mic Plug

with -40dB Balanced Resistor "H-Pad" Attenuator
(For direct rack-to-rig, line to mic interfacing)

XLR Balanced Output to Icom 8 Pin Mic Plug

The graphic schematic below illustrates my wiring scheme after it is all put together...

NU9N Wiring Schematic


Audio Isolation Transformer Wiring Schemes
If you will be using an audio isolation transformer from line level to line level, regardless if it is balanced or unbalanced on either side of the transformer, you will want to use a 1:1 transformer that keeps the signals equal. An excellent transformer for this application providing the best frequency response and hum/buzz suppression is the Jensen JT-11P. The two graphics below show both Balanced to Balanced and Balanced to Unbalanced configurations:

(Figure 6)

Jensen JT-11P-1 (1:1) Transformer Bal to Bal Wiring
Jensen JT-11P-1 Balanced to Balanced Line Level Connections

(Figure 7)

Jensen JT-11P-1 (1:1) Transformer Bal to Unbal Wiring
Jensen JT-11P-1 Balanced to Unbalanced Line Level Connections

If you will be converting a "Line-Level" signal to a "Mic-level" signal, the best transformer for this application would be the 12:1 JT-DB-E providing about -26dB of attenuation. The following graphic displays the wiring of this transformer in a Balanced Line Level to Unbalanced mic level configuration:

(Figure 8)

Jensen JT-DB-E 12:1 Line Level to Mic Level Bal to Unbal Wiring

Jensen 12:1 Line to Mic level audio isolation transformer

Level Adjustments

Improper leveling can either cause distortion or a bad signal to noise ratio. I wanted to cover this issue first before getting into any of the individual processor setups. I have outlined below what I believe to be the best approach to leveling your entire system.

Before continuing, it should be noted that some audio equipment includes a "Level" button that allows the equipment to be operated at -10dBv (consumer levels) or at +4dBu (professional levels). If you are using balanced XLR or 1/4" TRS audio connections throughout your audio chain, then operate your gear in the +4dBu levels.

The main thing to remember is this; If you will be sending +4dBu signals to a device that operates at -10dBv, then something must be turned down! If you are sending a -10dBv signal to a device that operates at +4dBu, then something must be turned up!

For more details on proper signal levels concerning the +4dBu / -10dBv option, click this link:

There is usually two options for sending your external rack audio to your transceiver:
(1) Microphone Input
(2) Line Input

If possible, your transceiver's "Line-Level" input is the best place to send your rack audio. By using the line input, you will not be using the microphone speech amplifier circuit in your transceiver which can sometimes roll-off low frequencies and add noise.

Step 1 (Only if using the microphone input of your transceiver)
Adjust your transmitter mic gain for a proper ALC reading

Plug a good quality dynamic microphone straight into the radio's mic input (preferably the one you will be using permanently) and set your mic gain level for proper ALC. (Occasional deflection) Then LEAVE IT ALONE... You will NEVER touch the transceiver mic gain again!

Step 2
Adjust your mic preamp input level

Speak loudly into your mic and adjust the preamp input level so that the clip led just begins to flicker. Then back it off slightly. If you have a preamp input VU meter, then adjust it for 0dB on voice peaks. Make sure it will never exceed that. For more details, see the "Microphone Preamp Setup" section below.

Step 3
Adjust your mic preamp output level

If you have an output VU meter, adjust it for -6dB on peaks. If you have no metering for output, then get your reading from the input metering of the next processor in the chain. For more details see the "Microphone Preamp Setup" section below.

Step 4
Adjust the remaining levels in your rack

Continue down your audio chain adjusting inputs/outputs to -6dB. If you are using any of the DSP processors, such as the Behringer DSP1100P, then keep their digital outputs under -6dB or as suggested in your manual.

Step 5 (Only if using the microphone input of your transceiver)
Convert your line level back to mic level

You will need to drop the line level back to mic level before presenting the audio to your transmitter's mic input. This will require a 40dB drop in signal. This can be accomplished in different ways:

- Reduce the level output from the last processor
( You won't get a-40dB reduction )

- Purchase or make a 40db Resistor Pad

- Purchase a 12:1 step-down audio transformer

- Consider purchasing the
Jensen JT-DB-E 12:1 audio isolation transformer and build your own box, or just purchase a plug-and-play Radial Engineering J-ISO Jensen Transformer direct-box specifically for this application.

If you will be using a mixer to send the final audio to the transmitter, then you could pad the output of the mixer in conjunction with a decreased output from the last processor to get your -40dB cut.

The best solution is adding a 12:1 audio isolation transformer. And, there are some benefits beyond just padding. It would accomplish three things:

1) - Provide -26dB of attenuation
2) - Virtually eliminate 60 Cycle Hum/Buzz
3) - Keep stray RF from entering the transmitter via the audio chain.

Jensen makes the very best, sonically pure transformers for this purpose. Jensen can be reached via phone at: (818) 374-5857 or on the web at:

Step 6
Adjust the output of the last processor or mixer for ALC

The output of the last piece of audio gear just before the transmitter should be adjusted so that your transmitter ALC level is correct. Normally, you only want your ALC to occasionally flicker on the first LED. Reaching the ALC threshold indicates full saturation
of the signal before RF leveling starts to occur. Your audio compressor should be adjusted aggressive enough so that minimal ALC will be required.

Selecting a Transceiver

Of all of the equipment in your audio rack, this one is key! The transceiver will be the deciding "Bottleneck" of what information will ultimately be transmitted. The old saying goes like this: "Your final audio will only sound as good as the weakest link in your audio system!" That "weakest link" IS your transceiver! All of your audio equipment will more than likely be capable of 20Hz ~ 20kHz in frequency response. Listed below are my recommendations for stock transceivers that can do a nice job:

Apache Labs
Kenwood TS-950SDX
Yaesu FT-2000

There are many transceivers capable of good clean, full eSSB audio. The most popular and latest offerings are by Apache Labs and Flex Radio Systems. These Software Defined Radios (SDR's) can just about do anything that you want them to do in terms of receive and transmit audio response. The sky is the limit with these and with their outstanding THD specs and screen display capabilities, they're hard to beat!

If you're into older traditional rigs, the Kenwood TS-950SDX and TS-850S with the optional DSP-100 are good and are capable of a bit wider transmit response than the rest. The Kenwood TS-850S with the optional DSP-100 has the best and most versatile transmit audio quality followed by the Kenwood TS-950SDX. I am definitely a fan of old Kenwood audio but buyer beware - There is not much support left for these old aging rigs and in some circumstances, parts are difficult or even impossible to find!

Also, there has been a crop of offerings from Yaesu, namely the FT-2000 and FT-5000 that can easily accomplish 4.5kHz of transmit bandwidth with no effort. These rigs sound decent with just a decent microphone attached. But the new SDR's are king when it comes to features and capabilities.

It should be possible to perform these mods to most analog transceivers with similar filters changes and a direct Balanced Modulator Feed to produce similar results. It all depends on the bandwidth of the filter you use and an appropriate carrier set point alignment.

Suggested Audio Components

The following suggestions are just that, suggestions only. However, if you are price conscious, like me, these components will deliver excellent quality and a good "Bang-for-the-Buck"! My personal picks are as follows:

Click on a picture for website

Audix D6 Kick-Drum Dynamic Microphone

Fredenstein V.A.S. Mic Pre  - Click for web site
DEQ-2496 Vocal Mastering Processor
Behringer MX-882 Mixer

Sony MDR-7506 Closed-Type Headphones

I believe that these products offer the best "Bang-for the-Buck" without breaking your bank account too bad! I would consider this lineup sufficient for high quality eSSB.

Component Ordering

Audio component ordering is important!
First I'll recommend the ordering and then explain why.

Microphone Preamp
Noise Gate
Soft Compressor 4:1
Spectral Enhancer
Aggressive Compressor 12:1
Peak Limiter
Effects Processor

The first item in the chain is obviously the Microphone. There are three major types of microphones available. Dynamics, Condensers, (large and small) and Ribbons. While choosing a microphone type is a personal preference, I have found that "Dynamic" type microphones seem to work best for amateur radio and broadcast applications for a few reasons.

First, because of the design characteristics of dynamic mics, they are not as susceptible to background noise as their condenser and ribbon counterparts. (FYI: Ribbon mics are "Bi-Directional") Also, dynamic microphones tend to be more "directional" in their pickup pattern that help suppress unwanted sounds from the adjacent sides of the mic.
Finally, dynamic microphones have a much lower sensitivity and output than condensers making them inherently quieter in a noisy room.

There are many operators who use condenser microphones successfully, but dynamic mics are still the best for amateur radio eSSB audio in my opinion.

Microphone Preamp (Converts "Mic-Level" to "Line-Level" Signal)
Your microphone signal will need to be amplified to a "Line-Level" signal (about +40dB) before it can be fed into a line-level device input such as an equalizer or compressor. Compressors require appropriate signal levels to compress correctly.

Even though an equalizer can be fed with an un-amplified mic-level signal, the "signal to noise" ratio would suffer substantially. The internal "Op-Amps" in the EQ would not run at their designed specifications. Also, if you are using a condenser type mic, it will require "Phantom Power", usually 48 volts, to power the mic transistors.

Noise Gate (Optional)
This is usually the best place to insert a dedicated noise gate. The reason being that a noise gate has more dynamic range to work with since it appears before the compressor. After compression, there is little dynamic range left to distinguish between voice energy and background noise energy. Also, you will kill background noise up front before it is EQed and compressed. So, if a dedicated gate is used, insert it just after the preamp and before any EQ or compressor.

Soft Compression
If you have a spare compressor, you could use up front in the chain to provide some soft compression before anything else happens dynamically. A ratio of about 4:1 would be plenty and would provide some headroom for the processing that will follow.

Spectral Enhancers
(Aphex 104 Aural Exciter / Behringer EX3200 / BBE Sonic Maximizer)

These processors were designed to bring life to old lifeless recordings that lacked bottom and top end fidelity. They do this in a very interesting way by actually creating even-order harmonics in the audio that produce resonance in the lows and highs. To some, this may seem like a "Voodoo" approach to processing audio, but it can be very effective at producing low and high end resonance, if that is what you are looking for. Spectral enhancers should be understood as "Harmonic Coloring" devices that attempt to bring more of a pleasing tone to the voice or music source.

Note: In our application of Amateur Radio audio, spectral enhancers should only be used for polishing already good sounding audio. They were never intended to be a stand-alone device in establishing your sound. You should be able to produce excellent audio characteristics without any spectral enhancement at all! Over-use of a spectral enhancer will sound too thick and boomy on the low frequencies and will sound like a very shrill buzz-saw effect on the high frequencies. Use these devices with caution remembering that a little effect goes a long way!


This would be my choice for the placement of the next component. The reasons I would place the EQ BEFORE the compressor are as follows:
If the EQ were placed AFTER the compressor, high frequency energy from the EQ would be able to distort the audio since the compressor would not see it, therefor not being able to react to it.

With the EQ BEFORE the compressor, we can cut midrange frequencies from influencing the compression action of the high frequencies before it gets to the compressor. And, if the compressor has a built in low frequency filter preventing the lows from influencing the compressor, we have an elegant solution where we can adjust the compressor so that ONLY high frequencies will be limited aggressively, while the low and midrange frequencies will be compressed conservatively.

Compressor / Limiter / Noise Gate or Downward Expander
The Equalizer / Compressor order was discussed above. Notice that I have the compressor BEFORE the effects processor. The reason for this is because most compressors on the market these day have built-in noise gates. In fact, the majority of compressors have three processors built in, usually in this order: Noise Gate > Compressor > Peak Limiter

A noise gate or downward expander placed AFTER an effects processor would destroy the effects tail on the audio and would sound chopped off! We want to do the gating BEFORE the effects so that when the gate closes, the effects will still linger as long as you keep the PTT switch closed and not be chopped off prematurely.

If you were running a separate Noise-Gate, I would suggest that it would be placed right after the preamp. If you don't want to spend the extra money on a separate gate, those built into compressors do an adequate job. If you will be using a dedicated noise gate just after the preamp as discussed above, you will probably not want to use the built in gate in your compressor. If you do not have a dedicated noise gate after the preamp, then this would be the next best place to have it working, although it will not be as effective as it would be just after the preamp for reasons mentioned above.

Many manufacturers include a "De-Essing" feature in their processing, designed to tame high frequency dynamics that can cause distortion or uneven high frequency content. De-essers are dynamic rather than static in the way that they process high frequencies. While de-essers can be a handy tool in your audio processing arsenal, they are limited in scope to just a single band of operation. A more elegant approach would be the use of a "Dynamic EQ", where several frequency bands of dynamic control can be assigned at once. The Behringer DEQ2496 is such a device and takes care of high frequency processing beautifully!

Effects Processor (Optional)
Usually, you want the effects processor as the last piece. This puts the final polish on the audio, such as a virtual room effect, or perhaps some plating or mild reverb. Since the effects processor will not noticeably add or subtract any frequency content, the already processed signal should still be okay for the final transmission through the transmitter.

Audio Level Processors (Not Recommended for Amateur Radio SSB)
Aphex Dominator 720 / Compeller 320A / etc...
These devices are designed to insure constant output leveling. If you want constant "full duty cycle" type of output, then these devices help achieve this. In AM broadcast applications, these devices are essential. They keep the output constant and loud!

However, I have found that these devices in our applications do nothing more than increase background noise and work your amplifier harder than necessary. I feel that it is better to have a good directional antenna system where the RF dB's count more than any external audio "Duty Cycle" processing. In our application, it seems that SSB fidelity and heavy duty cycle processing do not mix well for various reasons. All that is needed is some moderate compression and perhaps some peak-limiting. Leave the heavy Audio/RF leveling for broadcast AM applications and keep your SSB Hi-fi audio processing to a minimum for a more natural and transparent effect.

Multi-Function AM/FM Processors - Orban, Omnia, etc... (Optional)
These processors are commonly used in broadcasting and audio production studios where workable bandwidth are 6kHz and beyond. While these devices are excellent in what they are designed to do, their application for Amateur Radio eSSB is not specific enough unless you are transmitting at 6kHz of audio bandwidth or more. Their flexibility is very limited below 6kHz bandwidths and tend to focus better with wider bandwidths and professional RF equipment. These processors are best suited for AM and 6K+ eSSB operations where agressive talk-power and audio density is desired. Personaly, I'm not a big fan of this type of processing because they tend to add some distortion to the final audio as well as robbing you of what little dynamics you have left after all of the other dynamic processing in the chain.

A mixer can be a great tool if you plan on combining several signals together to be routed to your transceiver, or if you want to split several signals to other devices from a single source. For example, you may have the need to combine your audio output from your rack with the out put from a recording device, or perhaps from two or more audio chains to be routed to your transceiver. You may also need to combine audio outputs from several rigs to a signal input to a stereo amplifier for better receiver audio reproduction.

There are several mixers available, from simple to elaborate, from cheap to very expensive. I have found that the low cost mixers, such as some of the Behringer models, work just fine for our applications. I especially like the Behringer MX-882 for its versatility in not only mixing but splitting signals as well. It's a one-rack space unit and can also level your line source to mic levels suitable for direct input to your transceiver's front panel mic input connector. And, it's only about $100. Most of the Behringer mixers are quite affordable and should provide you with enough features to accomplish your particular needs.

A Final Thought

In the big scope of things, find what works for you! Not every processor will work for your particular application. By all means experiment and find what works and what doesn't. You will eventually find the best processing for your needs. The suggestions above are based on my experiences and not anything that should be considered as bible.

Have fun experimenting while dialing in your ultimate sound.

Microphone Setup

Mic Switches
Microphones vary greatly of course. Some have switches on the case that can roll-off bass frequencies and boost high frequencies. I would recommend placing all switches in their widest positions so that the bass and treble from the microphone is is not held back.

If you need to cut some bass, do it on the EQ! It's better, I think, to let the EQ do the job of EQing and not restrict the mic in any way. If you do restrict the mic, you may not be able to recover it properly later on if needed.

Mic Placement

Mic placement may vary depending on what kind of mic you are using. For Dynamic mics, you should be about 2 to 4 inches from the front and maybe slightly angled away to prevent "Popping". The beauty of dynamic mics is the insensitivity to background noise and their directivity, as compared to condenser types. Just experiment to see what works best for you.

Condenser mics are much more sensitive and require you to extend your distance to anywhere from six to a full twelve inches away! Be careful... These babies are sensitive and background noise can be an issue!

Mic Pop Filters
There are two types of Pop Filtering on the market. The first is a simple "Sock" that fits over you mic. The second and more effective filter is a specially made mesh screen that is circular, flat and is positioned in front of the mic, via a separate adjustable boom. The mesh screens are typically more transparent with high frequencies than the socks and are a better choice in my opinion.

Mic Support System
There are several types of mic support systems available. The Inno heavy duty, fully adjustable boom arm is an excellent choice! It sells for $60.00 and is a real bargin! To see this product, click on the following link or point your browser to:

Microphone Preamp Setup

Input adjustment:
The mic input control should be adjusted so that the clip led only flickers occasionally. This would be about the equivalent to 0dBu on an analog VU meter. The reason we want maximum VU levels here is to optimize the Signal-To-Noise characteristics. If you experience high levels of background noise because of high VU levels from the preamp to the compressor, it's usually because the compressor "Threshold" level is set too low. See the "
Gate/Compressor/Limiter Setup" section below.

Output Adjustment:
The output control should be adjusted to drive the input of the next component at about 0dB. If the next component is a digital unit, then you may want to decrease this to around -3dB.

Tips: Warm vs. Clean
If you are looking to your preamp to produce a warm sound up front, then you will want to set the preamp input control a little high and the output control a little low. If, on the other hand, you are looking for your preamp to produce the cleanest sound possible, then set the input control a little low and the output control a little high. Experiment a little to find what settings sounds best to your ears.

Equalizer Setup

The equalizer is definitely the most important piece of equipment in your audio rack! Most transceivers do an excellent job of passing all the midrange frequencies between 300Hz ~ 2.5kHz usually with an added dominance between 500Hz ~ 800Hz. Unfortunately, most stock transmitters roll-off the bass frequencies below about 100Hz and down, as well as the high frequencies above about 2.7kHz and up unless you are using a modern SDR. With ALL rigs, we need to do basically four things:

* Reduce the Midrange
* Increase the Bass
* Increase the Treble
* Control the Resulting Dynamics

Taking a look once again at GRAPH 1 and GRAPH 2 above, you can see what an EQ can accomplish when set up. Below is a graphical representation of what EQing I had to implement in order to get some flatness out of my Kenwood TS-850S/DSP-100 after passing through its i.f. and DSP filtering.

NU9N Pre Transmitter EQ Emphasis

As you can see, I had to boost the Low and High frequencies by 8dB, and cut the mid frequencies by -22dB resulting in a difference of 30dB The resulting Kenwood TS-850S/DSP-100 transmitted audio is represented below in its full bandwidth: (Note: Do not try this with a transmitter that can not produce 6kHz naturally or severe filter blow-by will occur.)

NU9N Spectral Analysis in Flat 10kHz Receiver Bandwidth

The "Behringer DSP1100 Setup Page" link below was designed to accommodate 1100 users, but the EQing principle is applicable for all EQ's.
Click here to view the Behringer DSP1100/1124 setup page

A final note: Set up your EQ before you setup your Compressor. Get the sound you are looking for and ignore any distortion you may hear in your monitor or second receiver. You can address that later as prescribed in the compressor setup section. As an aid to help you see what happens with EQing and too much of it, refer to the tables below: ( EQ Information source: Alesis )

Voice Fullness at 120Hz; Boominess at 200 to 240Hz; Presence at 5kHz; Sibilance at 7.5kHz; Air at 12 to 15kHz

16Hz to 60Hz sense of power, felt more than heard makes audio muddy
60Hz to 250Hz fundamentals of rhythm section, EQing can change audio balance making it fat or thin makes audio boomy
250Hz to 2kHz low order harmonics of most musical instruments telephone quality to music, 500 to 1kHz horn-like, 1k to 2kHz tinny
2kHz to 4kHz speech recognition 3kHz listening fatigue, lisping quality, "M", "V", "B" indistinguishable
4kHz to 6kHz clarity and definition of voices and instruments, makes audio seem closer to listener, adding 6dB at 5kHz makes entire mix seem 3dB louder sibilance on vocals
6kHz to 16kHz brilliance and clarity of sounds sibilance, harshness on vocals

This next table may be used as an aid to determine the -3dB points of the lowest and highest roll-off frequencies when the Center frequency and Q factor or octave range is known. To convert Octaves to Q, consult the chart below.

Formulas for Determining -3dB fL & fH limits
where fC and Q are known

Q = fC / Bandwidth
fl = fC - ( fC / 2Q )
fH = fC + ( fC / 2Q )
Q = Quality Factor
fH - fL = Bandwidth
= known Frequency Center
fL = Frequency Low Spread
fH = Frequency High Spread
OCTAVES Q FACTOR freq. Low freq. High
2 Octaves
Q = .67
fl = fC x .254
fH = fC x 1.746
1.5 Octave
Q = .92
fl = fC x .456
fH = fC x 1.543
1.0 Octave
Q = 1.41
fl = fC x .645
fH = fC x 1.354
3/4 Octave
Q = 1.90
fl = fC x .736
fH = fC x 1.263
2/3 Octave
Q = 2.15
fl = fC x .767
fH = fC x 1.233
1/2 Octave
Q = 2.87
fl = fC x .826
fH = fC x 1.174
1/3 Octave
Q = 4.32
fl = fC x .884
fH = fC x 1.116
1/4 Octave Q = 5.76 fl = fC x .913 fH = fC x 1.087

Example using the information above:

Known Center frequency (fC) = 1000Hz
Known Bandwidth in Octaves = 1/3 Octave (Q of 4.32)

fL = fC - ( fC / 2Q )

fL = 1000 - 1000/2 x 4.32
fL = 1000 - 1000/8.64
fL = 1000 - 115.7
fL = 884.3

fH = fC + ( fC / 2Q )
= 1000 + 1000/2 x 4.32
fH = 1000 + 1000/8.64
fH = 1000 + 115.7
fH = 1115.7

To confirm the fL & fH results
using the Q formula:

Q = fC / Bandwidth ( fH - fL )
= 1000 / (1115.7 - 884.3)
Q = 1000 / 231.4
Q = 4.32

This last two table are excellent resources. The first developed by Bell Laboratories, courtesy of AE4FB, is used to determine the desired Low or High frequency -3dB amplitude points based on the bandwidth of the signal to insure even distribution of high and low frequencies based, I'm amusing, on a 3dB/Octave distributed amplitude decline across the entire spectrum of available frequencies..

The second is a modification of the first assuming a -1.5dB/Octave distributed amplitude decline, instead of -3dB/Octave, across the entire spectrum of available frequencies, thus dividing the formula by half. This one will produce a much more robust low end and have a smoother sound.

(Assuming a Constant -3dB/Octave Amplitude Decline)

Formula 1
600,000 / available high freq = required low frequency @ -3dB

Formula 2 (Inversed)
600,000 / desired low freq = required high frequency @ -3dB


Using formula 1 where a known high frequency cutoff is 3kHz then:
600,000 / 3000 = 200 Hz @ -3dB
This means that for an audio bandwidth of about 3kHz, we should be rolling off below 200 Hz with 200Hz already dropping by -3dB.

Using formula 2 with a desired low frequency of 100Hz, then:
600,000 / 100 = 6000 Hz @ -3dB
This means that for a desired low end response of 100Hz, then the top frequency must be 6000Hz at the -3dB point.


(Assuming a Constant -1.5dB/Octave Amplitude Decline)

Formula 1
300,000 / available high freq = required low frequency @ -3dB

Formula 2 (Inversed)
300,000 / desired low freq = required high frequency @ -3dB


Using formula 1 where a known high frequency cutoff is 3kHz then:
300,000 / 3000 = 100 Hz @ -3dB
This means that for an audio bandwidth of about 3kHz, we should be rolling off below 100 Hz with 100Hz already dropping by -3dB.

Using formula 2 with a desired low frequency of 80Hz, then:
300,000 / 80 = 3750 Hz @ -3dB.

This means that for a desired low end response of 80Hz, then the top frequency must be 3700Hz at the -3dB point.

Gate/Compressor/Limiter Setup

The audio compressor, in my opinion, is the second most important piece of equipment in your audio arsenal. A poorly adjusted compressor will either sound too "Squashed", or not properly limit frequencies that will cause distortion at the transmitter. You may also experience heavy background noise due to an improper setting of the Threshold control. We really only need just enough compression/limiting to get the job done and no more. With the settings of your Threshold, Ratio, Attack and Release, you should be able to compress and limit effectively and transparently.

Noise Gate:
The noise gate should ONLY be used if your background noise is minimal! The reason I state this is because if you have a high background noise level in your ham studio, the noise gate will bring more attention to it than not gating at all.

It's true that the gate will filter out the noise between words, but as soon as you speak above the gate threshold and the gate is released, the noise will still be there! In fact, it will be there for a few milliseconds even after you speak until your gate reacts to the signal falling below the gate "Threshold". It is this transitional time that your gate will be very obvious to the listener. They will here a rushing noise and then hear the gate take it out until you speak once again. In short, it will sound as though you are running VOX! Yuk!!!

So, if you have to use a noise gate, minimize the background noise as much as possible first. The more noise you kill at the source, the more transparent the noise gate will be. Use the lowest possible "Threshold" you can. With only your background noise present, adjust the "Threshold" just to the point where the gate closes. If you have separate"Release" or "Ratio" controls, you will just have to experiment until you find the settings that work best in your noise environment. Conclusion: The BEST gate is NO gate! Clean up the background noise at the source!

Compressor / Limiter:
There are many ways to apply compression. If you do NOT have a separate Limiter circuit in your compressor, than I would recommend setting it up in such a way as to reap the benefits of both compression and limiting. To do this, set up the controls as follows:

Mode: Manual
Ratio: 5:1
Attack: .1 mSec or less
Release: .05 Sec or less
Threshold: Adjust for about 15dB of Gain Reduction during SSSing
Output Gain: Adjust for proper ALC on transmitter

Your threshold setting will depend on the levels entering your compressor. With your EQ adjusted the way you want and the above settings entered into your compressor, speak normally into your mic using words with a lot of "SSS"ing. During the "SSS" portion of your words you may hear some sibilance, distortion or tearing effects. Simply lower the "Threshold" of compression until these artifacts are removed.

The best test is to produce a long "Snake Hiss" sound like... "SSSSSSSSS". During your "SSSSSSSSS" adjust the threshold until it is clean.

If you want to try out some verbiage that I created for this test, click on one of the links below to either "View" or "Download" my "Sibilance Test".


If you have a separate Limiter section, then you can be less aggressive with your compressor. I would recommend the following settings for some mild compression:

Mode: Manual
Ratio: 2:1
Attack: 10mSec
Release: .5 Sec
Threshold: Adjust for about 3dB of Gain Reduction

Don't get carried away! We just want to control the peaks and add a little boost (3dB) to the final output. A 3dB boost will effectively double the sound of the audio!

A final note about the compression "Threshold" setting: If you experience high levels of background or room noise, it is most likely the result of over compressing. This is NOT caused by high input/output leveling, but rather by too much signal being processed by the compressor via the threshold level control. Again, we only want to adjust the compressor threshold so that normal speech produces about -3dB of gain reduction. The high frequencies will be compressed more aggressively, but most room noise will fall in the area of 30Hz ~ 600Hz, so high frequency gain reduction is not a factor.

If the compressor/limiter has been set up correctly, the Bass and Midrange frequencies should produce about -3dB of gain reduction, and the high frequencies (2 ~ 5kHz) should produce about 10 - 15dB of gain reduction.

Peak Limiter:
If you have a separate peak Limiter circuit, it is usually the last module in your processor. Adjust it so that the loudest peaks do not exceed the absolute maximum output that you want.

Output Gain:
If this will be last processor in your rack before the transmitter, then after all of the above modules have been adjusted, adjust the output gain of your Gate / Compressor / Limiter so that you are back to the original ALC level that you established in STEP 1 of the "Level Adjustments Setup" section. If you will be using an effects processor after the compressor, then adjust the output gain for about -3dB and continue with the Effects Processor Setup.

Effects Processor Setup


Effects Processing
Effects processors can add a final polish to your hard earned audio, making it sound somewhat bigger and brighter with more effervescence as opposed to dry and lifeless. Adjusted properly, the effects should add a very subtle liveliness and slight sustain to your high frequencies.

The desired effects that you want is a personal preference, but I would like to suggest a few things about effects setup that I believe sound the best.

Reverb vs. Echo
I often hear effects that sound more like an echo or bathroom than a reverb. This can occur when the initial signal delay is too long in time and is usually the result of selecting a small room effect. An echo sounds like the original signal being repeated one or more times within a given time frame. It sounds more like multi-path propagation than a concert hall. This is common in the CB world and in my opinion sounds horrible. A true reverb effect, on the other hand, sounds more like a smearing and continuation of the original signal giving the effect of a large room or hall where the signal gradually decays in a very smooth transition. So try initially selecting a large room or concert hall effect and work out the details from there.

A Cleaner Sounding Reverb Effect
All too often, I hear effects out there with way too much midrange and bass reflection, resulting in a very boomy and muddy sounding midrange in the final sound. This is not a desirable sounding reflection in my opinion. A great sounding reflection is one that highlights your high frequencies and not the low to mid frequencies as much. This can be accomplished easily by using the internal EQ of your effects processor. Set your effects processor's high frequency EQ to maximum, and your effects processor's low frequency EQ to minimum. By setting up your effects processor's EQ in this way, the high frequencies will be the only segment of your audio bandwidth with reflections. The result will be a very clean and lively sounding high frequency effect without all of the low and mid frequency muddy sounding reflections.

Inline vs. Side-Chaining
The absolutely best way to apply your effects processing is via a "side-chain" path rather than "inline". Every time your audio passes through a piece of equipment, there will be more degradation, artifacts, noise, and coloration added to the original audio. For this reason, it is best to "mix" your effects in a separate path from the main audio chain. In other words, your main audio path will NOT pass through the effects processor at all! The effects only will be mixed in separately and in parallel via a separate channel in your mixing console. This is referred to as "Side-chaining" the effects. For detailed information on using your mixers side-chain, consult your mixer owners manual.

Effects Level
How much of the effects should we mix in with the dry signal? Not much! I have found that the best effects level is one that isn't even noticed until you turn it off! No more, no less. Effects are just a way to put a final polish on our high frequencies without anyone really noticing the effects themselves. We want to draw attention to our voice, not our effects! So be careful not to add too much of a good thing.

For a nice effect, try a thin or light "Plate" or a medium room program. Again, experimentation is key. Find what you like!

Some Initial Settings to Play With
Here is a profile created by Greg, W5UDX, for the the Behringer DSP-2024P with the following "Plating Reverb" parameters:

Effect Button: PLAT
Edit A: PRE.D = 0.010
Edit B: DECA = 2.358
Edit C: DAMP = 10
Edit D: SIZE = 33
Edit E: SHV.D = 32
Edit F: DIFF = 20
EQ Low: BASS = -16
EQ Hi: TREB = +16
MIX: 6 to 9 (User preference if using the DSP2024 inline in "INTN" mode)

Here is a modified version of Plating using the Behringer DSP2024 that I personally use, with just a little less reverb length and depth:

Effect Button: PLAT
Edit A: PRE.D = 0.010
Edit B: DECA = 1.542
Edit C: DAMP = 15
Edit D: SIZE = 10
Edit E: SHV.D = 20
Edit F: DIFF = 20
EQ Low: BASS = -16
EQ Hi: TREB = +16
MIX: 7 to 10 (User preference if using the DSP2024 inline in "INTN" mode)

Make sure the DSP2024 is set to "Mono" mode.
(Press "Setup" then turn "INPUT" knob to "MONO".)

I also recommend using the DSP2024 in a "Side-chain" configuration where you can mix in the desired effect in a separate channel of your mixer.
If using the DSP2024 in a sidechain, make sure to set up the DSP2024 in "EXTN" mode instead of "INTN" mode.
(Press "Setup" then turn "OUTPUT" knob to "EXTN".)

Transmitter Setup

Turn off any internal transmitter EQ or Compression. Let your external EQ and Compressor do this. Whatever transmitter you are using, set it up for the maximum transmit bandwidth possible.

In the future, I will update this "Transmitter Setup" section to include as many transceiver specific setups as I can. If you would like to contribute to this with a setup of your radio to include in future versions of this page, please send me an e-Mail with your setup data, menu settings, etc...
Please submit to:

The eSSB Carrier Phenomena

"Do I Hear an "eSSB" Carrier?"
There are evidently some folks on the air who have been reporting that some of the "hi-fi" guys have a carrier in their audio as a result of their wide bandwidths. I just want to briefly address this issue and discard any notion that this is happening as a result of any external or internal processing, bandwidth or any low frequencies below 80Hz that are being fed to the transmitter.

When an SSB station has a relatively wide audio response and is modulating frequencies at or below 100Hz, an interesting phenomena takes place. When tuning off-frequency on such a given station, the listener may perceive what he/she thinks is a carrier. For example, if you are listening to someone on USB, and tune down frequency by about 500Hz, if the transmitting station is modulating 100Hz with any energy, then you will hear apparent tones at 600Hz. But, ONLY when they are talking!

This is NOT a carrier that is being heard, but the 100Hz tone in the voice being off-beated up by 500Hz resulting in 600Hz tones via modulation. The reason that this phenomena is not usually noticed as much on the average narrow-band SSB signal, is because there is not enough low frequency energy to bring up to a higher pitch to start with.

Just in case some believe that hi-fi audio effects a transmitter's carrier suppression, that's a hoax! The audio that you feed to the transmitter has NOTHING at all to do with the carrier suppression adjustments in a transmitter, especially with the newer DSP controlled rigs!

If a hi-fi station keys the mic but does not talk, and their background noise is suppressed sufficiently, you should hear no carrier sounds at all. If you do hear a carrier while they are not modulating, then they may have a carrier suppression problem. But that is a totally separate issue from their audio being fed to their transmitter. A transmitter carrier suppression problem will still be evident even if the mic input signal is disconnected!

Most modern DSP controlled transceivers like the Kenwood TS-950SDX,TS- 870S, TS- 850/DSP-100, Icom 756 Pro, Yaesu FT-1000MP etc... have transmitter carrier suppression figures better than -60dB or more!

Speech is a complex thing…
Human speech, as generated by the vocal chords, is very rich with harmonics. The way that vocal chords vibrate with passing air makes for interesting viewing on a spectrum analyzer. Of course, the vocal chords are not the only component to speech…

There is also the changing shape of the mouth, producing the various vowel sounds, as well as air passing between the top of the tongue and roof of the mouth producing the high “hissing” sounds heard in “S”, “T”, and “CH” consonants. We shouldn’t forget the “P” sounds produced by little explosions of air between the upper and lower lips… All components of speech. However, for this discussion, I will just be focusing only on low frequency tones produced by the vocal chords themselves.

SSB Only!
When a voice is listened to on Single Sideband that has been properly tuned on frequency (zero beat), all is well… Just as it would be if listening in the AM or FM modes. However, if miss-tuned, where voice pitch is raised slightly (up frequency on LSB, or down-frequency on USB) something rather interesting happens as a result of the mistuning. This strange mistuned phenomenon ONLY happens in the Single Sideband mode! It cannot be reproduced with external audio pitch shifting devices or by speeding up a recording… Only in SSB… Why? It has to do in part with how Single Sideband is processed and filtered in the RF/IF/AF stages of a receiver, and how we are trained to listen in respect to low, midrange, and high frequencies in context.

When audio pitch is raised as a result of mistuning (accelerated), the lower frequency contents of the vocal chords are also raised. This phenomena is present on ALL mistuned SSB signals, but is much more pronounced when the transmitter station has good low frequency response, well below 100 HZ.

Let’s assume that someone with a low rich voice is using a transmitter with good frequency response down to about 50 Hz. Let’s also assume that the low-frequency content of that person is prominent from 60 Hz. to 100 Hz. Now, you purposely mistune this person by raising the pitch of your receiver by 300 Hz. What you now hear are his bass components being remodulated and heard at 360 Hz. to 400 Hz. Instead of sounding like bass, it now sounds like lower midrange tones wobbling around as they speak!

Because lower frequencies of the voice are more prominent than the upper frequencies of the voice, (by 80%) the remodulated bass tones will sound separate and raised in volume over the upper frequency components of the voice. This will give an illusion of separation between the two that you never noticed before. This is because of how we listen to normal speech. With normal speech, we reference and interpret the bass, midrange, and high frequency components of speech in a certain way that we accept.

If we shift all of these frequency ranges up by 300 Hz., we have now moved the bass components of speech up into the midrange area of our hearing. Now we interpret it much differently than what we are used to. Since we have effectively removed the bass component of speech by raising it to the midrange area, our brain interprets the tones as being incorrect information. Bass components of speech do not belong in the midrange area of speech! So every time they talk, you hear these base tones remodulated into the lower midrange area of your receiver and of your hearing, along with the other components of speech that are also present..

This phenomena will be most prominent if the transmitting station has good low-frequency response to begin with, and low-frequency energy in the source voice. ALL receivers (wide or narrow) will hear the phenomena because all receivers can hear remodulated frequencies above 300 Hz. easily!

As mentioned earlier, ALL mistuned SSB signals will exhibit this phenomena to some degree and can be heard if you listen carefully – Even female voices, with little low frequency energy, will exhibit this phenomena! In fact, women are the easiest for me to tune in because of this fundamental rule of listening; “They only sound harmonically normal at one specific frequency and not normal anywhere else."

On the other hand, the hardest signals for me to tune-in correctly are men with very low voices, but with no bass content below 100 Hz. in their transmitted audio response. It drives me crazy to get their frequency right because they can actually sound harmonically normal on several specific frequencies! So in this case, a critical low-frequency reference is missing that would significantly aid in the proper tuning of a signal. How about that—eSSB can be a necessary evil after all!, HI.

Bottom Line
Basically, all you are hearing with this phenomena are low frequencies, remodulated to midrange frequencies, and then mentally interpreting bass as midrange. It’s very awkward… as it should be.

Even though eSSB transmissions do occupy more bandwidth, it is not the upper frequency bandwidth that is the cause of this phenomena. This phenomena is only caused by vocal energy in the lower bass range below about 100 Hz. So extended high-frequency bandwidth has nothing what so ever to do with this phenomena.

Troubleshooting R.F.I.

60Hz Hum & Buzz

You've connected your audio gear, set it up, tested it into your dummy load and are ready to present your hard earned delicious audio to the world. You flip your switch from dummy load to antenna, key up and...... Whooooooow........ what was that?
Radio Frequency Interference like crazy! If there is one thing that will drive you absolutely crazy, it's RFI. Hum and Buzz are easier to cure, but can still prompt you to get out a bottle of the Extra Strength Excedrin!

Look at some of the following sections and see if there is anything you may be able to do that you're not doing currently to prevent RFI or Hum from letting you enjoy your audio. Don't worry, I've been through the RFI scenario and can tell you with confidence, there IS a solution for it. Determining the solution may take a bit of research, but it can be dealt with!

Antenna Orientation:
If your antenna is relatively close to your operating position, your battle with RFI may be a difficult one! The further you can get your antenna(s) from your shack or vice-versa, the better! Of course, this may not be the most practical solution in your unique installation, so below are some general suggestions to resist the strong RF fields already present in your shack. But if possible, steps should be taken to minimize the RF entering your shack at the source.


A good RF earth ground is worth it's weight in gold when it comes to a clean transmitted signal. Unfortunately, some of us do not have the luxury of having a true earth ground only 4 ft. away and/or good ground conductivity. The following, is a summary of some basic grounding guidelines:

1 Keep the ground path from earth to station as short as possible!
2 Run multiple length ground cables from earth to station.
3 Make sure your tower legs are well grounded.
4 Use heavy braid for your ground connections.
5 Tie your station AC ground into same earth ground that your station uses. This keeps the ground potentials at a minimum and is code in some locations for lightning protection.
6 Use a series "point-to-point" grounding configuration for all ham and audio equipment. Connect the audio related chassis branch in series to your ground bus. Connect the exciter, RF amplifier, power supply, and ham gear branch in series to your ground bus. Eventually, everything will be grounded to the ground bus, but in series branches instead of separately. Even though the "Star-Ground" method has been widely accepted as good engineering practice in the past by some manufacturers, it's NOT recommended by current engineering standards.

(See this excellent paper bt Jim Brown, K9YC on this subject)

Building an Indoor Counterpoise:
(The following text is from the Radio Works catalog)
Counterpoise Length
160 meters - 123 - 136 feet
80 meters - 65 -70 feet
40 meters - 34.5 feet
30 meters - 24.3 feet
20 meters - 17.3 feet
17 meters - 13.5 feet
15 meters - 11.6 feet
12 meters - 9.8 feet
10 meters - 8.6 feet

"If you cannot get close enough to earth to run a very short ground wire and install a good quality ground system, try a counterpoise. An easy example of a
counterpoise is the ground plane used with vertical antennas when they are mounted high in the air." "In its simplest form, a counterpoise can be a single wire, one-quarter wavelength long or just slightly longer. For best results, a separate wire is required for each band. If you really want to get elaborate, use two or more wires routed in different directions to make up your counterpoise. The wires for different bands may be close together, insulated and routed in a convenient way around a room." "You can probably eliminate counterpoise wires for bands that are harmonically related in odd multiples. 15 and 40 meters or 80 and 30 meters are examples."

AC Line Isolation:
Use a good AC power strip with RFI/EMI/Surge and Isolation built in to distribute your AC power. Also, get your AC power from ONLY one AC source. Do NOT use multiple AC circuits, or AC ground loops will develop. The Triplite "Isobar" AC strip seems to work well.

Audio Cabling:

For maximum RF rejection, use high quality double-shielded, low capacitance twisted 4 conductor "Starquad" audio cable for all of your audio and speaker connections.

Transmission Lines:
Use high quality COAX with your antennas that use that type of transmission line and check your station jumpers from time to time for bad connections. If you will be using 300 or 450 ohm transmission line, and it enters your shack to your tuner, you may be dealing with RF entering via this route. Some hams use a remote Balun at the end of their balanced transmission line that terminates to coax just before entering the shack, helping to reduce RF entrance.

Shielding Unshielded Cables:

If you have any rotor control or AC line cables that are not shielded, you can wrap aluminum foil around them to help keep RF out. W5GI has reported that wrapping steel wool around these types of cable is also very effective.

Audio Isolations Transformers:
For absolute isolation between all audio components that ultimately break up the ground loops, use audio isolation transformers between every piece of audio equipment including one between the audio rack output to the transmitter mic input. Not only will this break up the ground loops, but also keep 60 cycle hum to a minimum... (-120dB). And as an added bonus, it will keep RF out of the signal path by presenting a very high impedance for RF while transferring your audio signal through the chain.

I use the Jensen JT-DB-E for my interface between my rack and the transmitter mic input. They work great! The following link will navigate you to this site:.

RF Chokes:

If needed, choke all of your RF and AF inputs in the system. "Mouser Electronics" (as well as others) makes a variety of toroids for all of your cable needs. What usually works the best for Amateur Radio HF applications and frequencies are the following:

For use with:
AC Line Cords
Control Cables
RF signals via Coax
Audio Cables
Speaker Wires
Mouser Electronics
RFI Suppression Snap-On Mix 31 Ferrite Beads
Inside diameter: 6.6mm
Click part link below:

Part number:

Bypass Capacitors:
RFI Bypass Capacitor Installation

Another trick you can use to suppress RF is the installation of ceramic disc bypass capacitors. For
balanced audio signals, use .001µF ceramic disc caps between plus and chassis ground, and between minus and chassis ground. This can be accomplished either in the equipment, or better yet, in the XLR shell to your equipment input. Place the .001µF caps between pins 1 & 2, and between pins 1 & 3. Thanks to K6JRF for this tip! For unbalanced connectors, place one between the Tip and Sleeve lugs inside the 1/4" plug.

For AC power bypassing, use .01µF 2000 Volt ceramic disc capacitors between chassis ground and hot. For grounded or balanced AC, use 2, one between chassis and one side of the AC line, and another between chassis and the other side of the AC line. Make sure that you use 2000 WVDC caps for AC so that there will be no breakdown or excessive heating.

Line Isolators:

The use of Line Isolators in the coax connections can create a very high impedance for RF traveling on the outer coax braid while allowing the normal RF to pass through and eliminating RF ground loops that contribute to the problem. These can be placed between the transmitter and amplifier and between the amplifier and tuner. These line Isolators have SO-239 connectors on each end. "" sells some very effective line isolators / Common Mode Chokes. Website here:

Feedline Baluns:

The use of "Current" type Baluns are far superior to the common Voltage type and can distribute the RF to your antenna more effectively. "The Radio Works" makes some very good Yagi and Dipole Current type Baluns which act like line isolators helping to cut down the RF from returning to your station, effectively de-coupling the Feedline from the antenna. These Isolators may also improve your Beam's front-to-back and front-to-side rejection.

Audio Component Location
In some cases, simply relocating an audio device may have a profound effect on RFI through RF inductance. It is always a good idea to keep power amplifiers and Rotor Boxes as far away from audio mixers and audio processors as possible.

Cable Organization
It is always a good idea, and good engineering practice to organize your cable bundles by type, and keep them separate from other types of cables. For example, keep all of the AC line cables together and as separate as possible from the Audio Cables. The same holds true for Rotor and Data Cables. Additional, if at all possible, keep the cables as horizontal as possible since most noise is vertically polarized. This may be an exercise in futility, but at least you will feel good about a clean and processional cabling installation. Make it a weekend project when you're really bored! HI.

Also, keep open-wire type feedlines (300 ohm, 450 ohm, etc.) away from coaxial feedlines if possible, to reduce any RF inductance or influence from occuring.

Ground Loops and Ground Lifting

"Ground Loops" are your enemy! These occur when equipment has more than one path to ground causing 60Hz Hum or even worse, R.F.I. See the illustration below where a piece of equipment is grounded three times causing multiple ground loops:

Multiple Ground Loops

Your equipment should be grounded once and ONLY once! In a typical installation, where XLR pin 1 and the AC Ground pin is common to the chassis (Standard manufacturer practice) one ground loop will occur. If an additional ground is added via a chassis screw to your shack's RF ground system, three ground loops will occur.While this seems like a good idea, it can allow multiple circulating RF currents to cause severe R.F.I. The solutions here is easy:

1) Use an Audio Isolation Transformer to break the XLR GND path between all rack equipment.
2) Use an AC "Cheater" adapter to break the AC 3rd-pin ground.
3) Use a chassis screw and connect it directly to your shack's RF ground system.

It is much better to rely on your stations RF ground than the ground provided by your AC service, since your AC service ground may be too far to true earth to do any good and may also cause your entire house to act as an antenna for RF pickup.

You could simply rely on the XLR chassis ground alone, but then RF currents will use your audio cable shield as the path to ground and this is not desirable because of interaction between multiple rack gear.

Remember, you will want to make sure that you have a good earth ground connected separately to the chassis of each piece of equipment.


I know this has been a long page and I have tried to keep it as simple and organized as possible. But when involved in technical matters, things can get quite wordy. I hope this page has provided some assistance and would appreciate any comments or suggestions on how it could be improved. Also, if there are any technical errors or typos, please drop me a note and I will make the changes.

Thanks and may your pursuit for high quality SSB audio be realized without too much frustration!


-John (NU9N)

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