Setting Up Your Transmitter Audio
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 SSBaudio that you have been
First, let me set the record straight; If you are looking for near AM broadcast quality audio on 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, a very wide open, more pleasing sound than the typical Amateur SSB station, with little emphasis on
the upper midrange frequencies from 1kHz ~ 3kHz. The idea they have is this: The flatter, the better! Simple!
The typical Amateur Radio SSB audio has a very emphasized midrange characteristic to it by design, especially in the 400Hz ~ 1kHz area. Another
attribute with this kind of audio is a severely rolled off bottom end below 300Hz and a tight roll-off above 2.5kHz. 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 it's narrow bandwidth and saw that it worked good under poor signal conditions, Amateur AM for
all practical purposes was abandoned. Fortunately, for those who loved the wider and more pleasing sound of AM, it's still around, barely.
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 at least capable of 3kHz 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 phsyco-acustically sound
like AM broadcast!
I'm not saying that anything less than a 3kHz bandwidth will sound bad. I'm only trying to point out that it will NOT sound remotely
close to what AM broadcast sounds like. 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. Several time 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. So, even though a perfect reproduction of our voices would be ideal, it isn't going to happen because of our limited
bandwidth. Instead, like AM, the best we can do is manipulate the audio in such a way as to keep the audio as rich and full as we can.
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.
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.
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
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 best way to connect
your audio gear together is via the Balanced XLR or 1/4" TRS connectors onyour 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. Furman has line-level-to-line-level pre-made dual transformer boxes made with both
XLR and 1/4" TRS connectors for all Balanced/Unbalanced configurations. This product is called the Furman IP-2B and sells for about $80.
- 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 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
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 Red, White, Green and Black, wire the
cable as follows combining the same two colors on one end with the same two on the other end. If your cable colors are different, just use
the same two colors for a connection on both ends as in the following example:
Starquad Wiring for Balanced XLR to XLR Connections
|Pin 1 (Shield - Chassis Gnd)
|Pin 2 (Plus Signal)
White & Black
|Pin 3 (Minus Signal)
Red & Green
Starquad Wiring for Unbalanced 1/4"
or 1/8" Mini-Plug Connections
1/4" or 1/8" Mini-Plug
|Sleeve (Shield - Minus Signal)
|Tip (Plus Signal)
White, Black, Red
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
|Pin 1 (Chassis Gnd)
||Sleeve (GND - Minus Signal)
|Pin 2 (Plus Signal)
||Tip (Plus Signal)
|Pin 3 (Minus Signal)
||Sleeve (GND - Minus Signal)
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.
BALANCED TRS 1/4"
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
If a TRS 1/4" or 1/8" 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 Lline-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, post mic amp section or mic input.
- The most direct approach is the best! If you are using an analog transmitter, then feeding the Balanced modulator (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 DSP enhanced transmitters,
then feeding the audio just after the mic amp section in the transmitter via a 1:1 transformer is the best way to go for the same reasons
I recommend the excellent W2IHY "i-Box" created by Julius Jones specifically for this application. This box
accomplishes a number of tasks and is a very convenient piece of gear to have in your shack. It can do the following:
||Convert line-level to mic-level
||Convert High Impedance to Low impedance
||Convert Low impedance to High impedance
||Isolates AC & RF ground loops via an audio transformer and RFI reduction networks
||Can be used in line-level or mic-level applications
||Pad can be adjusted between -1dB to -40dB
||Can interface to any rig and supplies a PTT control jack
Also available from W2IHY, is the "iPLUS" Splitter device that can split your rack
audio to up to 3 tranceivers so that switching between rigs is as easy as turning a knob. The iplus does all the things the iBox does,
plus the following:
||iplus technology allows routing of your external Rack or W2IHY audio to 3 radios. No more
disconnecting and reconnecting your rack to separate radios. Just select your radio and play!
||Separate level controls for each radio
||RCA linear keying inputs from 3 radios. Select your radio, and your amplifier is connected
and ready for that radio
||RCA stereo audio inputs allows you to use 1 stereo or mono speaker system for all of your
radio’s receiver audio
||Combination XLR / ¼” TRS audio input or 5-pin DIN audio input for total flexibility
and compatibility with your balanced XLR, ¼” TRS, ¼” TS, or existing W2IHY cables
||Audio Phase Reversal for proper AM audio asymmetry. A must for maximum AM modulation and
||2 iplus units may be connected together to control up to 5 radios
- The next best way is to use a 12:1 Line-Level to Mic-Level audio transformer between the audio rack output
and the transmitter's mic input. 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. 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, 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.
XLR Balanced Output to Kenwood 8 Pin Mic Plug
with -40dB Balanced Resistor "H-Pad" Attenuator
XLR Unbalanced Output to Yaesu 8 Pin Mic Plug
with -40dB Unbalanced Resistor "T-Pad" Attenuator
XLR Balanced Output to Icom 8 Pin Mic Plug
with -40dB Balanced Resistor "H-Pad" Attenuator
The graphic schematic below illustrates my wiring scheme after it is all put together...
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:
Jensen JT-11P-1 (1:1) Transformer Bal to Bal Wiring
Jensen JT-11P-1 (1:1) Transformer Bal to Unbal Wiring
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:
Jensen JT-DB-E 12:1 Line Level to Mic Level Bal to Unbal Wiring
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.
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!
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.
Adjust your mic preamp output level
If you have an output VU meter, adjust it for 0dB 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"
Adjust the remaining levels in your rack
Continue down your audio chain adjusting inputs/outputs to 0dB. 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. Their inputs should be okay at 0dB however.
Convert your line level back to mic level
Note: If you will be inputting your rack audio directly to your transceiver's Balanced Modulator or a mixer, you can skip this step!
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 the W2IHY "i-Box" interface. Click
here to view...
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
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 great transformers for this purpose and cost around $70 for just the transformer, or about $200 for a dual transformer pre-made
box called the ISO-MAX®. Jensen can be reached via phone at: (818) 374-5857 or on the web at: http://www.jensen-transformers.com
Adjust the output of the last processor or mixer for ALC
After implementing your line to mic level step-down, the output of the last piece of audio gear just before the transmitter should
be adjusted so that the original transmitter ALC level that was performed in step 1 is still the same.
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
There are many transceivers capable of good clean, full SSB audio. The Kenwood TS-950SDX and
TS-850S with the optional DSP-100 are my personal favorites 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
The Yaesu 1000MP, and some Icom's sound very nice when properly set up! Again, the key is to set up or modify any radio you might be using so
that the transmitter can cover at least a 3 kHz or more transmit bandwidth.
A good example of this, is audio produced a Kenwood TS-820S. The mods involved replacing the stock 2.4kHz 8.83 MHz i.f. filter with an INRAD
4kHz filter (model #475) and feed the line level audio output from an audio rack directly into the 820's Balanced Modulator. Also, a readjustment
of the Carrier Set Point for a 70Hz reference on the low end. The resulting -6dB points on the unprocessed SSB audio is 70Hz ~ 4.5kHz. With some
external EQing, it is possible to make the bandwidth very flat and natural sounding.
Also, there has been a crop of new offerings from Yaesu, namely the FT-2000 and FT-5000 that can easily accomplish 4.5kHz of transmit bandwidth
with no effort. These new rigs sound very good with just a decent microphone attached.
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.
Audio component ordering is important!
First I'll recommend the ordering and then explain why.
Noise Gate (Optional)
Soft Compressor 4:1 (Optional)
Spectral Enhancer (Optional)
Aggressive Compressor 12:1
Peak Limiter (Optional)
Effects Processor (Optional)
This one is pretty straight forward. The mic must be first for obvious reasons!
The preamp placement is also quite obvious since the mic signal needs to be amplified to a line-level
signal before it can be fed into a line-level device input such as an equalizer. This is especially true of compressors that requires high
enough levels to compress correctly.
Even though an equalizer can be fed with an unamplified 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 usualy the best place to inset 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 voise energy and background
noise energy. Also, you will kill background noise upfront before it is EQed and compressed. So, if a dedicated gate is used, insert it just
after the preamp and just before the EQ.
Soft Compression (Optional)
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 (Optional)
(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.
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
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 >
A noise gate 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.
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 Recomended for Amateur Radio SSB)
Aphex Dominator 720 / Compellor 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 Processors? (Not Recomended for Amateur Radio SSB)
Symetix 528E / DBX-286 / Orban Optimod 8000/ 9000 / 9100 / 9200 / etc...
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 SSB is not specific enough. Their flexibility is
very limited in 3~6kHz bandwidths and tend to focus better with wider bandwidths and processional RF equipment. These processors are better
suited for AM in my opinion.
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 $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 above paragraphs are only suggestions
based on my experiences and not anything that should be considered as bible.
Have fun experimenting while finding your ultimate setup.
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 may vary depending on what kind of mic you are using. For Dynamic mics, you should be about 3 to 4 inches from the front and maybe
slightly angled away to prevent "Poping". The beauty of dynamic mics is the "Proximity" effect when you get real close or
far away. 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!
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.
Mic Support System
There are several types of mic support systems available. The Heil Sound "PL-2" heavy duty, fully adjustable boom arm is an excellent
choice! It comes in a black finish, has internal springs and is made to hide your mic cabling inside the boom for a nice clean look! It sells
for $65.00 and is a real bargin! To see this product, click on the following link or point your browser to:
Microphone Preamp Setup
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
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.
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 150Hz and down, as well as the high frequencies above about 2.7kHz and up. So, we
need to do basically three things:
* Reduce the Midrange
* Increase the Bass
* Increase the Treble
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.
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.)
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 )
||Fullness at 120Hz; Boominess at 200 to 240Hz; Presence at 5kHz; Sibilance at 7.5kHz; Air at 12 to 15kHz
||WHEN USED PRODUCES THIS EFFECT
||WHEN USED TOO MUCH PRODUCES THIS EFFECT
|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
||3kHz listening fatigue, lisping quality, "M", "V",
|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
fC / Bandwidth
fl = fC - (
fC / 2Q )
fH = fC + (
fC / 2Q
Q = Quality Factor
fH - fL = Bandwidth
fC = known Frequency Center
fL = Frequency Low Spread
fH = Frequency High Spread
| Q = .67
fl = fC x .254
fH = fC x 1.746
| Q = .92
fl = fC x .456
fH = fC x 1.543
| Q = 1.41
fl = fC x .645
fH = fC x 1.354
| Q = 1.90
fl = fC x .736
fH = fC x 1.263
|Q = 2.15
fl = fC x .767
fH = fC x 1.233
| Q = 2.87
fl = fC x .826
fH = fC x 1.174
| Q = 4.32
fl = fC x .884
fH = fC x 1.116
||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 )
fH = 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 )
Q = 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
BELL LABS EVEN FREQUENCY DISTRIBUTION
(Assuming a Constant -3dB/Octave Amplitude Decline)
6000.000 / available high freq = required low frequency @ -3dB
Formula 2 (Inversed)
6000.000 / 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.
NU9N MODIFIED EVEN FREQUENCY DISTRIBUTION
(Assuming a Constant -1.5dB/Octave Amplitude Decline)
3000.000 / available high freq = required low frequency @ -3dB
Formula 2 (Inversed)
3000.000 / 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.
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
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:
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:
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.
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.
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 processors can add a simulated room or hall sound giving your audio a final polish, making it
sound somewhat bigger as opposed to dry and present. I don't personally care for effects, but adjusted properly I have heard some interesting
For a nice effect, try a thin or light "Plate" or a medium room program. Again, experimentation is key. Find what you like!
I use 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: 3 to 8 (User preference)
The only thing I would point out that I believe is important, is making sure that you do not overdo the Wet to Dry mix. Keep the effects very
subtle so that they are just noticeable. A good dry to wet mix would be one that gives the appearance that you are not running any effects at
all, until you turn it off.
The output level in your effects processor will be your final level adjustment that will reestablish the original ALC level that was set up in
STEP 1 of the "Level adjustments Setup" section.
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: John@nu9n.com
The Carrier Phenomena
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!
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
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!
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 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:
||Keep the ground path from earth to station as short
||Run multiple length ground cables from earth to station.
||Make sure your tower legs are well grounded.
||Use heavy braid for your ground connections.
||Tie your station AC ground into same earth ground
that your station uses. This keeps the ground potentials at a minimum.
||Use a "Star Ground" configuration
for all ham and audio equipment. Connect every chassis of ham/audio gear (radios, amps, rotor box, tuners, audio equipment, etc...) separately
to a central grounding point in your station. All equipment ground connections will converge at this location. Do NOT ground one chassis
directly to another.
Building an Indoor Counterpoise:
(The following text is from the Radio Works catalog)
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
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.
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.
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:.
If needed, choke all of your RF and AF inputs in the system. "Kreger Components, Inc." (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
RF signals via Coax
2.4" O.D. Donuts
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.
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. "The Radio Works" makes a
very effective line isolator called the "T-4" and "T-4G" for these purposes. They can be reached at: (800) 280-8327
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.
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
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:
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.
See also this excellent RFI troublshooting reference by K9YC by clicking here...
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!