Amateur Radio has always been my favorite hobby!   I've especially enjoyed the opportunity to build many meaningful friendships with other hams. 

I became a licensed ham while in the Navy in order to make telephone patch calls from aboard ships in the Pacific during my 28-year career in the Navy.  It was fortunate to be permitted to operate my own equipment, a Kenwood TS-520S, and trap vertical antenna from five Navy ships.

What is amateur radio?

Amateur radio, also known as ham radio, is the use of radiofrequency spectrum for purposes of non-commercial exchange of messages, wireless experimentation, self-training, private recreation, radiosportcontesting, and emergency communication. The term "amateur" is used to specify "a duly authorized person interested in radioelectric practice with a purely personal aim and without pecuniary interest.

The amateur radio service (amateur service and amateur-satellite service) is established by the International Telecommunication Union (ITU) through the Radio Regulations. National governments regulate technical and operational characteristics of transmissions and issue individual station licenses with a unique identifying call sign, which must be used in all transmissions. Amateur operators must hold an amateur radio license which is obtained by passing a government test demonstrating adequate technical radio knowledge and legal knowledge of the host government's radio regulations.

Radio amateurs are limited to the use of small frequency bands, the amateur radio bands, allocated throughout the radio spectrum, but within these bands are allowed to transmit on any frequency using a variety of voice, text, image, and data communications modes. This enables communication across a city, region, country, continent, the world, or even into space. In many countries, amateur radio operators may also send, receive, or relay radio communications between computers or transceivers connected to secure virtual private networks on the Internet.

Amateur radio is officially represented and coordinated by the International Amateur Radio Union (IARU), which is organized in three regions and has as its members the national amateur radio societies which exist in most countries. According to an estimate made in 2011 by the American Radio Relay League, two million people throughout the world are regularly involved with amateur radio.[2] About 830,000 amateur radio stations are located in IARU Region 2 (the Americas) followed by IARU Region 3 (South and East Asia and the Pacific Ocean) with about 750,000 stations. A significantly smaller number, about 400,000, are located in IARU Region 1 (Europe, Middle East, CIS, Africa).

 Station COMPONENTS and equipment layout


Kenwood TS-590SG atop Kenwood TS-870S.  HP I7 PC at right. 


Acom 1010 linear amplifier atop DBX 286S microphone processor at left.   One of two Behringer near field digital monitors atop Palstar AT1500BAL balanced line antenna tuner at right.
 

ABOUT THE KENWOOD TS-590SG

 ... a great value in an HF transceiver!  Perhaps you're thinking you might opt for a 590SG yourself.  

It should be noted that this page is purposely aimed at ham's desiring to utilize their 590 for virtual audio handling as described elsewhere on this website by clicking Digital audio for the Kenwood 590SG.  

■ An even higher performance receiver with superior adjacent dynamic range.
■ Advanced AGC control through digital signal processing from the IF stage.
■ Highly reliable TX outputs high-quality TX signal.
■ New morse code decoder. Scroll display on the 13-segment display unit.
Characters are shown in a dedicated window on ARCP-590G.
■ MULTI/CH knob (with push-switch) and RIT/XIT/CL key also configurable in
addition to existing PF A and PF B programmable functions.
■ New Split function (TS-990S-style) enabling quick configuration added in addition to the existing Split setting.
■ Transceiver equalizer configurable by mode.
■ FIL A/B configurable independently with VFO A/B (convenient during Split operation).
■ Front or rear PTT selectable for Data PTT.
■ Switching from HI CUT/LO CUT to WIDTH/SHIFT possible for reception bandwidth changing in SSB mode.
■  Large display with superior visibility.  LED backlight color tone configurable in 10 steps from amber to green.
■ The speech processor is independently configurable for microphone transmission and voice message transmission.
■ 20-step expansion of settings ranges including TX monitor and CW sidetone, etc.

New 590SG owners are urged to first study the operator's manual.  Also, you may want to become a member of the 590's user group at https://groups.yahoo.com/neo/groups/KenwoodTS-590/info as well as a frequent visitor to the 590 family resources website at http://www.g3nrw.net/TS-590/ .  

Here are K$QKY's suggested TS-590SG menu configuration if handling virtual tx audio via  the USB cable:  Other default settings as per pages 15 – 19 of the instruction manual except as noted below:

Menu number Description Default Current
27 Auto mode Off On
31/33 TX filter ssb/am low cut 300 100
32/34 TX filter ssb/am high cut 2700 2900
35 Speach processor effect (see note below) hard soft
36 TX equalizer (see note below) off Off
59 HF linear amp control relay (if applicable) off 3
67 & 68 Com speed 9600 115200
 69 Audio input line for data ACC2 USB
70 Audio source of send/ptt front rear
71 USB input audio level 4 3
 76 Data VOX off on
77 VOX delay 50 15

MIC -  Level 12 when routing outboard hardware processed analog audio to the 590SG.   This control is not applicable when handling virtual audio via the USB connection. 

Menus 36 and 37 are not applicable when using outboard processed transmit audio techniques to the rig as compression and EQing are integral to most outboard processing schemes.  

Hams who prefer using the 590SG's built-in TX EQ should install Kenwood's control software as discussed at http://www.kenwood.com/i/products/info/amateur/ts_590g/arcp590g_e.html which facilitates setting up this functionality.  

Use a type AB USB cable for hookup to the 590SG.   Windows should automatically install the necessary silicon labs driver.   Check Windows Device Manager to determine which port is designated for the UART and make certain that the 590 and ARCP-590 software is set up to reflect the correct port, baud rate (preferably 115200). 

Quick and easy pan adaptor hookup!

For hams wanting to integrate a pan adapter with their 590SG, a DRV connector on the back panel is switchable to the antenna output function.  This new capability is perhaps the single most significant advancement over the original 590S!    K4QKY uses SDRplay's RSP1A  entry-level SDR receiver together with the companion software SDRuno (see screenshot below) and Omnirig for control.  More about this at https://www.sdrplay.com/.

 The delta loop antenna is a great performer!

The station's delta loop antenna is fed with a 450-ohm ladder line running down into the ham shack to a Palstar AT1500BAL balanced line antenna tuner in the shack. The above graphic depicts the overall layout of this loop cut to one full wavelength (141 ft) on the lowest operating frequency (40 meters).  The mean height of the loop above ground is about 50 ft. 

Loop suspension attachment points consist of nylon pulleys attached to dacron halyards to maintain equal tension on all three sides of the loop.  

Note:  Hams employ various techniques for placing the halyards over the tops of trees.  Many even "over-engineer" the process by adding spring tension devices, etc.  K4QKY prefers to keep it simple by the use of 6lb colored (for best visibility) stranded fishing line, a hefty sinker, and a slingshot.   Once the line has been launched over the tree, pull over a lightweight nylon cord.   Finally, attach the cord to the halyard and pull it over the tree.    Alternatively, adjust each halyard to achieve the desired tension in the loop.    

Why feed a loop with a balanced feed line?

If you doubt the viability of feeding wire antennas like loops and dipoles with open wire line, read the following explanation courtesy of K5UA "Charles":

There are two kinds of line loss, the matched line loss, and the mismatch line loss.  Matched line loss is measured at different frequencies in db per 100 feet when the line is terminated into a load that is identical to the characteristic impedance of the line. The loss increases as the frequency increases.  Mismatched line loss is an additional attenuation of the signal because of the line being terminated into a load that is different than the characteristic impedance of the line. This loss increases with frequency, but it also increases with the magnitude of the mismatch. Needless to say, lossy lines like the small coax have higher mismatched line loss numbers for the same amount of mismatch than the lower loss, large coax lines.

Matched line loss is unavoidable, but mismatched line loss is avoidable if the load can be matched to the line at the load end of the line. An example of this would be a gamma match at the yagi terminals to transform the 16-ohm impedance of the antenna to the 50-ohm characteristic impedance of the coax. Another example would be to use a 4:1 balun at the antenna terminals to bring the 16 ohms closer to 50 ohms.

The match can also occur at the transmitter end of the line, but the mismatched line loss would be there. The transmitter would be happy looking into a 50-ohm load, but the mismatched line loss would still be present because the load is not matched.

The beauty of an open-wire line is that the matched line loss is virtually zero. Even with large line/load mismatches, like 10:1 or 15:1, the additional mismatched line loss is very low. As long as the user has a conjugate match on the transmitter end of the line using an antenna matching network, virtually ALL power is radiated by the antenna. Power can not disappear, it is either radiated or lost in the line by attenuation of the dielectric material between the transmission line wires. This is why a random wire of reasonable length may actually radiate MORE power when fed
by open-wire line through a tuner, then a perfectly matched half-wave dipole fed with small coax, assuming the feed line length of each is over 100 feet.

The net effect of this is that you can put up a random length dipole (or a loop as discussed here) and use it on all bands with very little line loss and not have to worry about a bunch or resonant dipoles interfering with each other.

K5UA, in another email, goes on to say:

The probability that any single element antenna is going to have a feed point impedance of 50 +/- j0 ohms is virtually ZERO.

Likewise, when I hear of someone bragging about their quad or yagi that is 50 ohms at resonance, I always ask them how much gain and front to back did they have to sacrifice to get that 50-ohm feed point impedance. Apparently, when God was designing the universe and the laws of physics, he did not realize that the tail (50-ohm coax) was going to wag the dog (gain/front-to-back/multi-band operation) in the antenna world.

The concept of resonance also appears to baffle most amateurs because they do not know or understand the three components of impedance (resistance, inductive reactance, and capacitive reactance). A lot of amateurs believe that resonance occurs only when the antenna impedance has the same 50 ohm resistive component as the coax impedance. Actually, resonance is simply defined as the absence of the reactive component of impedance, or in other words, a purely resistive load. If the impedance of a resonant dipole is 80 ohms and you're using 50-ohm coax, the best SWR you can achieve is 1.6 to 1.

SWR really messes with the ham mind, especially with beam antennas where the SWR curve is hardly ever centered on the resonant frequency of the antenna because the feed point impedance at resonance is rarely 50 ohms. The SWR curve is, therefore, skewed to one side or the other of resonance and non-symmetrical. The worse the mismatch of the coax and the antenna at resonance, the greater the skewing effect and asymmetry of the
SWR curve.

Note:  K4QKY has typically used ladder line-fed full-wave loops at previous QTHs.  He has also experimented with a two-element Delta loop array antenna fed with 300-ohm ladder line.  More about the design and construction of this array at http://k4qky.com/files/parasitic delta loop array.pdf .  

Voice equalization:

Often misunderstood by Hams, equalizers (EQ) should only be used judiciously to achieve improved transmit audio reinforcement. As such, knowing how to properly EQ your transmit audio is one of the most critical tasks to master.   From correcting problems and enhancing your sound to adding cohesion to your voice, there’s a lot you can accomplish with proper equalization.  Excessive EQing should be avoided!   Don't ruin a great-sounding voice and microphone with heavy-handed EQing!  Concentrate on achieving great-sounding audio at the source, and you will achieve far better results. This means choosing a high-quality microphone, minimizing shack noise, and employing the proper microphone technique.  As such, understanding how to use it to your advantage can greatly enhance your sound.  Learn to "work your mic by adhering to these techniques:

1. Placement of the microphone, relative to your mouth, plays a large role in the clarity and character of your voice.  Experiment with mic placement.  A good starting point is 3 - 5 inches.

2. Avoid lateral movements to either side of the microphone. Generally, it is necessary to remain "on-axis" (in front of the microphone) to ensure a clear tone. 

3. It is preferable to remain the same distance from the microphone to ensure a consistent volume.

4. Consider proximity effect whereby base sounding tones are enhanced by "close talking" a directional microphone, the type most hams use.  Be careful doing this as it may make you more prone to "popping your Ps" when a burst of air from your mouth overloads and distorts the microphone. Popping occurs mostly on "plosives" (words that begin with "p," "b," and "t.") A windscreen or pop filter is a useful deterrent.

Follow these techniques, and you will sound better and appear more experienced. While equalization can do wonders, it’s important to consider the bigger picture every time you reach for the EQ.  An equalizer (EQ) is a filter that allows you to adjust the level of a frequency, or range of frequencies, of a human voice audio signal. In its simplest form, an EQ will let you turn the treble and bass up or down, allowing you to adjust the coloration of your transmit or receive audio. Equalization is a sophisticated art. Good equalization is something to strive for.  Parametric EQ  The parametric EQ is the most common equalizer found because it offers continuous control over all parameters. A parametric EQ offers continuous control over the audio signal’s frequency content, which is divided into several bands of frequencies (most commonly three to seven bands).  A fully parametric EQ offers control over the bandwidth (basically, the range of frequencies affected), the center frequency of the band, and the level (boost/cut) of the designated frequency band. It also offers separate control over the Q, which is the ratio of the center frequency to the bandwidth. A semi-parametric EQ provides control over most of these parameters but the Q is fixed. Q is the ratio of the center frequency to bandwidth, and if the center frequency is fixed, then bandwidth is inversely proportional to Q—meaning that as you raise the Q, you narrow the bandwidth. In fully parametric EQs, you have continuous bandwidth control and/or continuous Q control, which allows you to attenuate or boost a very narrow or wide range of frequencies. A narrow bandwidth (higher Q) has obvious benefits for removing unpleasant tones. Let’s say you have a particularly annoying nasal quality to your audio.  With a very narrow bandwidth, you can isolate this one frequency (usually around 650) and remove, or reject, it. This type of narrowband-reject filter is also known as a notch filter. By notching out the offending frequency, you can remove the problem without removing the instrument from the mix. Narrow bandwidth is also useful in boosting pleasant tones as well. A broad bandwidth accentuates or attenuates a larger band of frequencies. The broad and narrow bandwidths (high and low Q) are usually used in conjunction with one another to achieve the desired effect. A shelving EQ attenuates or boosts frequencies above or below a specified cutoff point. Shelving equalizers come in two different varieties: high-pass and low-pass. Low-pass shelving filters pass all frequencies below the specified cutoff frequency while attenuating all the frequencies above it. A high-pass filter does the opposite: passing all frequencies above the specified cut-off frequency while attenuating everything below. 

Note:  Once popular graphic EQs use sliders to adjust the amplitude for each frequency band.  K4QKY does not recommend their use in audio processing. 

One of the easiest ways you can clear up your mix and reclaim a large amount of wasted headroom is by applying a high pass (low-cut) filter since extremely low voice frequencies do not contribute to effective, clean and pleasant sounding transmit audio. 
Cut First, Boost Second
Before you boost what you want to hear, cut out what you don’t want to hear. The best reason for doing this is to remove problem frequencies from your particular voice profile. Once you achieve this goal, you’ll find you often don’t need to boost much else.  Many hams will resist doing this for the commonly held belief that "more is better".  In short, boost only as necessary and always with care.   As you adjust the EQ, you’ll notice that frequency boosts are significantly easier to hear than cuts. This phenomenon causes many hams to boost frequencies they want to bring out, rather than to cut problem frequencies. There are two major issues associated with doing this. First, if you boost at 3kHz to achieve greater presence, your audio will likely become harsh and cutting. The other problem is that if you boost all of the frequencies around a problem frequency rather than simply cutting the problem frequency (like boosting the extreme lows, upper midrange, and high end instead of just cutting the lower-mid which is really the issue), you can easily overload the EQ gain stage and introduce distortion that you may not initially notice.
Understand the frequency spectrum of your voice

Your vocal tone needs to be as perfect as possible. That’s because as humans, we can’t help but scrutinize what we hear in an extremely critical manner. Trouble is that hams often disagree with what constitutes ideal-sounding transmit audio.  So, develop your own style as you see fit but try to avoid unpolished or harsh sounding audio that will likely annoy and distract your listeners.  Here are a few suggestions for properly EQing your voice:

  • Body (200–500Hz)

    This frequency range is where muddiness lives, but it’s also where the warmth of your voice comes from. If your vocals sound mushy, try cutting low frequencies in this range. If your vocals are clear but lack warmth, try boosting in this range.

  • Nasal (1-1.5kHz)

    Almost universally, 1-3kHz is where the nasal frequencies lie. Try cutting somewhere within this frequency range. Don’t go overboard though.

  • Presence (1.5 to 3kHz)

    When it comes to intelligibility, presence is absolutely critical but be careful boosting too much as this can render your vocals harsh and jarring.

EQ with Your Ears

It is important to point out that the best tools you have for EQing are your ears. You can memorize tables of important frequencies for all kinds of instruments and applications, but the most important thing is that your voice sounds great. The best way to evaluate your sound is to listen to yourself from your rig's built-in monitor or, better yet, on air from a separate receiver, perhaps an SDR.

The bottom line for EQing your voice is to find the biggest offender and fix that first. Then, boost sparingly to polish the results.  If this technique fails, then consider reevaluating the quality of your microphone and the correctness of audio level settings in the audio chain.   Remember that an EQ can't fix poor unprocessed input.   So before you resort to the EQ, listen closely to that input to avoid falling victim to the "Garbage in... garbage out" syndrome.

Dynamic range is the ratio between the loudest possible audio level and the lowest possible level.  For example, if an audio processor states that the maximum input level before distortion is +24 dBu, and the output noise floor is -92 dBu, then the processor has a total dynamic range of 24 + 92 = 116 dB.  The average dynamic range of an orchestral performance can range from -50 dBu to +10 dBu, on average. This equates to a 60 dB dynamic range.   Although 60 dB may not appear to be a large dynamic range, you’ll discover that +10 dBu is 1,000 times louder than -50 dBu!

Equalizers can't fix the character of your voice transmission but they can help emphasize the good stuff and help minimize the bad stuff... but only if it’s of the highest quality in the first place.

Compression

Do we need compression for voice processing?  

The average dynamic range of an uncompressed vocal is around 40 dB.   In other words, a vocal can go from -30 dBu to +10 dBu.  The passages that are +10 dBu and higher will be heard over prevailing noise.  However, the passages that are at -30 dBu and below will never be heard over the roar of the noise. A compressor can be beneficial in this situation to reduce (compress) the dynamic range of the vocal to around 10 dB. The vocal can now be placed at around +5 dBu.  At this level, the dynamic range of the vocal is from 0 dBu to +10 dBu. The lower level phrases will now be well above the lower level of noise, and louder phrases will not overpower the noise, allowing the vocal to “sit above the noise.”

Can we have too much compression? 

Over-compression often sounds horrible.  That statement can be qualified by defining over compression. The term itself is derived from the fact that you can hear the compressor working. Therefore, the over-compressed sound is likely to have been caused an improper adjustment of the compressor.  Most importantly, a well-designed and properly adjusted compressor should never be audible to other hams!  

So, what constitutes compression/limiting?

Punch, apparent loudness, presence—these are just three of the many terms used to describe the effects of compression/limiting.  Compression and limiting are forms of dynamic-range (gain) control. Audio signals have very wide peak-to-average signal-level ratios (sometimes called dynamic range, which is the difference between the loudest level and the softest level). The peak signal can cause overload in the audio-recording or sound-reinforcement chain, resulting in signal distortion.

A compressor/limiter is a type of amplifier in which gain is dependent on the signal level passing through it. You can set the maximum level a compressor/limiter allows to pass through, thereby causing automatic gain reduction above some predetermined signal level, or threshold. Compression refers, basically, to the ability to reduce, by a fixed ratio, the amount by which a signal’s output level can increase relative to the input level. It is useful for lowering the dynamic range of a vocal, making it easier to be heard over the air without distortion.  

Take, for example, a ham that moves around in front of the microphone during a QSO, making the output level vary up and down unnaturally. A compressor can be applied to the signal to help correct this phenomenon by reducing the louder passages enough to be compatible with the overall signal.

How severely the compressor reduces the signal is determined by the compression ratio and compression threshold. A ratio of 2:1 or less is considered mild compression, reducing the output by a factor of two for signals that exceed the compression threshold. Ratios above 10:1 are considered hard limiting.

As the compression threshold is lowered, more of the input signal is compressed (assuming a nominal input-signal level). Care must be taken not to over compress a signal, as too much compression destroys the acoustic dynamic response.

Limiting refers to the processing that prevents the signal from getting any louder (that is, it prevents an increase in the signal’s amplitude) at the output.

Vocals usually have a wide dynamic range. Transients (normally the loudest portions of the signal) can be far outside the average level of the vocal signal. Because the level can change continuously and dramatically, it is extremely difficult to ride the level with a console fader. A compressor/limiter automatically controls gain without altering the subtleties of the transmission.

Compressor terminology

Threshold. The compressor threshold sets the level at which compression begins. When the signal is above the threshold setting, it becomes eligible for compression. Basically, as you turn the threshold knob counterclockwise, more of the input signal becomes compressed (assuming you have a ratio setting greater than 1:1).

Ratio. The ratio is the relationship between the output level and the input level. In other words, the ratio sets the compression slope. For example, if you have the ratio set to 2:1, any signal levels above the threshold setting will be compressed such that for every 1 dB of level increase into the compressor, the output will only increase 0.5 dB. As you increase the ratio, the compressor gradually becomes a limiter.

Limiter. A limiter is a compressor that is set to prevent any increase in the level of a signal above the threshold. For example, if you have the threshold knob set at 0 dB, and the ratio turned fully clockwise, the compressor becomes a limiter at 0 dB, so that the output signal cannot exceed 0 dB regardless of the level of the input signal.

Attack. Attack sets the speed at which the compressor acts on the input signal. A slow attack time allows the beginning envelope of a signal (commonly referred to as the initial transient) to pass through the compressor unprocessed, whereas a fast attack time immediately subjects the signal to the ratio and threshold settings of the compressor.

Release. Release sets the length of time the compressor takes to return the gain reduction back to zero (no gain reduction) after the signal level drops below the compression threshold. Very short release times can produce a very choppy or “jittery” sound, especially in low-frequency instruments such as a bass guitar. Very long release times can result in an over-compressed sound; this is sometimes referred to as “squashing” the sound. All ranges of release can be useful at different times, however, and you should experiment to become familiar with the different sonic possibilities.

Hard/Soft Knee. With hard-knee compression, the gain reduction applied to the signal occurs as soon as the signal exceeds the level set by the threshold. With soft-knee compression, the onset of gain reduction occurs gradually after the signal has exceeded the threshold, producing a more musical response (to some folks).

Auto. Places a compressor in automatic attack and release mode. The attack and release knobs become inoperative and a preprogrammed attack and release curve are used.

Makeup Gain. When compressing a signal, gain reduction usually results in an overall reduction of the level. The gain control allows you to restore the loss in level due to compression (like readjusting the volume).

Noise Gate

Do we also need a noise gate?

Problems sometimes arise when shack background noise (air conditioner, linear amp fan, etc.) becomes more audible after the lower end of the dynamic range is raised.  This calls for the use of a noise gate. The noise-gate threshold could be set at the bottom of the dynamic range of the vocal, say -10 dBu, such that the gate would shut out the unwanted signals between the phrases.

What is noise gating?

Noise gating is the process of removing unwanted sounds from a signal by attenuating all signals below a set threshold. As described, the gate works independently of the audio signal after being “triggered” by the signal crossing the gate threshold. The gate will remain open as long as the signal is above the threshold. How fast the gate opens to let the “good” signal through is determined by the attack time. How long the gate stays open after the signal has gone below the threshold is determined by the hold time. How fast the gate closes is determined by the release. How much the gate attenuates the unwanted signal while closed is determined by the range.

Noise gates were originally designed to help eliminate extraneous noise and unwanted artifacts from a recording, such as a hiss, rumble, or transients from other instruments in the room. Since hiss and noise are not as loud as the instrument being recorded, a properly set gate will only allow the intended sound to pass through; the volume of everything else is lowered. Not only will this strip away unwanted artifacts like hiss, but it will also add definition and clarity to the desired sound. This is a very popular application for noise gates, especially with percussion instruments, as it will add punch or “tighten” the percussive sound and make it more pronounced.

Noise gate terminology:

Threshold. The gate threshold sets the level at which the gate opens. Essentially, all signals above the threshold setting are passed through unaffected, whereas signals below the threshold setting are reduced in level by the amount set by the range control. If the threshold is set fully counterclockwise, the gate is turned off (always open), allowing all signals to pass through unaffected.

Attack. The gate attack time sets the rate at which the gate opens. A fast attack rate is crucial for percussive instruments, whereas signals such as vocals and bass guitar require a slower attack. Too fast of an attack can, on these slow-rising signals, cause an artifact in the signal, which is heard as a click. All gates have the ability to click when opening but a properly set gate will never click.

Hold. Hold time is used to keep the gate open for a fixed period after the signal drops below the gate threshold. This can be very useful for effects such as gated snare, where the gate remains open after the snare hit for the duration of the hold time, then abruptly closes.

Release. The gate-release time determines the rate at which the gate closes. Release times should typically be set so that the natural decay of the instrument or vocal being gated is not affected. Shorter release times help to clean up the noise in a signal but may cause “chattering” in percussive instruments. Longer release times usually eliminate “chattering” and should be set by listening carefully for the most natural release of the signal.

Range. The gate range is the amount of gain reduction that the gate produces. Therefore, if the range is set at 0 dB, there will be no change in the signal as it crosses the threshold. If the range is set to -60 dB, the signal will be gated (reduced) by 60 dB, etc.

 ...This page provides a setup guide for a software-based digital audio workstation (DAW) as an effective alternative to conventional hardware-based sound processing techniques. 

Notes:

(1)   A digital audio workstation (DAW) is an electronic device or computer software application typically used for recording, editing and producing audio files such as songs, musical pieces, human speech or processing live sound such as Amateur Radio transmissions.  Regardless of configuration, modern DAWs use effects processors to tailor audio.  For Amateur Radio, DAWs are primarily used for real-time transmit audio processing purposes.

(2) The scheme presented in this article describes a methodology for using a USB microphone together with a software-based processing system for onward routing of process transmit audio via a USB cable to a Kenwood TS-590SG  (or TS-590S). 

(3)  Alternatively, Appendix A provides a guide for setting up up a conventional XLR microphone together with an outboard USB audio interface instead of a USB microphone.

(4) Perhaps you prefer a somewhat simpler approach consisting of a  USB microphone without any software processing.  If so, follow the guidelines presented in Appendix B.

(5) Some hams will prefer a purely analog (hardware-based) approach to audio processing.  Appendix C describes a simple, inexpensive and highly effective scheme that can be implemented for about $200.

 

 How the scheme diagramed at left works

  1. Digital audio output from the Rode NT-USB microphone is routed to the computer via a type A/B USB cable.  Note:  Type A/B USB cable is a standard-issue USB 2.0 cable. TS-590 is the most common A to B Male/Male type peripheral cable, the kind that's usually used for printers.
  2. Digital audio is then processed within the computer by the software-based DAW system "Presonus Studio One 3". 
  3. Digital transmit audio output from  Studio One 4 is then routed via Type A/B USB cable to the Kenwood TS-590.  Finally, it is converted from digital to analog and transmitted.
  4. 55 watts RF output from TS-590SG transceiver.
  5. 700 watts RF output from the ACOM 1010 linear.
  6. 700 watts RF output from Palstar AT1500BAL to a horizontal loop antenna.
  7. Received audio from the Kenwood 590SG is transported via USB cable from the 590SG back to the computer and outputted to the speakers.  Alternatively, receive audio can be routed from the 590SG's rear panel speaker output jack direct to a pair of near-field monitor speakers.  

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Why do it this way?

There are several advantages to using a USB microphone together with real-time software-based transmit audio processing software including:
  
  • Less expensive compared to conventionally audio processing hardware.
  • Several higher quality of USB microphone choices have recently become available for purchase over the internet as discussed at https://ehomerecordingstudio.com/usb-microphones/.
  • Potentially less susceptibility to RFI and hum than a conventional microphone.Ability to process transmit audio with software rather than more expensive external processing hardware.   
  • Facilitates remote operations via the Internet. 

Note: K4QKY enjoys using the Rode NT USB microphone.  More about this highly capable microphone at http://www.rodemic.com/nt-usb .  There are many excellent USB microphones to consider including those described at http://www.soundonsound.com/sos/feb07/articles/usbmics.htm. Alternatively,  as discussed in Appendix A of this article, a convention XLR condenser, or dynamic microphone such as the Heil model PR35 is used together with a Behringer U-Phoria UMC202 USB audio interface as described at http://adamivy.com/behringer-umc202hd-review/.

K4QKY's setup includes:

 Hardware: 
  • Kenwood TS-590SG
  • PC running Window’s 10 with internal Realtek sound card
  • Rode NT USB microphone which is a highly versatile side-address microphone that is ideal for recording musical performances in addition to spoken applications such as podcasting, voice-overs, and Amateur Radio. The body of the NT-USB features a zero-latency stereo headphone monitoring (3.5mm) jack which allows users to monitor the microphone input in real-time, along with dials to adjust the monitoring level and mix between the computer audio and the microphone input.  
  • Type A/B USB cable connected to the microphone and computer. Another type A/B cable connects the computer to the TS-590SG.

Software:  (just one example of many available options)

The following software needs to be downloaded and installed: 
  • “ASIO4ALL” which can be downloaded free from http://www.asio4all.com/ . Audio Stream Input/Output (ASIO) is a computer sound card driver protocol for digital audio providing a low-latency and high fidelity interface between a software application and your computer's sound card.  ASIO is a computer sound card driver protocol for digital audio specified by Steinberg, providing a low-latency and high fidelity interface between a software application and a computer's sound card
  • PreSonus Studio One® 4 Artist, a software-based audio processing digital audio workstation (DAW) that contains everything you’d expect from a modern digital audio workstation with a fast, flow-oriented, drag-and-drop interface.  You can purchase and download it from https://shop.presonus.com/products/new-noteworthy/Studio-One-4-Artist. It has the advantage of being a complete stand-alone system with its proprietary plugins.  Better yet, purchase a PreSonus AudioBox USB 2x2 audio interface from the webpage at https://www.amazon.com/PreSonus-AudioBox-USB-Audio-Interface/dp/B00154KSA2 and receive the software as a free add-on complete package! This is surely a "no-brainer" package deal!  You can then opt to use this interface instead of the Behringer interface described in  Appendix A.

       Note:   There are several other alternative software-based processing systems beyond the scope of this article that you can experiment using including the use of VST plug-ins, many of which are free.    

Kenwood 590SG settings: (Menu numbers differ in the 590S)
Menu number Description Default Current
27 Auto mode Off On
31/33 TX filter SSB/am low cut 300 100
32/34 TX filter ssb/am high cut 2700 2900
35 Speach processor effect (see note below) hard soft
36 TX equalizer (see note below) off Off
59 HF linear amp control relay (if applicable) off 3
67 & 68 Com speed 9600 115200
 69 Audio input line for data ACC2 USB
70 Audio source of send/ptt front rear
71 USB input audio level 4 3
72 USB output level 4 3
 76 Data VOX off on
77 VOX delay 50 15

MIC -  0 as the front panel connected conventional microphone is not normally used with this scheme.

Menus 36 and 37 are not applicable when using outboard processed transmit audio techniques to the rig as compression and EQing are integral to most outboard processing schemes.  

Hams who prefer using the 590SG's built-in TX EQ should install Kenwood's control software as discussed at http://www.kenwood.com/i/products/info/amateur/ts_590g/arcp590g_e.html which facilitates setting up this functionality.  

Use a type AB USB cable for hookup to the 590SG.   Windows should automatically install the necessary silicon labs driver.   Check Windows Device Manager to determine which port is designated for the UART and make certain that the 590 and ARCP-590 software is set up to reflect the correct port, baud rate (preferably 115200). 

 

Windows sound manager settings: 

"Playback" tab  
> Speakers High Definition Audio Device set as Default Device.  Properties set at level 82, no enhancements, 16 bit, 48000 Hz and no exclusive mode.
> Speakers-USB Audio CODEC set as Default Communications Device.  Properties set at level 56, no enhancements, 16 bit, 48000 Hz and no exclusive mode.   Adjust level as necessary to adjust transmit audio input to the 590 as necessary to achieve a slight ALC meter deflection on voice peaks.
  
"Recording" tab  
> Microphone Rode NT-USB set as Default Device.  Properties set at level 24, 16 bit, 48000 Hz and no exclusive mode.  Address the microphone at about 3 inches to alleviate pickup of shack noise.
> Microphone-USB Audio Codec.  Properties set at level 70, 16 bit, 48000 Hz and no exclusive mode.   Note:  Hams who prefer to route receive audio output from the 590sg to their desktop speakers should select the "Listen" tab and the checkbox opposite "Listen to this device". Then, enter your “USB audio CODEC”  playback device from the drop-down box. Select the “Levels” tab and set the slider to 70 as a starting point to adjust the audio output from your speakers.   Adjust as necessary.  Alternatively, Hams who prefer to route receiver audio via a 3.5 mm analog cable connected between the 590G's back panel speaker out jack directly to their monitor speakers can disable this device. 

Software setup:

Step 1 - ASIO device driver software setup:

“ASIO4ALL” should first be downloaded free from http://www.asio4all.com/ . Once installed you should search for and open the Windows "ASIO4all offline settings" to set up appropriate inputs and outputs as per the below screenshot.   It should be noted that the computer's built-in Realtek soundcard is not selected as it is essentially only used for non-ham radio applications.  

 

Step 2 - Purchase and download Studio One 4 as per the of instructions.  Once installed and started,  click "configure Audio Device" from the lower center portion of the opening splash screen which will reveal a "Options" window similar to the one shown below.   

Once you have completed entering data to conform to what is shown in the above screenshot, click the "OK" button in the lower right-hand corner of the window which will bring you back to the opening splash screen.

From the splash screen click "Create a new Song" which will reveal a window similar to the one shown below.

Enter selections like shown in the above screenshot (except set the Sample rate to 48KHz) and click the "OK" button which should reveal your new song project workspace similar to the screen shown below.

Right-click and select "remove" to delete any Inserts shown in the lower portion of the above screen.  Then select the "Effects" tab in the upper right corner of the screen.  From the drop-down select the "Fat Channel" effect and drag it to the open space under " Inserts" which will then reveal a setup GUI similar to the one shown below.

Notice the presence of a collapsible dropdown list under the Fat Channel where you can select desired presets as a starting point in perfecting your effects.  It is suggested that you initially select Male 1 as shown above.  Once you do this, the GUI will then reveal the details of this preset.  

You can individually double-click each of the four effects listed, i.e., gate, compression, equalizer, and limiter.   Unlike shown in the above screenshot, suggest you change the order of effects to make the eq precede the compressor.  You may also want to click the thumbtack icon on the upper bar of each effect to keep the effect on top in a free-floating manner.  Each effect is equipped with numerous presets for you to experiment with and modify to suit your audio preferences. Click https://www.google.com/search?q=fat+channel+xt&rlz=1C1EKKP_enUS753US753&source=lnms&tbm=vid&sa=X&ved=0ahUKEwi29ZGvgf3XAhVH54MKHcaPBkwQ_AUICygC&biw=1536&bih=759 to watch videos which is the easiest way to learn more about the functionality of this wonderful multi-purpose "Fat Channel" effects processor".  

Now comes the all-important task of making final adjustments to your entire system.   Do this by routing your transmit RF to a dummy load, listening to your transceiver's monitor with headphones and speaking into the microphone.  If necessary, adjust input and output gain levels on the above window together with the audio level of the “Speakers” device in the Windows sound manager playback tab together with the  “Microphone” audio level in the recording tab and the rig’s audio input gain control as necessary to achieve just a slight deflection of the ALC meter on voice peaks while all the while listening to yourself from the rig’s audio monitor.  This is a  necessary “balancing act” to find your “Sweet spot.”  Take your time and get it right!  

Once you have arrived at a preferred scheme, be sure to save your setup by selecting File/Save as.  

Conclusions:

USB microphone operations via a USB cable connected between a computer and the 590SG transports transmit audio exceptionally well.  Moreover, this can be accomplished without the necessity for using any Kenwood-specific software and without the usual latency shortfall. 

The Kenwood 590SG's simple to use USB cable connectivity is one of it's "killer" features!  Try it for yourself and Email This email address is being protected from spambots. You need JavaScript enabled to view it. with your questions, thoughts, and recommendations. It is anticipated that more hams will come to embrace the use of a USB microphone on the phone bands once they learn more about the inherent advantages of this method of operating.

Alternative processing software: 

This article has described a method of transmitting audio processing with the stand-alone DAW, Studio One 4.  There are many other alternatives that K4QKY continues to experiment with including "Breakaway One" audio processor as described at http://www.breakawayone.com/breakaway-one/.  You can download and install the fully functional demo version from this site.    Once installed and set up (consult the included quickstart guide), the graphical user interface (GUI) will look similar to the below screenshot. 

Note:  You might also want This email address is being protected from spambots. You need JavaScript enabled to view it.to experiment with the use of VST plug-ins, many of which are free.   Ham's wishing to pursue this approach should click https://bedroomproducersblog.com/2011/05/16/bpb-freeware-studio-best-free-vst-host-applications/ for a list of potential vst plug-in hosts.  Pay particular attention to "Reaper" which K4QKY considers to be the best of the list.

Comments and recommendations regarding this article and other information presented on this website are indeed welcome... Just call K4QKY at 601-658-2808 or email him at This email address is being protected from spambots. You need JavaScript enabled to view it..


Appendix A

XLR microphone together with a USB audio interface instead of a USB microphone

Hams often already own one or more conventional microphones and may understandably be reluctant to spend the extra dollars on a USB microphone.  Assuming that they desire to experiment with the digital (or analog) processing techniques, they may want to consider integrating their microphone with a USB audio interface or, as discussed in Appendix C below, a hardware-based analog processor.

Oher pros and cons 

  • USB mics are less complicated to set up. 
  • XLR mics require an audio interface.
  • XLR mics are typically more sophisticated in design.
  • USB mics are preferable for portable radio operations with a laptop.

Choosing an audio interface

The webpage at https://www.sweetwater.com/insync/audio-interface-buying-guide/#io provides a useful buying guide for hams who elect to purchase an audio interface to use with their favorite conventional XLR microphone.  

K4QKY's audio interface of choice

 

That's a Behringer U-Phoria UMC202HD USB audio interface sitting atop K4QKY's 590SG.  This one is hard to beat from a value perspective.  Moreover, unlike some other interfaces, Behringer's companion software driver is rock-solid dependable. More about that, later on, is this Appendix.  You can learn more about it by watching the video at

 
 or from another vendor of your choice. 

Setup 

Nothing could be more simple!

  1. Download and install the driver for the interface from https://www.musictri.be/Categories/Behringer/Computer-Audio/Audio-Interfaces/UMC202HD/p/P0BJZ/downloads.
  2. Run an XLR cable between mic and front panel of the interface.
  3. Hookup a Type A/B USB cable between your PC and the rear panel of the interface.
  4. Restart your PC and enter your PC's windows sound manager.  Setup the recording and playback devices to conform to the instructions below.

Recording tab - Note that the interface is set up as the default device.  Levels at 52, no Enhancements and Advanced tab at 2 channel, 24bit, 19,2000 Hz.

  

Playback tab - Note that this tab is mainly set up much like the one previously discussed above when using a USB microphone.  The Behringer speakers are not set as the default device in order to continue to listen to Internet-delivered audio and other entertainment through the ham shacks default near field monitor speakers.  This scheme runs contrary to the Behringer's documentation which calls for the audio interface to drive your speakers rather than your PC's Realtek sound card.  Moreover, to make this scheme work, you will need to rely on your ASIO4ALL driver instead of the Behringer driver.   Not that you shouldn't use the Behringer scheme, K4QKY prefers this scheme as it is far more straightforward to implement than the Behringer scheme.  Moreover, keeping ham radio audio separate from PC audio simply is the better way to do things.


Now, restart your PC and open your ASIO4All offline settings app and make it conform to the screenshot below.

Restart your PC once again and proceed to make a test run together with your overall processing scheme. 

That's about all there is to implementing this scheme other than ensuring all input and output levels are correctly adjusted between elements of your audio path to exhibit just a slight deflection of your 590SG's ALC meter on voice peaks.  The scheme presented in this appendix is equally applicable to processing audio input from a USB microphone. 

  

 

Congratulations, you've got your new Amateur Radio license and can't wait to start operating on HF phone!    Perhaps you're not certain how you want to conduct yourself.    After all, there very few mandated rules as such.   Most hams have developed good operating practices and etiquette simply by listening to more experienced hams and you will as well.  Here are some of my preferences for your consideration.  Have fun!

Opinions from Don "K4QKY"

 

  

    

Strive to

► Exercise politeness regardless of the circumstances.  Whenever you begin to feel that you can't, then stop transmitting.

► Be a good example especially for newly licensed Hams as well as short wave listeners who may be thinking about becoming a ham.

► Be a good listener.  It will help you better organize your thoughts before transmitting.

► Reply to a CQ, or call CQ yourself.  It is and always should be a major ingredient in the magic of ham radio.

► Speak clearly and slowly, especially when giving your call sign to someone you have never worked before.

► Promote friendship and goodwill to DX contacts.  Look for ways to get to know each other rather than simply exchanging signal reports and 73s!

► Try to keep track of everyone in the QSO.  Hopefully, someone has assumed the role of "traffic director" to make sure everyone has a chance to contribute to the discussion.  If not, don't hesitate to do it yourself.

► Make it clear at the end of each transmission which station is expected to transmit next.  Try to do this even when operating VOX.

► Operate on frequencies that are in whole Khz (e.g. 18.130Khz). This alleviates ambiguity and makes it easier for everyone to be on the same frequency.

► Openly praise other hams when you observe them doing something that you feel is especially deserving. e.g., helping demo ham radio to a group of scouts.

► Always be ready to quickly and calmly respond to emergency situations.  Rehearse what you would do if presented with various scenarios.

► Operate in keeping with good amateur practice.  Be certain to always comply with the provisions of Part 97 of the rules.

► Pause between transmissions.  "Quick keying" gives the appearance that other hams are unwelcome in your QSO.

► Consider using the Internet to enrich your QSO.  Avail yourself of the opportunity to add additional information and photos to your qrz.com page.  You may also want to consider developing your own comprehensive personal website.

► Respect the privileges of hams operating in other modes on the HF bands including those who enjoy AM.

► Make a point to try 17 and 60 meters.  Good practices are especially prevalent.

► Look for opportunities to "Elmer" newly licensed hams when you hear them on the HF bands.   Welcome them, solicit their questions and give them pointers on good operating practices.

► Remember that no one country can proclaim to be the leader of the Amateur Radio world.   Likewise, no one country's foreign policy is any more right or wrong than that of another country.  Let's always keep this in mind when operating on the ham bands.

► Talking politics is generally considered to be in poor taste on the ham bands.  However, if you are willing to courteously listen to the opinions of others while not insisting that you're right and their wrong, then you may want to do it.   Just be careful to not offend others, especially if they are in another country. 

► Develop good operating practices.  You will be doing your part in helping insure the continuance of our long and proud tradition of self-regulation.  Moreover, you just might convince someone else to also become a ham.

 

Avoid

► Acting like some sort of broadcast radio station.  Your fellow "Amateurs" will most likely not appreciate such a blatant display of personal ego.

► Acknowledge the presence of deliberate interference.  After all, that's their overall objective.

► Being excessively long-winded especially when in a round-table discussion. (KG9OM often fails to head his own advice)

► Just talking about ham radio.  Many hams often have other interests.  

► Operating when you are in a bad mood.   You will be that much more vulnerable to losing your temper.

► Overusing Q-codes and other ham radio jargon on the phone bands.

► Claiming any particular frequency for nets, schedules, etc.  If your designated frequency is already in use, simply move up or down as necessary.

► Transmitting before first determining if the frequency is clear.  This includes transmitting within 3Khz of other known QSOs.

► Breaking into an ongoing QSO unless you can hear the majority of the participants.

► Ignoring someone new to a round table QSO.  Do your part to make everyone feel welcome. It is inappropriate to say anything that makes the discussion appear exclusive to a particular circle of friends.

► Testing your radio on the air.  It is far better to use a dummy load.

► Coughing, sneezing, or clearing your throat into your microphone.

► Operating VOX when it may potentially tend to foster "quick keying" as it may tend to give the appearance that you don't welcome breakers.

► Becoming a "Band Policeman" quick to tell others what you feel they are doing wrong.  In instances where it may be called for, always be polite and constructive.

► Turning up your microphone gain or resort to excessive speech processing in order to be heard.  Such practices will most likely result in diminished audio quality and an increased likelihood of interference (IMD) to nearby QSOs.

► Using the word "break" when wanting to join an ongoing QSO.   Simply give your call sign between transmissions and reserve use of the word "break" for more urgent situations.

► Joining an ongoing QSO unless you have something to contribute to the discussion.  Try not to interrupt  other hams with a  request  for audio checks,  signal reports, etc.

► Knowingly interfere with an ongoing QSO just because you are working DX especially split frequency.   Working DX is great fun but should never assume to assume to include special operating privileges.

► Saying that the frequency is not in use if you hear someone ask if the frequency you are listening to is in use.  You should only respond if you know that the frequency or one nearby is in use.

► Ridiculing other hams or express any negative views of the overall state of Amateur Radio.  If you don't have something positive and constructive to say, don't say anything at all.

► Hiding behind your microphone.  Don't say anthing over the air to someone that you wouldn't say face to face.  

      

Notes: 

(1) The foregoing list represents the opinion of one ham.   As such, it is intended to be nothing more than a “shopping list” of suggested guidelines presented almost entirely from the perspective of a “rag chewer”.   DXers, contesters, and hams who enjoy other modes will most likely have somewhat different views.    Equipment-related issues are generally well known, are purposely not included here.     In any event, it is hoped that this list may prove somewhat useful especially for new operators.   The overriding theme is common sense and courtesy to others.  Let’s always remember what a privilege it is to operate on the ham bands!  This will help avoid doing anything that might impinge on the enjoyment of our hobby for others.  

(2) Equipment-related practices, although beyond the scope of this article, are nonetheless important.  Always be aware of your signal quality especially through careful adjustment of microphone gain and speech processor/EQ settings.  Let's all do our part to avoid generating interference.  

(3) Thoughts and recommendations are welcome.  Email K4QKY at This email address is being protected from spambots. You need JavaScript enabled to view it..  

(4) You or the organization you represent are welcome to create a link to this page.

(5) Links to other related resources:

http://www.ham-operating-ethics.org/files/1-Eth-operating-EN-IARU-R1-V3-CORR-2011.pdf -  "Ethics and operating procedures for the radio Amateur" by ON4UN

http://www.hamuniverse.com/proceduers.html - "Good operating practices and procedures for the ham bands" by AJ4D

http://www.dxzone.com/catalog/Operating_Aids/Beginner_s_Guides/ - "Guides to amateur radio for beginners" courtesy of The DXZone

http://www.rsgb.org/tutors/pdf/good_operating_practices.pdf - "Good amateur radio practices" courtesy of The Radio Society of Great Britain

http://www.w9uvi.org/?page_id=68 - "HF Operating Practices" by W9UVI

Other K4QKY amateur related publications:

Vintage Radio restoration

Restoration project

Some old photos

Two element Delta loop antenna array

 

Letter from a silent key (deceased):


This letter was received on the occasion of Don's retirement from a 28 year Navy career.  What wonderful support that Frank and others provided me then, afterward and still today.   Indeed, this is what our hobby is all about!
Don and Pat Snodgrass,  1331 Primrose Lane, Montgomery, AL  36111
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