DIGITAL TRANSMIT AUDIO PROCESSING

Voice equalization:

Your vocal tone needs to be as natural sounding and articulate as possible. Avoid relying solely on what other hams may think of your audio as they will often disagree with what constitutes ideal-sounding transmit audio. So, develop your own style while avoiding unpolished or harsh-sounding audio that will likely annoy and distract your listeners. Always be mindful to not interfere with hams on adjacent frequencies.

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 transmitted 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 your otherwise 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. The 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 the 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 clean up your transmit audio 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. Likewise, apply a low pass (high cut) filter since extremely high frequencies typically fall beyond your transmit bandpass.

Ununderstand the frequency spectrum of your voice

Low frequency (200–500Hz)

This frequency range is where muddiness often lives, but it’s also where the warmth of your voice originates. If your audio sounds mushy, try cutting low frequencies in this range. If your vocals are clear but lack warmth, try minimal boosting in this range.
Midrange (1-1.5kHz)

Almost universally, 1-3kHz is where nasal and other troublesome frequencies lie. Try cutting somewhere within this frequency range. More about the importance of surgical cutting in the midrange is discussed later on this site.
High frequency (1.5 to 3kHz)

Articulation resides here but be careful boosting too much as this can render your vocals harsh and jarring.

Cut First, Boost Second

Cut problem areas before resorting to boosting any portion of your audio. The best reason for doing this is to remove problem frequencies from your particular voice profile. Once you achieve this goal, you may likely determine that little if any boosting is necessary. Many hams mistakenly overly boost their audio for the commonly held belief that "more is better" especially when trying to break a pileup. These hams tend to boost frequencies they want to bring out, rather than to cut problem frequencies. All too often hams overdue boosting the high and low ends to achieve greater presence. Instead, their audio will likely become harsh together with less articulation. As such, boost only as necessary and always with care.

EQ with Your Ears

It is important to point out that the best tools you have for EQing are your ears. The best way to evaluate your sound is to listen to yourself with high-quality headphones from your rig's built-in monitor.

Compression

Do we need compression for voice processing?

Yes, but sparingly. Compression can be beneficial to reduce (compress) the dynamic range of your voice. At a proper adjustment, lower-level word phrases will rise toward already louder phrases improving the ability of your transmit audio to “sit higher above the existing band noise” with greater presence. Avoid over-compression. A properly adjusted compressor should never be obvious to other hams!

Noise Gate

Do we also need a noise gate?

Yes, unless you are fortunate enough to have a quiet shack. 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 spoken phrases. Be mindful to minimize the sensitivity of your microphone in picking up unwanted shack noise. K4QKY always uses microphones with a cardioid pickup pattern, correct microphone output setting, and speaks about 3 inches into his microphone.

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.

What else is necessary?

Nothing really, until such time you're ready to experiment with more advanced processing techniques.

...The remainder of this article 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) K4QKY recently replaced his Rhode NT-USB condenser microphone with a newer Diety VO-7U dynamic USB microphone originally designed for streamers and podcasters. It features a tight pickup pattern that allows it to reject room noise so other hams are less apt to hear shack noise. The VO-7U works natively on all major operating systems, with no drivers or software required.

(2) A Windows 11 PC is home for a digital audio workstation (DAW). This computer software application handles the processing of live transmit audio processing chores for Amateur Radio transmissions. Regardless of configuration, modern DAWs use effects processors to tailor audio. Most DAWs host separate vst plugins such as noise gate, eq, and compression.

(3) 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.

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

(5) Some hams may prefer a purely analog (hardware-based) approach to audio processing.

How the scheme diagramed at left works

Digital audio output from the Diety VO-7U dynamic USB microphone is routed to the computer via a type C/A USB cable.
Digital audio is then processed within the computer by the software-based DAW system called "Reaper" hosting noise gate, eq, and compression vst plugins.
Digital transmit audio output Reaper is then routed via Type A/B USB cable to the Kenwood TS-590SG. 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. Finally, it is converted from digital to analog within the transceiver and transmitted.
55 watts RF output from TS-590SG transceiver.
700 watts RF output from the ACOM 1010 linear.
700 watts RF output from Palstar AT1500BAL to a full wave 40-meter loop antenna.
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 directly 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 conventional audio processing hardware.
Several higher-quality USB microphone choices have recently become available.
Potentially less susceptibility to RFI and hum than a conventional microphone.
Ability to process transmitted audio with software rather than more expensive external processing hardware.
Facilitates remote operations via the Internet.

Note: K4QKY enjoyed using the Rode NT USB condenser microphone for many years before recently shifting to the newer Deity VO-7U preferring to use a dynamic microphone. More about this highly capable microphone at https://deitymic.com/products/vo-7u/. There are many excellent USB microphones to consider including those described at http://www.soundonsound.com/sos/feb07/articles/usbmics.htm.

Alternatively, as described in Annex A bof this article, K4QKY has sometimes used a conventional XLR dynamic microphone such as the Heil model PR35 together with an external USB audio interface such as the Behringer U-Phoria UMC202 USB audio interface as described at http://adamivy.com/behringer-umc202hd-review/. This scheme can be overly complex for some. Much simpler with a USB microphone.

K4QKY's setup includes:

Hardware:

Kenwood TS-590SG
Dell PC running Windows 11 with internal Realtek sound card
Sennheiser Professional Profile USB Microphone with Boom Arm and 3 m USB-C Cable described HERE,
which is a highly versatile front-address USB microphone that is ideal for Amateur Radio or streaming.
Type C/A USB cable connected to the microphone and computer. Another type A/B cable routes the computer's process audio output 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. Alternatively, you can skip this download and resort to using the Windows built-in sound management system as discussed elsewhere on this site.
"Reaper", a complete digital audio production application for computers offering a full multitrack audio and MIDI recording, editing, processing, mixing, and mastering tool set described and downloaded from https://www.reaper.fm/. Reaper comes with all necessary vst plugins including noise gating, eq, and compression. Google keywords "Reaper videos" to gain a more complete understanding of this powerful yet easy-to-use DAW.

Note: K4QKY often uses 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.

Kenwood 590SG settings for software-based audio processing: (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 initially at level 90, no enhancements, 16 bit, 48000 Hz, and no exclusive mode.

> Speakers-USB Audio CODEC set as Default Communications Device. Properties set at level 37, no enhancements, 16 bit, 48000 Hz, and no exclusive mode. Adjust the level as necessary to adjust the transmit audio input to the 590SG as necessary to achieve a slight ALC meter deflection on voice peaks.

"Recording" tab

> Microphone set as Default Device. Properties set at level 37, 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 80, 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 “Default Playback Device” from the drop-down box. Select the “Levels” tab and set the slider to 80 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 - Download the trial version of Reaper as per the above instructions. Once installed and started, click "options" and then Preferences from the main menu. Then click "audio" and then "device. Next select "asio" from the audio system drop-down box. Lastly, click the "asio" configuration button to insure that your microphone is selected as input device and usb codec as your output device

Step 3 - insert noise gate (reaGate), eq (ReaEQ) and compressor (ReaComp) vst pugins in that order. Once installed, your main screen should appear similar to the screenshot below:

How to set up this configuration is regrettably beyond the scope of this article. K4QKY suggests that you first become thoroughly familiar with Reaper by watching a Madrid of videos that exist on the internet, Google keywords "Reaper videos for podcasting and voiceovers."

Finally, 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.

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 its "killer" features! Try it for yourself. 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 as discussed below.

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!

Download and install the driver for the interface from https://www.musictri.be/Categories/Behringer/Computer-Audio/Audio-Interfaces/UMC202HD/p/P0BJZ/downloads.
Run an XLR cable between mic and front panel of the interface.
Hookup a Type A/B USB cable between your PC and the rear panel of the interface.
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.