How to Use Audio Equalization (EQ) and Compression to Partially Mitigate Effects of Toxic Sound in Remote Interpretation.

Update, February 6, 2021: I have been using the set up with TASCAM US 2 X 2 USB audio interface, DBX compressor/limiter, DBX equalizer and Beyerdynamic DT 394 SIS headset [see specs below] for about 9 months now between three and six hours of online use every day [except weekends] and I could not be happier. The platforms that I primarily used were WebEx and Zoom. No tinnitus, no dizziness, no headaches, no hyperacusis or any hearing shifts. Yes, the sound is still bad, but I do not have to cringe when somebody drops a microphone.

Hardware description:

Top rack: 

- Furman power conditioner 

- Two TASCAM US 2x2 USB audio interfaces (for 2 laptops) 

- DBX channel strip for my Beyerdynamic DT394 SIS headset mic (does not always work well)

Bottom rack:

- DBX 2 channel equalizer 

- DBX compressor limiter 

- HEADAMP headphone amplifier with ability to listen to binaural mono for either left or right channel.

The original article follows:

A very long disclaimer: we are interpreters and cannot be turned into audio technicians, it is not in our job description. We also cannot be responsible or liable for poor audio quality, ultimately, it is the responsibility of the platform provider and/or hiring organization. The best we can do is to say “inaudible” as often as needed. As professionals, however, we should follow best practices and attempt to achieve the highest quality of interpretation, and speech intelligibility plays a crucial role in it. The measures described below are nothing but stop gap measures in extremis, and, hopefully, will not be needed when remote interpretation, especially remote simultaneous interpretation (RSI), will re-assume its marginal position after the COVID-19 crisis. It also does not mean that remote-from-home should be recognized as an acceptable or preferred solution. Any practical instructions below are not professional technical advice and/or endorsement but an opinion of the author and following them is fully at your discretion. When configuring and using any audio equipment, follow all manufacturer's instructions and remember about safety practices, including hearing protection.           

Executive Summary: The COVID-19 situation resulted in an unprecedented deterioration of working conditions for interpreters, including having to work with toxic or “dirty” sound.

As a minimum, interpreters are recommended to explore using an audio mixing board with sound equalization (EQ) and a built in one knob compressor combined with an ISO compliant headset. 

While adjusting frequency bands of the input feed and using compressors alone will not solve the toxic sound issue, it might help remote interpreters mitigate some of the effects, improve speech intelligibility and significantly reduce fatigue.

Nature of the Problem

 â€œDirty” sound is not a new issue for conference audio technicians, in fact, a lot of their work time is devoted to cleaning up audio feeds to remove overtones, RF interference, hum, etc to achieve the best sound quality interpreters and delegates can enjoy. That is why they would ask you never to put your cell phone next to your console or headsets, for example! (“Dirty” sound needs to be distinguished from the acoustic shock, see https://app.box.com/s/p665w9tu7ox4ovwj1w213uyuybs9k34h )

In the situation of remote interpretation and RSI we are facing now, sound quality interpreters receive goes beyond just being “dirty” but becomes toxic.

A draft list of "toxic sound" characteristics may include, for example:

• Reduced frequency range of VOIP/Wideband audio resulting in diminished speech intelligibility as compared to ISO 20109 standard
• Various VOIP artefacts, overcompression, packet loss with comm interruptions, codec issues
• Not using proper headsets or microphones by speakers which results in too "hot" signal or low SNR (signal to noise ratio) with background noise
• Significant variations in mic gain/volume among presenters which is particularly bothersome in dialogue/polylogue formats.
• Automatic mic gain
• Network shrieks
• Prevalence of low frequencies, "muddy" audio
• Uncorrected or uncorrectable transients
• Outside interference, such as traffic noise, barking dogs, crying babies, etc.
• Echo and overall poor acoustic environment in delegate location, etc

Underdeveloped and poorly developed RSI platforms do not allow to correct or mitigate some of the issues eg by not providing bass and treble adjustments or introducing peak limiting.

While some of the issues may be mitigated by requiring delegates to use ISO compliant headsets (see the list of AIIC recommended headsets here: https://aiic.net/page/9007/thc-newsletter-issue-1/lang/1 ), most problems with the source language sound feed are inherent in remote interpretation and cannot be completely solved at this stage due to the state of the development of the technology. Wearing a headset by delegates must be a requirement not a request and compliance must be strictly monitored by RSI platforms and/or conference organizers. Imagine a delegate at a regular conference refusing to wear a wireless lapel mic? How is an online conference different?

One tweak may, however, help interpreters improve sound quality a little bit.

In professional conference settings all audio streams are passed through audio mixing boards, either hardware or software based. A mixing board is a physical or virtual electronic unit that allows to manipulate audio signals by combining or dividing them, changing volume and intensity, applying audio filters etc.

Sound Equalization (EQ)

 One of standard adjustments is sound equalization. Equalizers are devices built in into mixing boards that are “volume adjustments” (though “gain adjustment” would be a more correct technical term) not for the entire signal like the master volume knob is, but for specific frequencies.

For example, you may want to boost 2,000 Hz and make it sound louder than the rest of the signal, if needed.

Equalization Bands

The range of human hearing is generally considered to be 20 Hz to 20,000 Hertz (Hz), or oscillations per second. It is clearly not possible to install 20,000 physical adjustment knobs to tweak each frequency individually, although, hypothetically speaking, it may be done as a software solution if needed.

Luckily, it is not necessary for our purposes. We can operate with frequencies in “bands” or pre-selected ranges of frequencies. Each range can be adjusted at the same time by one tweak.

They usually speak of 7 bands:

• Sub-bass (20 Hz – 60 Hz)
• Bass (60 Hz – 250 Hz)
• Low Midrange (250 Hz – 500 Hz)
• Midrange (500 Hz – 2 kHz)
• Upper Midrange (2 kHz – 4 kHz)
• Presence (4 kHz – 6 kHz)
• Brilliance (6 kHz – 20 kHz)

However, you can break the hearing range into as many bands as needed for a particular task.

More advanced equalizers may have 30 or more bands, in other words, the human hearing frequency range may be divided into 30 or more segments each being controlled individually.

As interpreters, we certainly do not need that many, because it will over-complicate the setup. Let us concentrate just on three basic bands: low, mid-range and high frequencies.

Frequency of Human Voice. 

Fortunately for us, the frequency range of human voice is narrower than hearing frequency. The f0 or the frequency at which our vocal folds vibrate (called “fundamental frequency”) is only approximately 100-120 Hz for men, one octave higher for women and around 300 Hz for children. That is why ISO standards for simultaneous interpretation (e.g. ISO 20109) require that the lowest frequency we can hear in our headsets be 125Hz but 100 Hz is even better and may produce more rich sound if needed.

That basic tone or the vibration of the vocal folds, however, is enriched with additional overtones that have different frequencies.

As a result, some frequencies are more important for speech intelligibility, for example, in Western non-tonal languages the range between 1,000 Hz and 2,000 Hz is the most important. That is why if we give a priority to that range and boost its volume compared to other bands ( completely remove anything under 80Hz i.e. use a high pass filter plus boost the 2,000Hz range), it may improve speech intelligibility, make the sound cleaner and easier to understand. It will also remove a lot of low frequency humming in the system and will also mean that we may need less volume and our ears will be less tired.

Interestingly, in different languages, different frequency bands may be important for speech ineligibility as audio engineers' experience with fine-tuning hearing aids for various languages shows. In Chinese, for example, lower frequencies, may be important because Chinese relies more on pitch changes in lower-frequency vowels. (They can even customize hearing aids for bilinguals by creating different listening programs e.g. one for English and one for Japanese! See an excellent introduction into the topic here: https://www.hearingreview.com/practice-building/practice-management/how-hearing-aids-may-be-set-for-different-languages)

Frequencies above 3,000 Hz are very important too and their loss means diminished speech understanding. Listening effort is also increased if that range is not reproduced. That is one of the contributing factors to the toxic sound, because VOIP and online systems may only reproduce frequencies up to 8,000 Hz and not 15,000 Hz as required by the ISO standards for simultaneous interpretation. Regular phones only reproduce 300 Hz to 3,400 Hz, so a lot of important frequencies are lost. (see video comparison of sound quality here: Remote Simultaneous Interpreting: Options and Standards https://www.ata-divisions.org/ID/remote-simultaneous-interpreting/ )

The range above 15,000Hz can be safely removed because it is not used much for human speech.

High quality headsets that reproduce the full spectrum sound (20 Hz- 20,000Hz) may actually work against you and reproduce the range you do not need as an interpreter (i.e. 20Hz to 80Hz and 15,000Hz and above). It simply adds acoustic pollution, especially when you interpret voice overlapping with music. Ironically, a headset can be “too good!”

NB: This quick headset test can show you what frequencies you can actually hear in your headset with your current audio setup. Note that at high frequencies it can reveal the degree of your hearing loss (if any) OR the actual high frequency response of the headset, so take it with a grain of salt. This is not a hearing test. The Ultimate Headphones Test https://www.youtube.com/watch?v=G-w0nUWau1Y

 What does it mean practically for interpreters?

Simply, that we cannot afford any longer not to manage ourselves the frequency bands of the sound we are interpreting.

RSI and other online platforms simply feed you the sound “as is” probably automatically processed and without consideration for our needs as interpreters. These systems mostly have no controls to adjust frequency ranges.   

Adjusting frequency bands will improve frequency response of full range headsets, make speech more intelligible and, hopefully, mitigate some of the effects of toxic sound!

Experiment with an equalizer, either as a standalone unit or as a part of a mixing board!

You have a choice between software and physical ones. Some software ones (some are freeware) may not allow you to manipulate sound in real time though.

Hardware mixing boards and equalizers will:

-      Provide real time signal processing

-      May have a steep learning curve because you will need to learn to adjust at least 5 parameters for a standalone equalizer and many more for a mixing board.

-      Cost you US $80 or more

-      Take some real estate on your desk

-      Have actual knobs including rotary volume controls like a regular interpreter console.

-      Allow quicker adjustment compared to software solution, especially if you are using an online platform at the same time although using software ones is certainly an option.

Hardware equalizers and mixing board cannot work with USB headsets, though. 

In that case you can:

-      get an external USB audio interface for your computer you will connect your mixing board and/or equalizer to and plug in an ISO compliant headset into your mixing board.

-      get a mixing board with a built in USB interface you connect directly to your computer: some equalizers have USB interfaces already, some do not. (See review of this configuration as an example: https://app.box.com/s/fm9iy4laut8v71u7amojsv931st9p3js The configuration successfully tested by the author was to add to this setup a mixing board with a one knob compressor, so the XLR mic connector goes into the TASCAM unit and the source language sound is fed into the mixing board via TASCAM and the headphones are plugged into the mixing board. The reason for the setup is that the used mixing board has only one output.)

Sample frequency adjustments.

As an example, we will take a very simple but very reliable Yamaha analog Mixing Console MG10 (Brochure: https://usa.yamaha.com/files/download/brochure/1/330961/mg_brochure_en_webcatalogue.pdf Specs: https://usa.yamaha.com/files/download/other_assets/0/331000/mg10xu_en_ts_c0.pdf ) and show what adjustments need to be made.

Please, note that sound mixing does not have many hard rules but a lot depends on individual preferences, presence or absence of hearing loss, model and type of headsets and a score of other factors. Readers are encouraged to experiment and see what works best for them.

It may be equally easy to achieve better sound or mess it up completely. Also, keep in mind that your settings need to be adjusted for each situation or even each speaker: you cannot set it up once and forget about it. It will pay off with better sound quality, however, and much less fatigue. 

The console has a 3 band fixed equalizer:

-      100Hz, LOW

-      2,500 Hz, MID

-      10,000 Hz, HIGH

 While the 3 bands will provide only rough adjustments, they will work, and we do not want to over-complicate the setup for colleagues who never used an equalizer or mixer before and to review a more complex equalizer with more bands (For more complex vocal adjustments see A Master Guide To Voice Equalization—How To Apply EQ to Voice Recordings https://music.tutsplus.com/tutorials/a-master-guide-to-voice-equalization-how-to-apply-eq-to-voice-recordings--cms-25184 )

Note: before you make any EQ and GAIN adjustments with this mixer, make sure compression is turned off. For this mixer, turn the yellow COMP knob above the EQ knobs (see picture) counterclockwise all the way to the left to zero. 

After you have set up and connected the mixer and the headset (see manufacturer instructions for that) and are ready to make EQ adjustments:

-      Make sure your volume and gain controls are not too high, the red PEAK LED should blink only occasionally and the PEAK indicators on the right should not go too much into yellow, that is stay around 0 dBFS (see dBFS definition here: https://en.wikipedia.org/wiki/DBFS ).

-      The first adjustment is always GAIN

-      When working with EQ, you can “cut” (reduce) or “boost” (amplify) audio signal for each band individually by +/- 15dB

You need to decide whether to boost or cut and by how much (dB) for each band for each interpretation event or even speaker depending on audio quality you are getting. For example, in low quality VOIP/online you may want to boost HIGH significantly to compensate for the lack of high frequencies and improve speech intelligibility. 

Below are only very general recommendations (different adjustments may need to be made for male and female voices or depending on the distance from the microphone, online and offline). Do not make big adjustments to start with: 3-5 dB steps are usually recommended and you have 15dB each way to play with):

See the table below for common tweaks:

Start all EQ adjustments from the "detented position", it is the "12 o'clock" position and you will need a tiny bit of force to get the knob moving.

As a starting point in good quality audio feeds try to (see if it works after each adjustment):

-       Always start in the MID range (or LOW MID in equalizers with more bands than 3) Boost MID to bring voice to the front but do not make it too bright

-       Slightly boost LOW but do not make it too boomy 

-       Cut HIGH without reducing speech intelligibility

Online VOIP speech is extremely “muddy” with a lot of low frequencies that hinder speech intelligibility and increase fatigue.

For poor quality online sound try:

-       Boost HIGH to improve speech intelligibility and make the sound more bright

-       Boost MID as needed

-       Try to reduce LOW to see if it helps remove low frequency online noises and make the sound brighter too.

 As a general rule, start with cutting and see if it helps first, cutting is better than boosting.   

Opposite scenario: sometime you, actually, encounter an opposite scenario: a voice that is too bright with a mic that is too close to the mouth ie too "hot" and a shrieky high pitch voice: it sounds almost painful. Limiter/compressor here (see below) is a must and you can try to kill high frequencies and play with mid lows to remove the excessive parts of the spectrum.

Your goal is to achieve sound that is pleasant to the ears, bright but not shriek, rich but not dull or dark. Sometimes you may have to sacrifice low frequencies (cut LOW, boost HIGH) to get better speech intelligibility online.

Experiment! There are very few hard rules in audio mixing, so experiment and see what works best for you in a given situation. Over time, you will train your ears and the process will become almost automatic.

You can try this professional solution for sound engineers too: Train Your Ears is a piece of software that presents you with a number of audio quizzes. One of them is trying to guess what frequencies have been changed. Not free, but lots of fun! https://www.trainyourears.com/

If you have a more advanced equalizer, you may want to fine tune voice frequencies even better as per this cheat sheet that shows where to cut and where to boost:

(Source: the Vocal EQ Chart (Vocal Frequency Ranges + EQ Tips https://producerhive.com/music-production-recording-tips/how-to-use-a-vocal-eq-chart/ )

Remember that EQ frequency settings may also be language specific.

For example, “In Arabic (Semitic) languages, there are many high-frequency consonants (velars, uvulars, and pharyngeal fricatives) that indicate a need for more high-frequency gain than would be specified for an “English program.” (Source: https://www.hearingreview.com/practice-building/practice-management/how-hearing-aids-may-be-set-for-different-languages )

Because you may have vastly different equalization needs in your specific language combination, it may be hard to find a frequency setup suitable for both languages at the same time.

 In that case, fine-tune for the language you cannot hear well because of poor audio quality. Tweaking for B or C language is probably better because speech intelligibility may be better in our A language. 

Compressors and Limiters

Sound equalization (EQ) is not the only adjustment you can make yourself, fortunately.

A major problem with toxic sound is the so called “transients.” They are sound peaks that have a high amplitude and are short-duration sounds at the beginning of a waveform, for example, bumped microphones, microphones that are too close to the mouth (too much gain), sounds like “p”.”t”,”k”, etc. Primarily, they are explosive type noises. The problem online is that many speakers use low quality microphones that are too far away from the mouth and are just not loud enough compared to the background noise. In technical terms, there is a low signal to noise ratio (SNR), for speech to be intelligible, SNR has to be at least 35dB. (see more on what decibels are here: What Every Interpreter Must Know about Decibels. https://www.dhirubhai.net/pulse/what-every-interpreter-must-know-decibels-cyril-flerov/ )   

We end up in a situation when low signals are too low and high signals are too high, it is called a wide “dynamic range” Working in this wide range is very unpleasant for the interpreter, because he would increase volume to hear low amplitude “valuable” sounds (speech) but will then be exposed to explosive high amplitude ones (transients), it is a constant acoustic assault and a major contributing factor to fatigue and stress.

A solution would be to make loud sounds quieter and quiet sounds louder, it is called “compressing the dynamic range.” Though "making loud sounds quieter and quiet sounds louder" is not a correct technical description for how compressors work, the idea behind their operation is similar. Compressors are devices (hardware or software) that work with a certain sound threshold you set up and reduce amplitude (“volume”) of anything that is above the threshold. They can do it quickly or slowly and to the degree you set up.

They may also boost quieter sounds and eventually make average amplitude of your audio higher and the speech more prominent and easier to understand no matter how quiet or loud people actually speak. Its' effect be observed best in situations when several people speak at different volumes, a compressor harmonizes that difference, compressed sound has more energy and “punch”. You will also be exposed to fewer unpleasant transients if any.     

Should I use a limiter or a compressor?

A limiter is basically a compressor that has been configured in a very peculiar way: while a compressor may let in some “louder” sound than the threshold you set up, a limiter absolutely forbids it and keeps the level strictly at threshold (brick wall, infinite compression ratio).

For example,

-       you set up 93dB as your threshold,

-       a compressor is set up to let through only 1 decibel of excessive signal per 4 excessive dB it receives (1/4 ratio)

-       the compressor receives extra 4 dB so the total signal is 97dB

-       the compressor lets through only 1 extra dB so the total resulting signal is 94dB which is ISO requirement for peak loads in simultaneous interpretation (ISO 20109) and not 97dB without the compressor.

Comparing limiters and compressors:

Limiters prevent a bad sound, compressors make a sound better, that is why while limiters are perfectly OK to use, compressors will be more useful to interpreters and should be your first choice. Correct models of compressors can be configured too to block dangerous acoustic peaks.

What does it mean in practical terms?

Look at Yamaha MG10 yellow COMP knob (above the EQ panel) that we set to zero earlier. It is a “one knob compressor”, a quick way to try to harmonize your sound and make it more pleasant and get rid of transients.

While it is a “quick and dirty” and far from being precise solution, it allows you to use compression without having to learn more complicated compressor interfaces with many more settings and a pretty steep learning curve.

After you adjusted EQ bands, rotate the yellow knob clockwise and see if it helps speech intelligibility and creates fewer explosive peaks in the audio feed.

Now, when you know how compressors work, you know what to expect. Apply compression after you set up correct levels of GAIN, EQ, LEVEL, and comfortable headphone volume and then you may have to tweak EQ again.

A proper compressor or limiter may have up to 5 parameters to play with: threshold, compression ratio, attack time, release time, output gain.

Remember that applying compression will usually reduce the overall volume so you may want to compensate by increasing GAIN or headset volume very slightly but without the red PEAK LED blinking too often and not going into the yellow zone too much (or adjusting OUTPUT GAIN for compressors with a more complex interface).   

Use EQ combined with the compressor function to achieve the best realistically possible sound quality for given conditions. Experiment and tweak as needed. It will not be a full solution for the toxic sound, though, becasue currently n such solution exists.

Adjustments may need to be made for each model of the headsets you own or use too. Each headset has its own "frequency response" i.e. reproduces some frequencies stronger. Because many headsets are customized for the mass consumer music or computer game market, they may have exaggerated bass and elevated higher frequencies, with some headsets reproducing up to 35,000 Hz!

The author also used the Yamaha MG10 compressor on its max setting during remote interpretation and experienced mic feedback from a remote delegate who put his earphones next to his microphone. The Yamaha MG10 compressor behaved admirably and reduced the feedback to such a degree that it took the interpreter a fraction of a second to realize that the barely audible mosquito type buzz was actually a mic feedback. So using compressors may be studied as an option to reduce the acoustic shock hazard in its broader definition of acoustic shock (below 120dB).    

Should I Equalize or Compress First?

If your equalizer and compressor/limiter are separate units, which one should go first in the signal path? "As a rule, using EQ in front of your compressor produces a warmer, rounder tone, while using EQ after your compressor produces a cleaner, clearer sound." (Source: https://online.berklee.edu/takenote/eq-before-or-after-compression/)

So it seems, in this setup you should let the compressor go first because clearer sound is what we are trying to get for voice!

Other Hardware Options

(have not been tested by the author)

CANFORD Headset Limiters

This is a most interesting option for a limiter: you can custom order the limiter that will be fit into the headset cable.

The limiters are manufactured for Canford Audio under a BBC license and are supplied fitted to a chosen pair of headphones and calibrated to a defined SPL. You need to specify 94dB (ISO 20109) as the threshold. They are not available separately, due to the requirement for specialized calibration equipment.

Note that these are built in/non-removable pre-configured inline limiters for individual headsets. They are not compressors and cannot be moved between headsets.

https://www.canford.co.uk/CANFORD-HEADPHONE-LIMITERS

Another good candidate for a mixing board is Yamaha AG06: https://americansongwriter.com/yamaha-ag06-mixing-console/ it has a compressor on it (Yamaha, actually, specializes in compressors), and you can even set attack times etc. in the software. (Attack time is how quickly the compressor responds, for voice the recommended setting is around 2-10 milliseconds (source: https://www.musicianonamission.com/how-to-use-a-compressor-plus-10-top-tips/ ) for vocals and voice, a very fast response.)

iEQ31 Dual 31-Band Graphic EQ/Limiter with Type V? NR and AFS?

A decent option albeit more expensive.

https://dbxpro.com/en/products/ieq31

iFi IEMatch seems to work well with sensitive headphones as per this review and definitely deserves more testing. https://www.soundphilereview.com/reviews/ifi-iematch-review-1846/ Another advantage is its portability. The device has not been tested by the author.

Technical solutions to combat toxic sound may be very expensive and RSI platforms pass that cost to the interpreter instead of assuming technical and legal responsibility and liability for severely deficient sound quality and substandard working conditions.

Interpreters, who have no choice but to work remotely, have to, however stay professional and try to save their hearing and sanity by adhering to best practices including, if they choose, by cleaning the sound feed, for example, as described above.    

Happy interpreting!

IMPORTANT UPDATE, 28 May 2020. I added a dbx 31 band equalizer and a dbx compressor / limiter to my setup (see below). It had a surprisingly positive effect on my hearing and mental state when listening to online and even YouTube sources, and even music. The limiter removed unpleasant peaks and softened my mild to moderate hyperacusis developed from Zoom over the past few weeks. Filtering out hot parts of the spectrum and mud helped immensely too.

With that setup the most useful protocol for Skype/Zoom type of online sound seems to be:

• install limiter/compressor before the equalizer in the signal path
• enable EQ bypass
• do not use gate
• set up threshold to -40 dB as initial setting
• set up compression ratio to infinity as an initial setting
• do not use OverEasy mode
• set up the fastest attack time
• set up slow release time (3 pm to 5 pm dial position)
• do not use auto mode
• dial up Output Gain to +2 dB though it may not be necessary if the signal is too hot.
• After this initial setting change threshold and decrease release time until you get more natural sound without distortions.
• Compare before and after by using compressor/limiter bypass.

This procedure if done correctly should remove harshness, transients and equalize different mic inputs.

• Then disable EQ bypass and equalize frequencies as needed by cutting or boosting.

List of the hardware in the reviewed configuration successfully tested by the author:

Disclaimer: this is not the most efficient and more expensive configuration (although it works very well), because Yamaha MG10 has only 1 output and no USB interface, that is why the headset XLR mic was plugged into the TASCAM USB interface. The mixing board was used originally as an equalizer/compressor/monitor and when separate EQ and compressor/limiter were added - as a headset volume control.

-       TASCAM US 2x2 USB Audio Interface and included USB cable

-       Beyerdynamic DT394 SIS headset with XLR mic (another ISO compliant analog (not USB) headset may be used. Professional connectors in the described hardware are ? inch so you may need up to two 1/8 female to ? male adapters if your analog headset uses 1/8 male connectors)

-       ? male to ? male patch cable to connect TASCAM to Yamaha MG10 (length as needed).

- Rolls XLR MS111 mic cough cut between the Beyerdynamic mic and the TASCAM audio input. The switch has been configured not to lock in the muted or the unmuted positions, just like a professional console cough cut. Used with XLR patch cable. https://rolls.com/product/MS111

- dbx 166xs Compressor / Limiter / Gate

Built in PeakStop? limiter provides an absolute ceiling for peak excursions or large transients that could damage your equipment (or hearing).

https://dbxpro.com/en/products/166xs

- dbx 231s Dual Channel 31-Band Equalizer

The dbx 231s includes two 31-band channels of 1/3-octave equalization, ±12 dB input gain, switchable ±6 dB or ±12 dB boost/cut range, 20mm nonconductive sliders, an intuitive user interface, and output level metering.

https://dbxpro.com/en/products/231s

Additional Sources and Reading/Viewing Materials:

General:

A compilation of terms commonly used to describe sound and audio equipment.

https://www.head-fi.org/articles/describing-sound-a-glossary.12328/

Head-Fi: Headphone Reviews and Recommendations

https://www.head-fi.org/

A very decent headphone buying guide. Remember that for simul we need to use semi-open or semi closed and preferably "on ear" headphones or headsets. NEVER use anything that goes into your ear such as in-ear monitors or earbuds. https://www.head-fi.org/articles/headphone-buying-guide.14163/

Facts about Speech Intelligibility

https://www.dpamicrophones.com/mic-university/facts-about-speech-intelligibility

Sound Equalization:

Beginner’s Guide to Using EQ

https://homerecordinglab.com/beginners-guide-to-using-eq/

How to EQ Vocals

https://www.youtube.com/watch?v=jT_753byV8I&list=RDbybWokSUnBg&index=27

Key frequencies to improve your mixing

https://www.soonvibes.com/en/blog/frequences-cles-pour-ameliorer-votre-mixage/#4

Sound Compression:

How to use an audio compressor or limiter in music, podcasts and video

https://medium.com/keypleezer-music-tech-blog/how-to-use-an-audio-compressor-or-limiter-in-music-podcasts-or-video-dead2b2b3e68

Example of Compression for Voice Over

https://www.videomuzik.biz/video/compression-for-voice-over-the-vla-3a-black-rooster-audio-hd-05h5g4r4y5i37565m4o4r3.html

Compression 101: How to Use a Compressor

https://www.youtube.com/watch?v=E2CxMCkvml8

How to use a one knob compressor

https://www.youtube.com/watch?v=YpOZcaT3txM

Understanding One Knob Compressors on Mixing desks

https://www.youtube.com/watch?v=YxW4HxgX1KM

Compressors Explained

https://www.youtube.com/watch?v=IbIC7B4BU6g

How To Use A Compressor: Compression 101

https://www.youtube.com/watch?v=s1U2hZHHFY0