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I have recently taken over as the Technical Manager for my local Talking Newspaper (local news recorded for visually impaired listeners) - my technical background is more in IT/Web Development than in Audio, so I am trying to work out our set-up and see if it can be improved upon. At the moment, our weekly recording is output in three formats - C90 cassette, USB (MP3) memory stick and MP3 upload to our website. The biggest bug-bear in this is having to create audio for our cassette duplicators! Our set-up is five microphones, 1 CD deck, 1 Minidisc deck, 4 cassette decks and a PC. The mic's, CD, minidisc and one cassette are connected to separate channels on a Soundtracs Topaz 14:4 mixing desk. (CD and Minidisc are connected to the two stereo channels, rest are mono). The master out feeds into a compressor, which in turn feeds into a distribution amp. The outputs of the amp go to the four cassette decks and the PC. Our cassette recording is actually digitally mastered (we use Telex EDAT for creating the cassette copies), but we still run a set of analogue masters "just in case". That said, we have so few cassette copies being produced each week that the analogue masters are really of no benefit. Our PC based audio recording software is Adobe Audition 2 (I know it's out of date, but it does the job), and our recordings are 99% spoken word. The microphones we have have an ATT (I assume) switch on them, marked as 0dB, -7dB, -20dB. All the mics are set to -7dB. The mixing desk has a phantom power switch, but there is no noticeable difference when it is switched on or off (I assume this means our mics don't require power).

So, the background now set, my questions: 1) The compressor is currently between the mixer out and the distribution amp. I have mostly seen it suggested that the compressor is put as an insert on the mixers master channel. What difference would this make?

2) if using the compressor as an insert, should I group the mic/non-mic inputs to the mixer and just run the compressor on the mic inputs? Best I can work out, the desk supports two groups so I think this could be done.

3) one of the requirements of the cassette duplication is a signal that peaks about -3dB, and pretty much hits -3dB most of the time - we seem to get poor reproduction otherwise. We currently use the compressor to amplify the signal so we hit -3 and pull back if we go over, but I wonder if we would be better using less amplification and setting the mics to 0dB rather than -7? I'm really not sure how this setting on the mics effects the system, but feel we would get a less boxy sound if we did this.

4) We seem to get a lot of hiss when no-one is speaking, we also pick up the beer pump for the cellars of our neighbouring pub (our recording studio is in a basement). Is the EQ on our mixing desk sufficient to cut this out, or would a better EQ on an insert be sensible? If an external EQ, as it would be for all mic channels would it be placed before or after the compressor if the compressor should be run as insert rather than between output and amp?

I know this is rather a broad, and probably basic, question of setup, but any help would really be appreciated! At the moment I feel that the tech we have is hindering rather than helping as no-one really understands what it does.

  • Just to clarify on the beer pump... are you picking up audible noise with the microphone, or electrical noise in your recording? – Brad Feb 17 '14 at 5:01
  • it is an audible noise – OwenM Feb 18 '14 at 7:32
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1 & 2) You seem to understand the reason your compressor is a part of your signal chain. It sounds as though your compressor is set up to adjust the level it receives from your console to about -3dB. If it is working for you then it is in a good spot IMO. However, I don't think your Topaz console has a master insert.
An insert literally places your device (compressor) into the hardware signal chain and will change the actual peak information of the signal before it reaches the next stage; in your case, the compressor in a master insert would make that same ~-3dB adjustment before going to the master fader, instead of after like it is now. This has more benefits in an environment, like live, or in a studio, where you are more likely to adjust the listening level, as opposed to your current environment where your goal is to attain a steady level for a recording medium like your cassette and CD recorders. You should basically put your compressor wherever you want to even-out the level of your audio.

3) The ATT on your microphones stands for attenuation. At -7dB it is just bringing that level down a bit before it reaches your mixer input. It is truly up to you which setting on your microphones works best, just know that if you switch from -7dB to 0dB, it is likely you will just end up turning your GAIN/TRIM knob down approximately that much to compensate: overall, I think you will achieve a similar sound either way.

4) I'm glad to see somebody asked the question of electrical or background noise. I think if it is hiss, that the hiss will only be increased if you adjust your microphone ATT switches up to 0dB. Your absolute best bet IMO is finding a noise reduction plugin in your existing software, Audition, (or download Reaper multitrack software for free).

You can probably tame the hiss a little bit from your console just by rolling back the high EQ on your mic channels, but do this sparingly as it will deteriorate your audio pretty quickly - so try and find a balance. An additional hardware graphic EQ is the way to go if you are going to keep your existing setup, and it should be placed on the output channel as the last thing in the chain, or 2nd last (before your compressor) if you can get a console with master inserts. You may have other options if you consider different sound cards and whatever but that is beside your questions.

My apologies if parts of this is vague but your question begs a lot of information! I don't think this is all the answer you could receive, but please comment if you have further questions and we can try and help you further.

I hope this helps!

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    Adding to the many useful points made here: you say the hiss is only apparent when nobody is speaking. That could happen because the compressor is lifting the level way up when there is little signal. If the compressor has a gate function, that could be used to get rid of the noise in silent sections. Or you could add an outboard gate before the compressor. Just make sure you spend some time setting the treshhold and timing of the gate so that silently spoken words are not cut off. – EMV Feb 17 '14 at 9:58
  • Good point EMV. – Chris Bolseng Feb 17 '14 at 17:35
  • thanks emv - this is one of the reasons I was wondering about about the mic setting. I assume changing them from -7 to 0 would allow me to reduce the gain on the compressor, and therefore reduce it amplifying nothing. – OwenM Feb 18 '14 at 7:47
  • Your theory is correct but you would achieve the same result if you turned any item in your signal chain up +7dB and reduced the gain of your compressor. Putting your compressor earlier in the chain (I would use a microphone channel insert on the 2 most commonly used channels) would be a healthier change, however, you then have to be mindful of your output levels. If at any point you can consider an 8in-8out audio card and route everything through software, you might be able to improve upon all of the concerns you have mentioned.. – Chris Bolseng Feb 18 '14 at 16:17
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1) This isn't invalid, but it is a little weird. An insert goes before the fader where as the output goes after. Using the insert would give the master fader total control over the output level where as the positioning on the outside of the board means that a) it will protect the output gear from accidentally going too loud and b) it will make the behavior of the master fader perhaps a little less predictable as it will push more above the threshold of the compressor.

2)Compression on multiple channels is generally not the best idea, but may fit particular situations. Keep in mind that if any of the channels gets hot, it will pull everything in that group down, so if one person is talking quietly at the same time someone else is talking loudly, the threshold will be reached and the entire group will have the volume reduced. Sometimes that may be what you want, but it is fairly rare.

3) For best quality, you want to maximize signal level without additional amplification and without clipping. You want to have the mics at unity unless they are clipping. Pre-amps should supply good strong signal to keep noise down and then you should mix reductively wherever possible.

4) It really depends, sound proofing would be your best bet, but you could also make pretty good work with a hum eliminator for the noise (or a power conditioner if you don't already have one). You'll also want to make sure you don't have any ground differential issues from devices on multiple circuits. For the beer pump, you may want to try a 31 band EQ for increased fine tuning if sound proofing isn't an option, but it will likely still be audible since the motor isn't a fixed frequency but rather a complex set of frequencies.

  • thanks for the reply. we normally only have one reader speaking at a time so on the whole only one of the five mics will be live at any one time. next time I'm in the studio (we only record on 2 evenings a week) I'll try changing the mic settings to 0dB. I've arranged with one of our recording teams to come in early next week so I can play with the compressor settings without worrying about getting the recording done! – OwenM Feb 18 '14 at 8:00

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