How to do a phone interview

30 June, 2010 at 12:28 pm (Production Studio, WTJU)

1) Setup, headphones and monitoring

– Check that no channel faders other than the two to be used are up.


– Announcer and assisting engineer should both be wearing headphones.
– Check that L/R MIX is selected in the MONITOR section.
– Check that MONITOR is selected in the PHONES1 and PHONES2 sections.
– Check that phones levels are right – about 10 o’clock is a good starting point.Monitoring and headphones section

Main L/R faders

– Check that the main LEFT/RIGHT MIX faders are at unity gain (marked ‘U’ next to the fader.)

2) Setup the announcer’s microphone channel

Announcer mic channel

– Put the channel 1 fader at unity gain (marked ‘U’ next to the fader.)
– Check that it is not MUTE and is selected for L/R MIX.

– Check that the AUX SEND 1 is at 12 o’clockTrim and Aux send 1

– Have the announcer speak with emphasis as he would on the telephone.

– Adjust the channel 1 TRIM control so that the main L/R meters are bouncing just above 0 dB.Meter bounce

– Push the red button which mutes the speakers.


3) Start recording audio

44k 16bit Mono– In Adobe Audition use 44100 Hz, 16 bit, mono
– Record to a CD in the HHB burner as an emergency backup

– Excess talk at the start or end of the recording can be edited out later.

4) Call the interview guest

The telephone channel– Put the channel 18 fader (labeled ‘PHN’) at unity gain (marked ‘U’ next to the fader.)
– Check that it is not MUTE and is selected for L/R MIX.
– Call from the white telephone or have the guest call 434-971-8678.
– Push the ON button on the Gentner phone interface.

Phone interface on
– Hang up the white telephone.
– Chat a bit with the guest and adjust the channel 18 TRIM control so that the main L/R meters are bouncing just above 0 dB when the guest is speaking.

5) Final adjustments

– Push both the channel 1 fader and the channel 18 fader up above unity to +5 dB.
– This will provide for a louder signal going into the computer and the CD burner.
– Make sure the record meters are not clipping on either.
– Check that the balance between the announcer and guest level is good – make fine adjustments to the faders.


Announcer on left, phone interviewee on right, but use your ears!

6) Do your interview

– Be prepared!
– Get a station ID!

7) Ending the interview

– Stop the recording devices.
– If you don’t actually want to hang up yet, pick up tGentner offhe white telephone.
– Push the OFF button on the Gentner phone interface.
– Save your files / finalize your CD.
– Turn down the PHN and MIC channel faders.
– Proceed to editing your final product.

That’s it!


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Session 3: Microphones

10 March, 2010 at 11:01 pm (Live Studio, Production Studio, WTJU)

For those who have missed it or would like some more review, there will be one more session on microphones and setting up the mixer channel next Tuesday the 16th at 6:30.

After that, starting Wednesday the 24th, the workshop will be every Wednesday night, and so will start to move faster.   The next subject will be introducing Adobe Audition for recording promos, etc. in the production studio.

There area few different ways to categorize the microphones you are likely to encounter:

  • Technology: dynamic or condenser
  • Directionality: unidirectional or omnidirectional
  • Sensitivity: how much amplification is needed

Dynamic Microphones:

The dynamic microphones we have are the Electrovoice RE-20 and the Shure SM-58:

Both are suitable for vocals;  The RE-20 is good for kick drum or acoustic bass;  Both are good  for horns, electric guitar or other amplified speakers.   They are both considered general-purpose mics which do not necessary excel at anything but are usually workable for just about everything.

Electrovoice RE-20


Shure SM-58


Dynamic mics tend to be not very sensitive.  So that means you will have to turn up the mixer channel TRIM control higher for these mics than you would for a more sensitive microphone.

Both are unidirectional, meaning that they are most sensitive to sound arriving from the front and least sensitive to sound from the back.   For sound arriving from the sides, above or below, their sensitivity is somewhere in between the two limits.

Unidirectional microphones are useful for keeping the sound coming into the mixing board clean:  A singer’s guitar won’t tend to bleed into the vocal mic.  The drums from across the room won’t tend to be very loud on a saxophone mic.

Another important characteristic to keep in mind is that whatever sound does arrive from the sides and the back tends to have its high frequencies rolled off.   This is called the off-axis coloration of the mic.   So if a singer does not directly address the front of the mic but instead sings into the side of it, a more muted tone will be captured.  This is generally to be avoided but sometimes can be used for effect.

Condenser Microphones:

Condenser microphones require phantom power (described below.)  The condenser mics we have are one Audio Technica 4033 and two Earthworks QTC-40.

The AT-4033 is great for acoustic stringed instruments;  It also works well for close-miking an electric guitar amplifier.  (Just keep in mind that it is very sensitive so you won’t be turning the TRIM control up much at all.)

Audio Technica 4033

AT-4033 front-forward

The AT-4033 is unidirectional and sensitive. In the photo there it is shown with the front of the mic facing the camera.  There are a couple switches on the back, which should be facing away from what you are trying to capture.   It is always used installed in its shock-mounting clip.  Before you use this mic, be sure you know how to properly install it in its shock-mount.  It looks like this:

AT-4033 in shock-mount

AT-4033 in shock-mount

The Earthworks QTC-40 is the ideal ambient room mic.  It’s also great for an acoustic group to gather around and share.   Just bear in mind that you will probably still want to put a dynamic microphone on the lead vocals unless the singer is a real belter.  Even then you really need the dynamic for talk portions of a live broadcast.  The QTC-40 is good for acoustic stringed instruments but you have to be more careful about what other sounds in the room bleed onto it.

Earthworks QTC-40s


The QTC-40 is omnidirectional and sensitive.  So that means that the only factor affecting how loud a sound appears through this microphone is the distance of the sound source from the microphone tip.  (And of course how loud the sound is in the first place.)   It matters not at all which direction the tip of the microphone is pointing relative to the sound source.  That’s why you have to be more concerned about what bleeds into it from other unintended sound sources.   Being sensitive, you won’t often be turning the mixer channel’s TRIM control up very high with these mics.

I think it would work well to place one of these centered above a drum set, just out of the drummer’s way.   Coupled with an RE-20 on the kick drum this would make a nice minimalist but high-fidelity drum kit microphone setup.

As shown above, the QTC-40 clips are stored in the box with the microphones after each use.  The clips do not fit any other mics in the room so it is best to keep them separate.

Protecting microphones against free-fall:

RE-20 on stand

RE-20 on stand

When placing microphones on stands, always make sure they are stable and will not fall over if bumped.   This is especially critical for the RE-20 which is heavy and fragile.

In its case you should always use a tripod stand and make sure that the weight of the microphone is centered over one of the three legs, as shown here.

The front of the mic and the “front” leg are both facing the singer.

Generally you want to avoid having the “tails” of the mic stand stick out into the room where they will be disturbed.

Phantom Power:

Condenser microphones cannot operate without phantom power.   It’s called phantom because it doesn’t affect dynamic microphones whether it is there or not.  Phantom power is a low-current 48 Volt dc power supply which is provided over the microphone cable.   On the Mackie mixing board there is a switch at the very top of channels 8, 16, and 24.  Each switch enables phantom power for a bank of 8 channels (1-8, 9-16, 17-24.)

Make sure the phantom power switches are off (UP) before connecting condenser microphones.   Once all the mics are connected, turn on (press DOWN) only the banks you require.   Turn the phantom power back off before disconnecting microphone cables.

For convenience, and to keep things simple, I prefer to only run condenser mics on channels 17-24, plugged directly into the mixing board.   Dynamic mics go through the snake box on the floor and appear on channels 1-8.  This way only the phantom power switch above channel 24 is ever used.

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Session 2: The mixer channel

24 February, 2010 at 11:01 pm (Live Studio, Production Studio, WTJU)

Tonight’s session was all about the mixer channel.  Specifically each of the channels on an early 1990’s Mackie 8-bus 24 channel mixer.   One of the reasons Mackie sold so many mixers over the years is that their documentation is very good.  The manual will educate and entertain you while going over all the features of the mixer.   The mixer channel is covered in detail on pages 4-9.  We’re going to look at page 1 in detail below: the checklist for setting up each mixer channel before use is so important they put it on page 1 of the manual.

Here’s a block diagram of the mixer channel:

Mixer channel block diagram

So what that’s saying, from top-to-bottom:

  1. The signal comes in through one of the inputs at the top of the channel, either a microphone input or a line input.
  2. The signal is amplified.  The amount of amplification is controlled by the TRIM control, a white knob at the top of the channel.  For line-level inputs like CD players and direct outputs from electronic keyboards, very little amplification is required, so the TRIM control will be fully counterclockwise.  For microphones and quiet signal sources such as a turntable, the TRIM control will be set somewhere above the minimum.
  3. Next the signal passes through the equalizer section, mostly consisting of the blue knobs halfway up the channel.  There are five different stages to the equalizer:  HI, HI MID, LO MID, LO, LO-CUT.  I’ll go into detail on them below.
  4. From there it goes to the FADER which is the main linear volume slider for mixing at the bottom of the channel.
  5. After that, the MUTE button.
  6. The PAN control just above the FADER distributes the signal in the left-to-right stereo mix.  For stereo sources you will want one channel panned hard-left and its companion channel hard-right.  For microphones you will usually want them somewhere in between.   When in doubt, centering the PAN knob is always a good choice, putting equal parts of the signal into the left and right output mixes.
  7. Finally the L/R MIX ASSIGN, a small white button, located at the very bottom of the channel next to the FADER controls whether or not the signal is in the mix.

As an aside, literally, the AUX SENDS shown on the diagram above allow the signal to be tapped off and sent to special outputs.  There are six of these auxiliary outputs and the red knobs near the top of the channel control how much signal is sent to each.  You will never use these in the production studio.   In the live music studio, SEND 1 is connected to a reverb effects unit.

Here’s what the controls look like on actual channels:




EQ (BLUE + buttons), MIX-B



Note the unity gain marks, labeled “U” alongside the FADER.  Position the slider centered on this mark for unity gain.  This will be your home base for each fader when mixing for broadcast or recording.

Next we have the procedure for setting up a channel for correct operation, from page one of the manual:

Channel setup procedure

So in light of the discussion at the top, what this page is saying is:

  1. These steps are done for you in the production studio.  In the live studio you have to make sure the MIC/LINE switch is in the correct position for the input you are using.  Usually you are using mics and it is already set correctly.  If are using a 1/4 inch phono plug line input plugged directly into the board (not the snake) you may have to change this switch.  The FLIP switch should always be UP in our studios.
  2. Start with the TRIM at minimum, AUX sends at minimum, the EQ disabled, the LO CUT set appropriately (both of these are covered below where we talk about the EQ section.)   Set the PAN control hard left, even if you won’t be using the channel that way in the mix.  Set the FADER to unity, meaning centered on the “U” alongside the slider.   Press the SOLO button located near the top of the FADER.  This will cause only this channel to appear in the left ear MONITOR and headphones.  The RUDE SOLO LIGHT over near the main meters will blink very brightly, alerting you to the fact that the board is not in the normal mode for mixing.
  3. Ask your guest to produce sound into the microphone or whatever source is connected to the channel.  Ask him to play, sing, or talk as loud as he is likely to get during the session.
  4. You see one of MAIN L/R output meters bouncing with the sound.
  5. Adjust the TRIM knob upward until the meter bounces at or just above the zero mark.   If it is a quiet source, you may have to take the TRIM quite a ways up.   If it is loud, not so much.  For most of the vocal mics we have, you will need to have the TRIM most of the way up.
  6. If you do subsequently apply significant equalization to the channel, you might need to adjust the TRIM once more.
  7. De-select SOLO on the channel.  Also should be mentioned here is to set the PAN control back to where it belongs.

The above procedure has the effect of making the signal nice and loud in the channel, but not so loud that it will distort.  It also ensures that the signal is well above the noise floor of the channel, so that if you do need to turn up the FADER higher than unity gain, you won’t be also putting a lot of noise in the mix.   The goal is that when you are mixing, you can start with every fader at unity and then adjust upward or downward from there.  This makes mixing much easier than if you have to start from zero and try to find the right level for each part.

The final topic tonight is EQ, short for equalization.  The manual gives a very detailed description of the EQ functions on pages 7 and 8 which is worth looking at while you try out different settings.  This is a great way to learn what kinds of sounds fall in which frequency ranges.  I preach a minimalist style when using EQ:  Rarely will I ever put an EQ level knob lower than ten o’clock or higher than two.

There are five types of EQ:

  • HI:  This boosts or cuts frequencies above 12 KHz.  This is often described as the “air” or “shimmer” in sound.  It is the brightest part of cymbals and violin.  It is the breath and mouth noises at the edges of a vocalists words.  Boost it a little to give the sound a more airy quality.  Cut it a bit if the sound is too bright and harsh.  This is a single light blue knob near the bottom of the EQ section.
  • HI-MID: Boosts or cuts mostly treble sounds but is very versatile.  This has three knobs:   The FREQUENCY controls the center pitch that is being boosted or cut.  The WIDTH controls how wide a range of pitches is affected.  Setting a large WIDTH makes this much like a treble control on a stereo — a broad range of upper pitches are affected.  Setting it narrow can emphasize or reduce particular elements of sound.  The LEVEL control sets how much to boost or cut.
  • LO-MID: Boosts or cuts sounds ranging from middle bass up to lower instrumental sounds.  This has a FREQUENCY and a LEVEL knob.
  • LO: Boosts or cuts deep bass sounds below 80 Hz.  This has only a LEVEL knob.
  • LO-CUT switch:  This is just a single switch at the bottom of the EQ section.  When engaged, it rolls off everything below 75 Hz.  This is intended to limit microphone handling noise, vibrations on the mic stand, footsteps, passing trains… low frequency rumble.  I recommend using the LO-CUT switch on ALL microphones unless the mic is on a true bass source such as a kick drum or an acoustic bass.

NOTE that apart from the LO-CUT, you must engage the switch labeled EQ IN, located near the bottom of the knobs, for the EQ to have any affect whatsoever.

Ok that’s everything for now.  The next topic will be microphones, on Tuesday March 2nd and Wednesday March 10th.  After that we will begin to focus on doing and making things in the studios.

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Digitizing Records

10 February, 2010 at 10:32 pm (Digitizing Records)

I’ve got over 1000 records.  Or what they call vinyl these days.  I’ve never really been an avid collector.  Mostly I have either been given or have bought other people’s collections when they “went digital” or moved or for other reasons.   I’ve got most of the rock and new wave I bought as a teenager and the stuff I bought as a WTJU DJ from 2001-2007.  I’ve got my mom’s collection of reggae, African music, Bob Dylan, and Curtis Mayfield.   I’ve got Angus’s Van Halen and Iron Butterfly records which he left at my apartment 15 years ago.  I’ve got a friend of a friend’s father’s academic electronic music.  Not sure if or how to give that back someday.  I’ve got my late mother-in-law’s Columbia record club items from the 30s 40s 50s.  And most of all I’ve got a 600+ piece collection I bought from Kai consisting mostly of late-90s house and techno.

Alphabetizing records while the snow falls

Alphabetizing records while the snow falls

That’s most of them.  It’s really cold in the basement so the alphabetizing project is kind of stalled right there.   Soon I’ll get it all sorted out and back on the shelves (including one not shown above.)  Then I’m going to go through it one letter at a time and pick out the goodies and eliminate the junk.

When is it time to throw a record in the trash and recycle the cover?  My criteria is:  Mass-market records widely available on CD (think Pink Floyd, for one.)  Or something that a true fan already has or wants on CD.  (Miles Davis, perhaps.  Though I would probably hang onto that record anyway unless it was in bad shape.)  The third category is just terrible music.  (Any number of DJ 12″ house or hip-hop records which may have been interesting once, when first issued, and you were already buzzed at the club and would dance to anything.)  Anything rare or unavailable on CD is worth digitizing, even if it is in poor condition.  If it is not your style, donate it to the library sale.   I haven’t yet figured out how to sell records online but I think it must be in my future.

Anyway I’m going to explain the equipment, methods, and software I’m using to transfer all this good stuff to digital files for burning to CD, etc.

There are any number of ways to do this.  My preference is to keep the analog path as simple as possible.   That means the minimum number of analog components between turntable cartridge and the analog-to-digital converter.  I’ve settled on the following setup:

My turntable is a Technics SL1210M3D, a variation of the venerable 1200 line:

Technics SL-1210M3D

Just like any self-respecting DJ I’ve got two of them.   These days I strongly prefer playing from laptop and CD and don’t take the turntables out anymore, so that means I can keep one in the cold basement music studio and one in the warm cozy living room.

For cartridges I’ve got Ortofon Concorde Pro S.  It looks cool and is very nice for back-cueing.  Probably fine for scratching too.  I’ll never know.  A “hifi” cartridge might be more appropriate but this sounds fine to me.

Ortofon Concorde Pro S

The turntable outputs are connected to a Rane PS1 phono pre-amplifier.  It appears that it might be discontinued but I’m sure Rane and several other manufacturers have high-quality phono preamps available on the market today.  One really nice thing about the PS1 is that it has two sets of outputs:  one with RCA jacks for normal use, plus another with balanced screw terminals which are buffered.  This allows a long cable run from the PS1 to your mixer or amplifier.  I use this in my basement music studio to get the clean amplified signal across the room from the turntable to the computer audio interface.

From there the signal goes into an analog-to-digital converter, as part of a computer audio interface box.  I have owned several over the years. Right now my favorite for portability is the Echo AudioFire2:

Echo Audio Fire 2

This is a Firewire device.  My main computer is a Apple MacBook Pro so that works for me.  If USB connectivity for Windows is what you need then I’d recommend the M-Audio Fast Track Pro 4×4.  M-Audio has some cheaper interfaces but this one is the baseline for me because I’d want to use it for DJing as well.   The key feature here is that the headphones output is a separate stereo pair from the main L/R analog outputs.  That makes it possible to cue music in the headphones while the program plays out the mains.

I think it is important to use a semi-pro audio interface like the ones described above.  If you use the built-in inputs on your computer or laptop you might get acceptable results.  I’ve never tried to use them for recording.  I have done A/B listening tests for playback comparing a PowerBook output with the Echo interface and the outboard interface was the clear winner.  Also I think that playing directly from an iPod sounds dreadful.  They use really cheap electronic components in laptops and iPods to keep the cost down, and you can easily hear the difference if you compare the sound to that from a CD player through the same set of PA speakers.

From there the audio interface is connected to the computer and you record, clean-up, and edit tracks.  After that you save and archive them to a lossless format and probably also to MP3 for everyday listening.   I’ll go into software tools in more detail in a future post.  For now I’ll just say that I’m using Audio Hijack Pro for recording, ClickRepair for cleanup, and Fission for editing.  I’d really like a better editor for the Mac because I grew up with Cool Edit Pro (which became Adobe Audition) on Windows.  It was a fantastic editor, had a nice built-in click removal tool, and had great help files which educate you about how to properly and carefully edit audio.  Adobe has not been good to this product in my opinion.

I’ll show the details and connections of the above-mentioned equipment in a future post.  Sometimes it looks like this:

Transferring records on a sunny winter day

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Session1: monitoring, headphones, speakers

27 January, 2010 at 4:13 am (Live Studio, Production Studio, WTJU)

Ok we had our introductory session for the Tuesday crew tonight.   Thanks to you all for coming.  Be sure to be aware of how the alternation is going to work.  This group will meet again Tuesday February 16th.  The intro session, documented below, will be given to the Wednesday folks on February 10th and so forth… I’ve spelled it out in the inaugural post for this blog.

Studio orientation:

Both the production studio and the live studio have a selection of input devices, which are wired to the mixing board (a Mackie 8Bus 24), and then routed to devices for monitoring, recording, and/or broadcast.  So for example, when a CD is dropped into one of the players, say CD2, and then you bring up the pair of faders for CD2, you hear music on the speakers.

Here’s the lower section of the channels including the faders for CD2 in the production studio:

CD2 channel faders

Things to notice about the channels:

  • The PAN knobs at the top are set hard-left and hard-right.  This is to ensure that the CD goes into the mix in stereo.  (Normally the PAN knob positions a sound somewhere in the left-to right stereo field.  In this case we want the left channel from the CD player to appear only in the left channel of the output mix.  Ditto for the right channel.)
  • The channel MUTE buttons are not depressed.  The yellow light above each is not lit.
  • The L/R MIX buttons at the bottom next to the faders are depressed.
  • No other buttons in this view are depressed.

Getting the microphone into the mix is also straightforward.  On that picture above, channel 1 is the main mic in the production studio.   Here we see the channels for the mic and CD player in the live music studio:

Mic and CD channels on live mix board

Note that in the live music studio, the main mic is labeled DJ MIC and it is channel 9.  The same conditions for getting it into the mix apply in both studios.

Monitoring options:

Whichever channels are set up as above will appear in the mix.  And “the mix” is called L/R MIX.  There are three monitoring output options for the contents of L/R MIX:

  • the studio speakers, known as CNTRL ROOM
  • the HEADPHONES 1 output
  • the HEADPHONES 2 output

Recording options:

In the production studio, the contents of the L/R MIX may be recorded to the CD burner in the rack to the right of the mixer.  It may also be recorded into the production editing computer.   You might also want to record to a reel-to-reel tape recorder or the DAT deck.  But probably not.

In the live music studio there is a CD burner which has a history of being rather unreliable.  I’ve used it sometimes to record a live music performance as an emergency backup copy.  But as often as not it hasn’t worked.  Perhaps in the future this capability will be restored so you should be aware of the possibility.

The monitoring mixer:

On both mixing boards we find this area, about halfway up and to the right edge of the board:
Monitoring mixer, production studio

Monitoring mixer, production studio

This is a Mixer! Actually it is a cascade of one mixer called MONITOR which feeds into two more mixers called PHONES 1 and PHONES 2.  Normally we think of mixers as having knobs or faders to control how much of each signal ends up in the target mix.  In these mixers there are just push-buttons.  Either a given signal is in the mix or it is not.  We shall next see which signals we might desire in our monitor mix.

Here’s an annotated image:

Monitors mixer drawing

I’ve highlighted the important controls in the monitoring mixer section.   In the rightward section labeled MONITOR:

  • The CNTRL RM knob (green) controls the volume to the speakers.
  • The L/R MIX button (red) selects the main mix from the channels and faders to go to the speakers.
  • The EXTERNAL button (also red) selects the on-the-air signal to go to the speakers.  This is an actual radio receiver with an antenna on the roof.  It is the same as feeds the Air Monitor signal in the broadcast studio.
  • The 2nd and 4th buttons (MIX-B and 2-TK) are rarely or never used.   MIX-B has an advanced usage that comes up when mixing loud live music.
  • The MONO button (yellow) forces the sound sources selected above to be mixed down to monophonic.   Generally you will want to check that this is switched off so that you can hear your mix in stereo.
  • It is possible to have multiple source buttons selected simultaneously.  This can cause confusion.

I’ve just described this mixer as controlling what you hear on the speakers.   However it has a more fundamental function:  The output of this mixer becomes a stereo signal called MONITOR.  This signal appears in each of the two headphone mixers to the left.  Some key points about the headphones mixers:

  • The PHONES LEVEL knob (green) controls how loud the headphones are.  Always check that your guest’s phones level are comfortable.
  • The MONITOR button (red) in PHONES 1 or PHONES 2 causes whatever is selected in the MONITOR mixer described just above to be heard in the respective headphones.
  • The EXTERNAL button (also red) in PHONES 1 or PHONES 2 causes the on-air signal to go to the respective headphones.
  • The 2nd, 3rd, and 4th buttons in the PHONES mixers are rarely or never used.
  • It is possible to have multiple source buttons selected in one of the PHONES mixers simultaneously.  This can cause confusion.
  • It is possible to route EXTERNAL to the headphones through two different pathways if it is also selected in the MONITOR section.  This will make it sound very strange in the headphones causing severe confusion.

Headphones outputs:

Here’s where you plug in the headphones at the top-right corner of the boards:

Headphones outputs, production studio

In the production studio (above), the phones are plugged directly into the PHONES 1 and PHONES 2 output jacks.

Headphones outputs, live music studio

In the live music studio, PHONES 1 is for the engineer operating the desk.   PHONES 2 is routed over to the 6-channel headphone amplifier, mounted in the rack to the left of the mixing board:

6-channel headphone amplifier (top)

So the PHONES 2 signal appears on all six of these outputs.   Each output has a separate volume control.   The pictures above are the approximately correct setup for the live music studio:

  • In the PHONES 2 mixer the PHONES LEVEL knob should be at about 12 o’clock.
  • On the 6-channel headphone amplifier, the INPUT LEVEL at the left should be at about 9 o’clock.
  • Individual guests may want their levels higher or lower, but the VOLUME will generally be somewhere around 12 o’clock.

Channels 3, 4, 5, and 6 out of the headphone mixer are fed to jacks across the top of the snake breakout box on the floor:

live music studio snake breakout box

A final word on headphones:

When you are doing anything which requires detailed attention to sound and music, your most important tool is your headphones.  Every engineer or producer really should own his own pair of decent quality closed-back headphones which you bring with you to every session.  Closed-back phones are a must because they prevent room sound from leaking into what you hear.  Most of the cheap phones around the station are not closed-back.   It is wise to bring them to every broadcast you do as a DJ as well.

The more stuff you hear through your own cans the better judge of sound quality you will be.  Try listening to some of your favorite recordings, “that track” with which you are intimately familiar, on a few different headphones and speaker sets and you will appreciate how subjective “good sound” can be.  Having always at the ready your “reference” headphones will go a long way towards teaching you how to listen for the qualities you want to hear in your work.

Here are some suggestions, by no means comprehensive:

  • I’m partial to the Pioneer HDJs right now.  Designed for club DJs they are loud with nice bass.  They suffer in the high frequencies when compared to true hi-fi sets.
  • Before that I had a pair of Sony MDR-V700 which worked great for years.   Other folks at the station have these even now.  Same comments apply as with the HDJs.
  • There are undoubtedly great offerings in many price ranges from AKG, Sennheiser, and Audio-Technica.
  • Read reviews before buying and try to find a way to try them out first with “that track” and several others like it to make sure.

That’s it for tonight.  Please comment below or email me with any questions.   I’m still figuring out this blog thing so this page will be shifting and bending to my will over the next days so be sure to check back.

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Audio production + engineering series at WTJU

24 January, 2010 at 4:29 am (WTJU)

Ok so this is the new blog.  I’m Morgan McLeod, production volunteer at WTJU since 2000.  I’ve been making music and mixing live bands since the ’80s,  recording and editing audio on the computer since the ’90s. I encourage all of you to comment here, email me, or otherwise get engaged with this little thing called audio production. Now…

This is the new training series.  I’m dedicated to promoting enthusiastic generalists at WTJU.  That is, volunteers who understand how all the studios work;   Who are eager to step up when there is a band/interview/promo opportunity;  Who are not afraid of this mere tangle of wires we call ‘the studios.’   The studios are more similar than they are different:  they have the same mixing board (Mackie 24.8 bus), the same microphones are often used, the signal labels and meanings are consistent.

The calendar is below.  It will be every week until summer, alternating between Tuesdays one week and Wednesdays the next.  There is a gap next week because I’ll be away.  Every session will be documented on this blog and questions and comments are most welcome.

* 1/26 & 2/10: Monitoring/headphones/speakers – general orientation to how to control what you are listening to,  how sound goes into and out of the mixer.  Fundamental but critical stuff.

* 2/16 & 2/24: The mixer channel – how to get good sound into the mix.

* 3/2 & 3/10: Microphones: RE-20 and SM-58.  The most common mics in radio and in the world.

* 3/16 & 3/24: Mixing/recording/broadcast:  Putting it all together to get sound to your chosen destination.

* 3/30 & 4/7: Adobe Audition:  Basic use of the audio recording, editing, and mixing software we use.  Indeed the most commonly used software in radio.

* 4/13 & 4/21: TBD… advanced topics include phone interviews, audio effects, live loud music, live quiet music.  We’ll see where it goes from here for the rest of the spring.

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