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Running Multitrack Audio Support LIVE with SoundTrack Pro

This past weekend I had the privilege of running the FOH position for “Let There Be Light”, the Christmas 2010 program at Crossroads Community Church in Kokomo, IN. This particular production was very complex and advanced for a church this size and at their stage of production development. Here’s a quick overview of the basics:

  • live drumkit in a fishbowl (isolation plexi with absorption lid and back) – 8 mics
  • Bass gtr, two electric gtrs, and two acoustic guitars rounded out the rhythm section (electric guitars were run 30′ offstage for isolation and stage volume reduction)
  • Live grand piano with both a pickup and pzm mic (2 lines), and a synth player
  • 10 piece orchestra (4 tpts, euphonium, 2 saxes, 2 clarinets, and flute)
  • 40 voice live choir
  • 5 lead vox
  • 4 drama/speaking wireless
  • stereo video and CD feeds round out the normal requirements
  • Rhythm section and lead vox were all in-ear buds via Aviom mixers
  • Choir Director and Orchestra Director each had an Aviom mixer that fed floor wedges.

OK. With the background info out of the way, it’s now for the subject of this article – running multitrack audio support using SoundTrack Pro.

Early in the preproduction process, I made the case that anytime you want a choir and full rhythm section simultaneously you’ll be disappointed with the lack of volume and clarity from the choir mics. Even when you go to exremes of all IEMs, drums in a fishbowl, and guitar amps offstage, you will still not be able to get sufficient volume out of a live choir – especially one that is less than 50 persons and is made up of church volunteers (which usually means less than half of them will be really great vocalists who can project properly). This assessment matched what the production team had experienced in previous productions in years past.

I suggested that we pre-track the choir so that we could supplement the sound of the live choir with the pre-recorded voices. Our team at Oakbrook Church had used this technique a few times previously with great success. Crossroads had never attempted anything on this scale before, but were open to the possibilities. After a series of meetings, including a couple with the Choir Director, the decision was made to go for it. The Worship Pastor then spent a couple of weeks getting things ready (making background loops and clicktracks).

We blocked out a Saturday and scheduled each choir section (Soprano, Alto, Tenor, and Bass) in 2 1/2 hour blocks. We then recorded each of the sections on all 8 songs using the Worship Pastor’s MacBook with ProTools and 2 Shure KSM44 microphones. At the end of that recording day we had our pre-tracked choir! We had to do a little post-sweetening (gating, combining different takes, blending, etc.), and then exported the files from ProTools to disc. On a different day, we also recorded some narrator voiceover tracks which would be needed for the production.

Then it was time to set up SoundTrack Pro. A few of my friends questioned this choice as a playback medium, but I was confident that it would work. The choice to use STP was kind of made for us. The Worship Pastor’s computer had ProTools, but PT only works with Avid-approved hardware, and none of us had any PT interface with enough outputs. But I did have a MOTU 896 firewire interface in my aresenal (though I didn’t have any current versions of AudioDesk or Digital Performer to use with it.) We considered GarageBand briefly (which would have played back all the tracks) but it would not output in anything other than stereo. So SoundTrack Pro was chosen mainly because it was the only multitrack audio app we had that would utilize the output capabilities of my MOTU interface.

The MOTU 896 interface sitting beneath my MacBook Pro running SoundTrack Pro.

One bonus feature of using my 896 in this application is that the main FOH console is a Yamaha DM2000 – a digital board. So we were able to use the 896’s ADAT optical digital output right into the Yamaha. (We had to run a Wordclock BNC cable from the Yamaha back to the MOTU to make this work right. This made sure the two digital clocks in the two devices were synchronized).

We ended up with a total of 7 tracks (more than could have been accomplished had we used DVD Dolby 5.1 as a playback medium). The tracks were:

  1. Soprano
  2. Alto
  3. Tenor
  4. Bass
  5. Tracks
  6. Click
  7. Narration

When setting up the channel layout, I ran the pre-recorded tracks alongside the three live choir mics so they were all together. In the picture below, you can see the mixer channels. chrL stands for Choir Left, chrC for Choir Center, etc.

Back to SoundTrack Pro. Because the transitions happened very fast in this program, and because the Narration often happened in between songs, I decided to make the entire program one long song. Here is a screenshot of the SoundTrack Pro window:

A SoundTrack Pro screenshot. The timeline moves from left to right. The tracks from top to bottom are: Track, Click, Soprano, Alto, Tenor, Bass, Narration.

On the Yamaha console, I combined all choir mics (live and tracked) to one common Subgroup. This way I had an overall level for the entire choir:

8 Subgroups on the Yamaha DM2000: B/K (bass/kick), Perc (all other drums), Gtrs, Keys, Brass, Choir, Lead Vox, and BGVs

And everything went off without a hitch! My MacBookPro never crashed or froze throughout an entire week of rehearsals and 3 performances. Here’s the “money shot” of the program:


Mixing Two Services at Different SPLs

At one of the churches I’m working with, the tale is all too common. It is a large multigenerational congregation with a lot of history. As such, the attendance trend is for an older crowd to populate the early service, with the later service being dominated by younger families. This is not surprising as the same thing is probably happening all across the country.

One thing that can’t be avoided is the SPL preference of these two unique groups. I use the word preference pretty lightly. It’s stronger than that. The older generation becomes pretty vocal in their displeasure when the volume gets too much for them. Conversely, for some in the second service the volume never gets loud enough. If they don’t feel the Kick busting them in the chest, they’re not happy!

So what does the sound tech do? There isn’t time to rebuild the mix between services. A simple Master Fader change doesn’t work. That makes everything lose its presence, including vocals. Yet we don’t want to get crazy with changing instrument faders or we will upset the Pyramid mixing technique.

So we experimented this week. Here is our test method:

  1. During the first half of rehearsal, we build a normal Pyramid Mix as we normally would. We start with the drums and bass, mixing them at the appropriate level for the acoustic volume of the drums in the space. Then move up the pyramid until we achieve a great mix.
  2. Then for the last half of rehearsal, we reduce the master fader 6 dB.
  3. Then at the subgroups, we pull up the vocals and piano groups 3 dB.
  4. We ask the drummer to “ease up” on the first service.

So we have effectively “halved” (-6 dB) the apparent volume of the entire mix, while compensating the vocals and piano a bit. Now for 2nd service, we only have to adjust 3 or 4 faders ( Groups 5, 7, 8 back down 3 dB – and then Master Fader back up to its original position). Voila – we’re back where we need to be for 2nd service.

Some would suggest that we do the opposite (pull down drums, bass, guitars, keys, etc.) But for our situation, our method is the least amount of fader changes.

Have you done something similar? How did it work?

Using The Various Types of EQ

You can use EQ to do a lot of things. Some are very basic and some are kind of exotic. In this post, we will focus on the most typical uses for the different kinds of EQ in a Pro-Audio / Live Sound environment. Also keep in mind that as we progress into the digital age, these EQs may not be separate pieces of hardware in an equipment rack. They may be options in a menu in a system processor, digital console, effects unit, or computer plugin. But the function and purpose is exactly the same.



  1. Crossover Filters and System Processor settings
  2. Channel Strip HPF
  3. Driver (speaker) protection
  1. Channel Strip EQ (ie: 3 or 4 band EQ)
  2. Equal Loudness Curve Correction see Fletcher-Munson wiki article
  3. Tone correction on instrument
    amplifiers and preamps



  1. Ringing Out Monitors (feedback control)
  2. System Toning see Tuning vs. Toning article by Bob McCarthy
  3. as a Channel Insert for more control
  4. in a compressor’s Sidechain for use as a De-esser
  1. System Tuning (a.k.a the Room EQ)
  2. Channel Strip EQ (on more advanced consoles)
  3. Feedback Control in monitors or as a channel insert



  1. Tight Feedback control for monitors
  2. as a channel insert for precise control on a specific input
  3. as a Group Insert for precise control on a group of inputs (lavalier mics, for example)

  1. an extra level of feedback protection in a fast-paced, high stage turnover environment (such as a festival or open-mic night)
  2. as a channel insert on a specific
  3. as a Group Insert for precise
    control on a group of inputs (lavalier mics, for example)

Caution should be exercised where it concerns auto feedback busters. There are several concerns:

  1. The more inexpensive models (Behringer) and some of the earlier versions of the technology (Sabine) can truly do some horrific things to the sound. Yes, they may actually kill some feedback, but what else does it kill? Your frequency response, good tone, and phase response for starters!
  2. Just because they are so-called “smart” devices, that doesn’t mean they are. Do they know the difference between feedback and certain legitimate musical sounds? NO. In fact, any source that has a pure, steady fundamental pitch without a lot of harmonics is going to get cut by the device.
  3. They can do a lot of things without you being aware of it. Because they auto-sense and auto-engage filters, they can do things when you’re not paying attention. And why would you pay attention? There’s probably a million other things going on at the stage that need your attention!

So because they can be fairly destructive, you should be cautious with them. I know a lot of the Pro Audio world just outright bans them from the premises. Seriously! If you do feel you need one, follow these couple of rules. First, stick to the really good ones. The Shure units and the Peavey Ferrets are particular standouts. Second, use them primarily on speaking mic inputs or groups – they seem to do a really good job there. Try to avoid using them on master monitor sends, and absolutely refuse to use them on the House Mix. That’s just the wrong tool for the job.

An EQ Primer

There are a few different types of equalization. This post is an overview of them. In part two of this topic, I will talk about the specific uses for these types in a Live Audio setting.


  1. Bandpass. This is the method used in crossovers, High-Pass Filters and Lo-Pass Filters. Full bandwidth material enters the circuit, and at a given frequency, the material is processed separately. In a HPF set to 80 Hz, for example, the material below 80 Hz will be lowered dramatically (typical would be 12 dB per octave), while material above 80 Hz will not be processed at all.
  2. Shelving. This is similar to bandpass, but it is variable and can be adjusted for boosting as well as attenuation. The simplest way to think of shelving EQ is your stereo system’s Bass and Treble knobs.
  3. Graphic. A common EQ type which can be found not only in pro audio, but in car audio, home theater, and computer applications. It could be as simple as 3 filters (what an average Joe might call “sliders”), or it could be as complex as 31 filters. Depending on how many filters there are, the manufacturers use different names for them. An EQ with 15 filters (also called 15-band EQ) is often described as a 2/3 Octave. This is because each individual filter affects 2/3 of an octave of frequencies. A 31-band EQ is also known as a 1/3 Octave. If you are a musician, imagine a filter that is centered on middle C. If you decrease that filter, you would not only affect C, but Bb through D as well.
  4. Parametric. A much more precise tool, a parametric EQ allows you to manually select the exact frequency you wish to change, how wide a bandwidth you want to affect, and the level of change you want to apply. With a parametric, you can select a slice of the frequency spectrum as narrow as 1/10 of an Octave or more.
  5. Notch Filters/Feedback Supressors. Sometimes going as narrow as 1/30th of an Octave or more, these very precise tools are used primarily for eliminating feedback from a system while trying not to damage the frequency response of the overall system. In recent years all the manual models have been discontinued and the feature has been included in system processors. But there are also many “automatic” feedback detection devices which claim to be able to set themselves. Your mileage may vary :-)
  6. Software-only Special EQs. In the world of digital audio and mastering, there are a couple of newer types of EQ – specifically the Paragraphic and Dynamic EQ. Neither of these types really exist as a piece of hardware, but they can be useful tools in the digital realm. *CORRECTION: The BBE Sonic Maximizer appears to be a Dynamic EQ, which applies what is essentially a Fletcher-Munson Curve dynamically (meaning the louder the signal, the more EQ is applied. This is how they disguise the hiss from the boosted high frequency EQ. When the level drops – so does the correction.)

My “Bass Player’s Compression Dilemma”

I am a bass player. I play a passive bass (one where the pickups do not require a battery inside the guitar). I love the tone of them – I love the low maintenance aspect of not having to wonder if I accidentally left the guitar plugged in and my battery might be drained.

But I’m not a rich man. I do not own a Lakland, or other high-end bass. My 5-string Yamaha cost me about $450 new several years ago. As such, the low B-string is a little “flabby” sounding – it doesn’t have a real strong fundamental tone down there. In order to correct that issue, I’ve been able to use compression. Compression is generally needed on all bass guitars because the volume tends to jump up and down a lot depending on the note, the fret position and the pickup selection. But with a rather inexpensive 5-string bass with a flabby B string, good compression is an absolutely CRITICAL part of my signal chain. [Here’s a quick video of my playing with the bass so you can hear what it sounds like with compression on the front end]

But now comes the dilemma – part 1: Do I totally depend on the sound guy of whatever venue I’m playing to instinctively know I have Flabby B Syndrome and assume he has the gear and knowledge to fix it? I’m not a fan of that option. How do I know he isn’t using too much or too little compression and completely desroying my dynamic control? No – I’d much rather have control of the compression on stage with me so I can set it correctly, monitor what it is doing, and can actually hear the results live.

Now for the dilemma – part 2: 99% of all stomp-box type compressors STINK for bass. They are noisy. They pump and breathe (an audible, non-pleasing compression side effect). How about the other 1% of them that are decent? They’re EXPENSIVE! Like $185 expensive. No thank you. If I’m going to pay that much for a compressor, then it had better be recording studio grade, baby. Come to think of it – that’s what I’m really looking for anyway – a studio-grade compressor for my bass rig.

So here comes dilemma – part 3: Studio-grade compressors are made for Line Level usage. As in +4 dBm (1.23 volts). These compressors want to see that hot of a signal coming into them. But wait! Instruments – especially passive ones – don’t put out even CLOSE to that amount of signal. Another issue is that of pickup loading. If the bass pickups do not sense the correct high-impedance load connected to them, then they will not output at their maximum capability and will lose some frequency response and level. So directly connecting my bass right into a studio-grade compressor is not really a viable option. There would also be the problem of monitoring. How would I connect my amp? The output of the compressor would be line level – and my amp’s input will be expecting instrument level! ARGH!

But here’s where I found the magic answer. An older dbx 163x compressor was EXACTLY what I was looking for. It has some unique features that fit my purpose wonderfully. First, it has two different inputs – a front and a rear. The rear input is line level and works like most high quality compressor inputs do. But the front input is High Impedance – SPECIFICALLY MADE for instrument level. The unit actually has a little preamp built into this input, so it even loads my pickups correctly! As for output, it has a very easy to adjust output level – so I can send out instrument level to an amp or direct box!

They don’t make these any more, but you can still find them on eBay and such. I was able to purchase two of them for $80! I now have exactly what I was looking for – studio grade compression with on stage control – and it wasn’t expensive!

The Order of Input Channels

There are lots of theories on channel layout. Some board operators prefer vocals first, followed by the band. Some use the opposite scenario, with the band first and then vocals. And some don’t give much thought to the process, just plugging things in order as they come from the stage (Stage Pocket 1, 2, then 3, etc.) regardless of what instrument is represented on that line.

Over the years, I have advised my clients and friends to generally adopt the “Pyramid Mix” channel order. For those of you not familar with the Pyramid Mixing technique, here is a diagram (created by Curt Taipale) which explains the concept:

So how do you build a pyramid? You start with the bottom layer and work your way up. I do modify it slightly, in the sense that I want to keep like instruments together (guitars, keys, drums, etc.). Also, in most settings I’ve worked with in the last several years, guitars are dominant with keys being primarily ambience (organs, pads, etc.) – so the pyramid is slightly modified there. It’s also helpful if you lay out the channel order with regard to your Groups. The typical large format board has 8 subgroups, so I take that into consideration as well. So here is the general layout I recommend to most clients:

01 – Bass Gtr
02 – Kick Drum
03 – Snare Drum
04 – Tom 1
05 – Tom 2
06 – Tom 3
07 – Hi Hat
08 – Crash (OHL)
09 – Ride (OHR)
10 – Percussion 1
11 – Percussion 2
12 –
13 – EG 1
14 – EG2
15 – AG 1
16 – AG 2
17 –
18 – Piano L
19 – Piano R
20 – Synth L
21 – Synth R
22 – Track
23 – Click
24 –
25 – Lead Vox (or Worship Leader)
26 – BGV 1
27 – BGV 2
28 – BGV 3
29 – BGV 4
30 –
31 – Anouncement Mic
32 – Pastor Mic

Notice there are some blank channels in between the groupings. This is for that last-minute request that always seems to come, ie: “Bob is going to be using 2 amps today”, or “I fogot to tell you we’ll need 3 acoustic channels this week”, or “We’re using the Djembe today so we need 3 percussion mics.”

Of course, this order can be scaled up or down depending your exact situation, but now let’s look at the group situation. Here’s a look at the typical 8-group layout:

Group 1 – Bass/Kick
Group 2 – Drums (all drums/percussion except Kick)
Group 3 – Electric Gtrs
Group 4 – Acoustic Gtrs
Group 5 – Keys
Group 6 – Misc (could be brass, choir, lavs, whatever)
Group 7 – Lead Vox
Group 8 – BGVs

You may have to make some modifications if you are running true stereo. Group 1-2 may be Stereo Drums. Group 5-6 may be Stereo Keys, etc.

If you only have 4 groups, it might look like this:

Group 1 – Drums/Bass
Group 2 – Gtrs
Group 3 – Keys
Group 4 – Vocals

I realize this specific layout may not work for everyone, but it has proven to work in many situations I’ve been involved with over the last several years. Whatever layout you choose – it should make sense and should allow you to mix properly without having to go hunt for something.

Amplifying an Acoustic Guitar

Today’s topic is acoustic guitars in an amplified band setting. The acoustic guitar is a common component of today’s pop/rock sound – and there are many options and considerations.

Type of Transducer

There are several different types of pickups and microphones to choose from. Here is a basic overview.

Soundhole Magnetic pickup

Pros: It’s easy, inexpensive, and can give decent results very quickly.
Cons: Because it is a magnetic pickup (like an electric guitar pickup), its sound is a little more dark and electronic sounding. Also, you have to deal with the annoying cable coming out of your soundhole.
Connection to PA: a direct box, external pre-amp, or acoustic amplifier

Piezo Under-Saddle pickups

The most popular method so everyone generally understands it. A clever way to get a lot of sound without causing a visual distraction. Very good at capturing the dynamics of the instrument. Not that expensive. Many models come with an included preamp.
The raw unamplified sound from these pickups is harsh, brassy, or banjo-esque. They need quality preamps in order to add the warmth needed to balance the sound out. The better the preamp, the better the resulting sound.
Connection to PA:
a direct box, external pre-amp, or acoustic amplifier. You can always tell if the pickup is “active” (includes a preamp) if the instrument requires a battery. A battery always indicates the presence of a preamp.

Contact Transducers

Cheap and easy.
Sound quality is noticeably lower, requires adhesives (foam tape) to stay in place.
Connection to PA:
a direct box, external pre-amp, or acoustic amplifier

Taylor “ES” system

Uses a unique combination of a built-under-fretboard pickup and dynamic “body sensors” (pictured) which attempt to amplify the body characteristics in addition to the string sound. Great for percussive body-slap type players. It does produce a slightly more “natural” sound than piezo systems. Good at rejecting feedback. Balanced output.
Only included on expensive Taylor guitar models. Many players do not like the sound. Often requires drastic EQ. Uses special cabling for best results, which make it cumbersome to tune on-stage or use effects pedals.
Connection to PA:
because of its balanced output and special cable, it can be plugged directly into a console or snake. Or it can be run with a standard cable to standard connection options, but with a slightly noticeable drop in sound quality.

LR Baggs I-Beam system

Pros: A unique system that is placed inside the guitar, underneath the bridge plate, attaching with adhesive. Gives a very natural acoustic sound which many players prefer over typical under-saddle systems.
Cons: a little more expensive than its piezo under-saddle counterparts. Requires some finesse on installation to find the “sweet spot”.
Connection to PA: a direct box, external pre-amp, or acoustic amplifier

Internal Microphones

Pros: Can give a more natural, high quality sound and image, especially in classical and traditional fingerstyle formats.
Cons: But the quality comes with a price – high feedback probability in loud stage environments (ie: drums or brass). The fact that the microphone is inside the guitar cavity means the mic will be prone to swelling tones and sympathetic resonance.
Connection to PA: most systems come with an included preamp, which allows the instrument to connect directly with a sound system or guitar amp.

Combo Systems
Pros: Blending the best of both worlds – the articulation and feedback rejection of a piezo system combined with the natural warmth and tone of an internal microphone. These systems usually include the ability to blend the two sources any way you want – giving you the best of both options.
Cons: Can be pretty pricey and can require extensive modification to the guitar (not for vintage instruments!)
Connection to PA: sometimes requires two connections – a DI for the pickup and an XLR for the mic.

External Microphones

Pros: Very high sound quality, reproducing the beauty of the instrument.
Cons: Very susceptible to feedback if put in the monitors at a high level. Not enough separation from other instruments (the mic will also pickup drums, bass, brass, monitors, stage noise, etc.).
Connection to PA: standard microphone cables.

Types of PreAmps

After we have our pickup system, the next thing to review is the external preamp options (if needed). There are basically three grades of these:

  1. Music Store preamp boxes. These are the most common and can give very good results right out of the box for around $150. Two popular units are the L.R. Baggs Para DI and the Fishman Pro EQ.
  2. Boutique boxes. These are targeted more at the discerning professional, and can range from $500 and up. One popular choice is the Avalon U5.
  3. Special Purpose Boxes. There are a myriad of other options out there. One pretty common device in the Taylor community is the Taylor K4.

Types of Monitoring

Along with the on-board and external electronics, the next thing to consider is how the guitarist will hear his amplified sound. In a loud stage environment with drums and electric guitar amps, there are 3 main options:

An acoustic guitar amplifier.

There are many good makes and models of acoustic guitar amps now. In a shared-mix environment where the guitarist is sharing a monitor mix with others, this is probably the easiest solution. The signal chain goes like this – from the guitar’s internal or external preamp into a high quality Direct Box. The Direct Box provides an XLR connection to the mixing console. But it also has a parallel input or pass-through that you can use to connect to the amplifier. Doing it this way means the guitarist can adjust the volume and EQ setting on his amp without affecting the signal that goes to the board. NOTE: There is a big difference between a typical electric guitar amp and a nice acoustic amp. The acoustic version is made for full frequency response and often includes a high frequency horn, whereas the electric guitar will roll of high frequencies dramatically.

A floor wedge monitor with a dedicated mix.

The dedicated mix would most likely be an unused aux send from the mixing console. This allows the guitarist to have his own blend of instrument and mic he wishes to hear, along with a healthy dose of himself.

In Ear Monitors with a dedicated mix.

This solution ensures that the guitarist can hear as much of himself as he wants, and will be isolated from other loud sounds on stage. It is an elegant solution, but comes with a significant upfront investment.


Audio Training for Live Sound. All content ©2010 Jeremy Carter Consulting.

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