Monitors – Board Unity Gain vs Amplifier Output Gain

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Today’s post is regarding monitors.  At North Ridge Community Church, NRCC, we run our three separate monitor mixes from the front of house.  These monitor mixes are fed via the pre-fader aux sends, go through a graphic equalizer, amplified by a Crest Audio amplifier and then show up to our JBL floor wedges.

This is a very normal way to do monitor mixes and all is great, until… I discover how it is truly set up.

  • The amplifiers in the amp room are all turned up to full.
  • The output of the board aux sends are not full signal.  In fact they don’t have enough signal to light up the -21dB light on my equalizer.
  • 100-105dB volume normally output by monitors.
  • AC hum present when system is turned on.

Because the are turned up so loud, the aux send knobs are in the 9 o’clock to 11 o’clock range.  While this doesn’t sound that bad, on the Allen & Heath ML4000 the unity gain (+0dB) for aux sends is at 3 o’clock and the knobs are a logarithmic potentiometer.  They act just as a normal fader does on the sound board, where there is more resolution in the higher areas of the fader.  Basically if you move the aux send knob from 9 o’clock to 10 o’clock you would increase the volume by 10dB.  If you were to move the knob from the 2 o’clock to the 3 o’clock position you would increase the volume by 1-2dB.

Found at http://www.bigmuffpage.com is a perfect graphic for explaining the logarithmic taper of the potentiometers that are used in aux sends.

Another thing that is happening by having such a low output from the board and the amps turned up high is easily explained in the graphic below that I made:

By having the board turned down and the amplifer turned up to full, you have made your signal to noise ratio very small. But when changed to a full board output and to a lower gain on the output of the amplifier you will have a high signal to noise ratio giving you less noise in the monitor system and better clarity.

As shown in the above graphic, by having the board aux output turned down and the amplifer turned up to full, you have made your signal to noise ratio very small. By keeping it this way you amplify the noise along with the small signal sent from the board.  This makes for a noisy system. But when changed to a full board output and a lower gain on the output of the amplifier you will have a high signal to noise ratio giving you less noise in the monitor system and better clarity.

By keeping your board output in a unity gain region you have higher resolution on the aux knob giving you an easier time mixing the monitors.  Also this leads to when you do an AFL (after fader listen solo) you don’t need to turn your headphones up to hear it.

After turning the amplifiers down and turning the board up I was able to keep the same overall volume with the monitors but now there is no audible noise in the room from the system being on.  If you find yourself having a hard time getting a consistent monitor mix or a noisy system, you might want to double check your amplifiers output gains to make sure they are receiving a unity gain signal.

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Polarity issues in CD recording from bad wiring

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Balanced audio signals are a great thing for the audio industry. If you don’t know much about balanced vs. unbalanced signals, I highly suggest doing some reading on Wikipedia.

Balanced signals have a differential amplifier on the output and the input side. They basically carry a mono signal down 3 conductors. Two wires carry the signal and then the last wire carries the ground. The beauty of the differential amplifier is that two wires that carry the signal. One wire carries the normal signal, then the other inverts the polarity. So, the two signals are out of polarity of each other. At the receiving end of the balanced signal, there is another differential amplifier which inverts one of the wires and sums them together to get the signal back to normal.

If there is any noise injected into the wire on its run, it is injected into both wires in polarity. When the signal reaches the receiving end of the wire, that differential amplifier inverts the polarity of one of the wires which cancels out the noise. This is because the noise is now out of polarity with each other and, in the summing process, cancels out itself.

Illustration from http://www.ians-net.co.uk/images/articles/balanced/balanced.gif showing how noise is rejected in a balanced signal.

At North Ridge Community Church, NRCC, we have a Tascam CD recorder which we use to record the spoken word of our services. I noticed one day that I had to turn up the CD really loud in my car to be able to hear it. This surprised me because earlier in the day I was almost clipping the meters with the CD recording levels. I decided to rip the CD into Cubase 5 to see what was going on. Here is what I found:

Out of Polarity signals which were recorded to the CD like this.

As you can see, the left and right channels are recorded out of polarity to each other. Basically, they are cancelling each other out from the inversion.

After seeing this on the screen, I went back to the church to find that the CD recording feed is coming from a balanced mono send off of a matrix on the Allen & Heath ML4000. The cable then goes into a 1/4 inch stereo to R and L RCA adapter. The way this adapter works is for a headphone cable which takes the left from the tip and sleeve and the right from the ring and sleeve. With a balanced signal, the tip and ring are inverted in polarity. So this is sending the + into the left while sending the – into the right.

I recorded a video of myself explaining what is going on when you have a signal that is out of polarity. In the video, I accidentally mention that this could be called 180 degs out of phase which is incorrect. The only term to call this is out of polarity.

After making the correct type of adapter with the balanced to RCA, the recordings will now have full volume on the CD recording with a full spectrum sound. This is another reason not to trust any installed wires without checking them.

Drew Brashler

“Vocals are the most important, so I’ll start with the Drums”

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I heard this saying a long time ago and I love it and still say it to this day.  “The vocals are the most important part of my mix, so I will start with the drums.”  The way you approach mixing may differ from mine, but I think of audio as a triangle.  Just like the food industry thought up the food group triangle, I am going to make one for you here for audio.

First and foremost, we have the Drums & Bass at the bottom of the triangle making a very good – sorry for the pun – base.  Next the keyboard and piano would go, some of the other backing instruments for fills could either go in this level or the next up.  Then acoustic & electric guitar on the next step up.  Then at the very top there is the vocals, which is what most listeners are primarily hearing.  When someone walks away from a show, not many people (excluding drummers and sound guys) will say “that is the best kick drum I have ever heard” most of the listeners will say “man xyz vocalist was great today.  The idea behind this is if you have a bad mix on any one of these levels the one above it will not be supported correctly.

Enough theory; lets get into some changes.  At North Ridge Community Church, NRCC, we have an acoustic Pearl Session Custom drum kit, in front of that we have a plexy drum shield which helps a little for the front row of our worship center.

Next come the drum microphones… this is where it is lacking big time.  We had some off brand microphones that were made by Audio-Technica.  There is no data on the mics and they honestly sounded horrible the way they were set up.  We had a mic on the Snare, Hi Tom, Low Tom and Kick.  I started with the kick drum and pleaded for two microphones to be purchased, working with a budget that was far too exhausted for the year is tough when trying to get funds for microphones.  I settled on the Shure Beta 91A and the Shure Beta 52A and they got approved.

From http://www.microhone-data.com. Here is the specs of the Shure Beta 91A.

From http://www.microhone-data.com. Here is the specs of the Shure Beta 52A.

I place the Beta 91A inside the drum on a small towel and the Beta 52A is just inside the sound hole of the front head pointed a bit off center from the beater.  The combination of these two microphones is amazing.  You can really hear the body and natural sustain of the kick with these mics.  Here is a recording of a resent Sunday at NRCC worship song is called Happy Day:

Next, I grabbed an SM57 out of our amp room and switched out the no-name mic on the snare drum, this made a great difference as the SM57 is an awesome mic for the toms or snare drum.  My placement is having the rear of the microphone pointed at the hi hat to reduce bleed from the hi hat (cardioid microphone pattern).  I put the mic about an inch above the top head and an inch or two in from the rim pointed down toward the center of the bottom head.

From http://www.microhone-data.com. Here is the specs of the Shure SM57.

Next was the hi tom, with the old no-name microphone on there. I compared an SM57 with it, but I liked the sound of the older microphone there.  I just changed the placement a bit.  Placement is an inch above the top head, inch in from the rim, angled down pointed closer to the mic than the center of the bottom head.

Floor tom, we replaced with a Sennheiser e906 with the switches set to normal.  This is about two inches away from the top head about two inches in from the rim, pointed down toward the center of the bottom head.

From http://www.microhone-data.com. Here is the specs of the Sennheiser e906.

Lastly, there was no microphone on the overhead.  While in some venues you can get away with this, I want my main source of audio being from one point source.  We have a mono system so there is no use for stereo panning, but when your mind hears the same audio coming from two different places it does tricks.  So I added a AKG SE300B with the AKG CK91 capsule on it.  It is a small diaphragm condenser mic that we had in the back room which sounds really nice as an overhead.  Because we have a mono system in our worship center we do not need a stereo mic setup for the overheads.  I have the single overhead about 1/2 ft above the cymbals and equidistant between the kick and snare.

From http://www.microhone-data.com. Here is the specs of the AKG CK91.

Up at the FOH we have two DBX 1064’s which are quad channel compressors.  I decided to put two of the channels on the snare and kick running a pretty high compression ratio getting about 8dB of gain reduction on peaks of the snare with a limiter just incase our drummer decides to do a bunch of rim shots.  The kick drum is running about 8-10dB of gain reduction with no limiter.  I wish I had controls over the attack and release however the DBX compressors are doing just fine.

Overall, the drums have improved tremendously in comparison to before.  The drum set could use a new set of heads, our church budget is renewed in a month or so.  In the room, the kick drum is filling the room quite a bit more and really sounds great.

Spending a few practices trying different microphones out, changing mic positions, adding compressors and tweaking EQ settings can really make a huge difference.  I will say this time and time again, if you are coming into a church as a new lead engineer do not take anything for granted, check every mic cable, every microphone, just make sure that everything is setup the way it should be.

Drew Brashler