Article

Tuesday, May 01, 2012

Put Another Nickle In: How The Jukebox Kick Started The Beginning Of Rock and Roll

For some of those who were there, Buddy Holly and Bill Haley will never sound better then when they were first blasting from a jukebox after inserting a nickel in a Wurlitzer...

Somewhere in its early years, the coin operated record player acquired the name “Jukebox”. There are several theories about the origin.

The most accepted is that the word “juke” is a corruption of the word “jook”, an African American slang term for dancing. The source of the music for this dancing would have been called a “jookbox”.

A second version is that “jook” meant “sex” which may have made sense since brothels were some of the first establishments to install jukeboxes, thus replacing the piano player.

A third source of the word may have been from the term “jute”, or “jute joints” where jute pickers would relax, drink and dance. Whatever the source of its name, the jukebox of the 1920s was generally associated with “speakeasies” and the “low-life” of prohibition since they were featured entertainment in such places.

To pay to hear a record played first started through the entrepreneurial activities of carnival and penny arcade operators who made their own recordings and then charged admission to hear them on the newly invented gramophone.

It was in response to requests by this group of users that the phonograph/gramophone manufacturers began to produce prerecorded product.

This was an unexpected life-line for the Columbia company that in 1890 seemed headed for liquidation, because the intended use of the phonograph as a dictating machine had been a dismal flop.

Columbia and Edison began to realize that their market was somewhere else. They also recognized that in order to sell players, they had to produce and manufacture prerecorded product that the public wanted to hear.

Initially the preferred programs for coin-operated players were comic songs, bands, monologues, and whistling. The revenues from these “pay for play” machines was amazing in light of the fact that the quality was poor and the selection meagre. In 1891, some machines earned up to 14 dollars a day - a lot of money at the time.

While accepting there was a market for coin operated carnival players, Edison feared they might create the impression that the phonograph was only a toy. His worries were unjustified, since the showman-operated players cultivated a consumer appetite for recorded music and a desire for home players.

As the turn of the century approached, mainstreet penny and nickel arcades were becoming an increasingly popular center for entertainment. There were hundreds of different coin operated amusements. The most popular of these were those that played music. Into this market came the nickelodeon and the jukebox.

The first jukebox appeared close on the heels of the introduction of the phonograph. Louis Glas installed an Edison cylinder system at the San Francisco Royal Palace in 1889.

In 1906, the Automatic Entertainer, which used flat disks recently invented by Berliner, was introduced by the John Gabel Company. The system was entirely mechanical but required regular winding of its spring mechanism. It was popular in spite of the poor quality.

In Paris, at the Pathe Salon du Phonograph, patrons could choose a musical selection, which would be played for them from the floor below where there were a battery of players. As in San Francisco, they would hear their selection through long listening tubes connected to the player’s diaphragm.

Composer Claude Debussy, after hearing this system for a few coins, was concerned that the low cost of the disk and its availability would have the effect of cheapening the music. He did, however, acknowledge that the discs preserved a certain magic.

In 1913, Debussy wrote: “In a time like ours, when the genius of engineers has reached such undreamed proportions, one can hear famous pieces of music as easily as one can buy a glass of beer.

“Should we not fear this domestication of sound, this magic preserved in a disc that anyone can awaken at will? Will it not mean a diminishing of the secret forces of the art, which until now have been considered indestructible? “.

Debussy, like so many other classically trained musicians had fears that this new technology would impact on his beloved art, and probably his concert income. The jukebox and nickelodeon changed the way people heard the music of the day by placing it within reach of the masses.

Mechanical jukeboxes continued to be one of many amusement machines in these penny arcades, but in the late 20s with the introduction of the electric phonograph, motors and amplification, the modern jukebox became a reality.

In 1926, J.P. Seeberg, a Swedish immigrant to the U.S., invented an electric system that was coin operated and would play any of eight records.

A year later, Automated Musical Instruments introduced its electric jukebox. Unlike their mechanical predecessors, that could only be heard by fee-paying patrons standing near the machine, these systems were capable of filling an entire room with sound.

These innovations further popularized the jukebox, and so began the modern jukebox craze.

The other two major manufacturers of jukeboxes appeared in the early 1930s. Wurlitzer, a long-time manufacturer of pianos and player pianos, introduced its first jukebox in 1933.

And in 1935 David Rock-Ola (his real name), whose company had been building scales and coin-operated games, introduced its first jukebox.

When the great depression occurred in the 1930s, the jukebox business became the one bright spot for the record industry.

For the public, a nickel would pay for six plays and like the movies of the day provided a few minutes escape from the depression.

There were two other historical events that helped the jukebox gain prominence.

The repeal of prohibition in the U.S. in 1933 meant that there were now tens of thousands of bars, clubs, and other drinking establishments that were installing jukeboxes for entertainment.

The second was the outbreak of World War II, and the relocation of millions of young soldiers to camps in far-away locations. For entertainment, the armed forces installed hundreds of jukeboxes in PX’s and service clubs all over America and overseas.

While these young people would have frequented their local jukebox back home, those machines would have had only a couple of types of music in the 24 available selections, and would have been chosen to suit the area and the jukebox’s clientele.

But the military jukeboxes were unique in that they were stocked with a range of music to satisfy the varied tastes of those who had come from every part of the country and ethnic background.

American blues, gospel, country and pop records were all thrown together on military jukes that introduced GIs to all sorts of music that came from outside of their home community and culture.

Almost overnight, American regional music, never really played on radio before, was heard by those from every region of the country. Many of these young people were also musicians that would now explore, absorb, learn, appropriate, and embrace pop music styles they had never heard before.

After the war this would have a significant impact on the coalescing of those musical roots that would form rock and roll.

On the home front during W.W. II, there was a growing juvenile delinquency problem with so many parents unable to pay attention to their teenagers. Dad was away at war, and Mom was working in a defense plant.

During the early 1940s, throughout America, youth centers were opened for after-school and weekend activities. To bring in the teens, free jukeboxes were brought in, turned up, and rarely turned off. The program was successful.

But, by the late 40s, the jukebox had fallen out of favor with the conservative establishment and was increasingly considered a corrupting influence. One prominent critic wrote in 1948 that the jukebox was responsible for ‘the musical tastes of America’s youth starting on a steady decline.”

That year Frank Sinatra was the most popular artist in the country. For such critics, things would get far worse.

For many Americans in the early 1950s, rock and roll was the devil’s tool, and existed for no other purpose than to morally corrupt the youth. For the first time teenagers had their own beat, and it could be found blasting out of the malt shop jukebox.

By 1956 there were somewhere around 750,000 jukeboxes swallowing dimes in America. Since most radio stations were only playing the most sanitized rock and roll selections, the jukebox was the source for the majority of rock music, particularly those machines in racially mixed neighborhoods. These machines had records of black artists who were singing rhythm and blues and early rock.

The public had heard from the pulpit and conservative press about the evil, passion firing sounds thumping from those machines sitting at the end of the bar or in the middle wall of the malt shop, but when Evan Hunter’s book, The Blackboard Jungle, was made into a movie in 1955, the older public was convinced. They had not beaten Hitler to see their children’s minds lost to the devil’s music.

When you added up the title song “Rock Around The Clock” with the images in the movie, it was obvious to anyone over 30 that rock and roll equaled teenage delinquency. The jukebox had become an integral part of rock and roll imagery.

In many areas of America, the government required a sticker on the jukebox stating that “minors are forbidden by law to operate this machine,” but generally, the jukes remained uncensored.

However, the jukebox operators were frequently placed under suspicion of jukebox stacking, a form of payola where they would be paid to put a record in the machine. Those who operated jukeboxes didn’t kick this image until the 1970s.

Coin operated music delivery systems did not decline as gramophones became a common addition to homes. The opposite was the case. With the spread of domestic record players within the upper middle class, along with radio, a desire was created for recorded music throughout the entire population.

Coin operated systems allowed anyone for the price of a few pennies to hear their favorite and/or the latest record. Increasingly, these customers were the young. In general the first phonographs were controlled by older people (parents) whose musical tastes were toward classical and music of their generation. 

To hear the latest. young people had to go to the juke at their local hangout. Not until the late 1950s was the cost of reproduction systems, headphones, and the records themselves so affordable that young people could have a record player of their own that they could control.

Most of them got that first record player with the detachable speakers as a Christmas present from parents who never realized that from that day forward “turn it down” would become one of their most-often used phrases.

Choosing what records would go in the jukebox was probably the origin of the “Hit Parade”, due to the limited number of records that could go into a machine, and the practice of installing new records weekly based on which ones were and were not played.

The jukebox brought the choice of what music would be played down to who wanted to hear a song badly enough to spend a nickel.

Often these would include recordings of local acts that were prominent in that specific community. In the mid 1930s, every jukebox held a smattering of local releases.

By 1940, those who chronicled the U.S. record industry were recognizing the importance of the jukebox. Jack Nelson wrote in Billboard that “coin operated phonographs, through a tremendously wide distribution, appeal to millions of individuals everyday, thus ensuring for this industry an important part in the next phase of American music”.

The jukebox had become a significant centerpiece anywhere small-town America gathered, and record sales to the jukebox operators were becoming significant.

The jukebox provided anyone with nickle instant grass-roots musical satisfaction.

As Chris Pearce describes it, “It was the jukebox into which the lonely trucker at the coffee shop dropped his nickel to inspire dreams of his baby back home, the jukebox that the kids made for in Chuck Berry’s song when they wanted to hear something really hot, the jukebox that linked communities whose local operator stocked it with songs and dances from the old country”.

From the 1920s to the 1960s, jukeboxes electronically and mechanically advanced by increasing the capacity of their changers, better amplifiers and speakers, selectors at each table, roll around selector, and so on.

Of paramount importance was the “look” of the machine. The jukebox had to be visually exciting. The exterior design became a key to the jukebox’s success. Seeberg and Wurlitzer hired top industrial designers just when Modernism was coming into vogue.

Translucent colored plastic was starting to be widely used and was ideally suited for the illumination of the jukebox. Most manufacturers believed that the customer wanted to see the record changer work and a cabinet that lit up.

Wurlitzer dominated the post W.W. II market with its classic machine, the 1015, which featured colored arcs and floating bubblers.

But in 1948, Seeberg introduced the first jukebox to handle 100 selections, the Select-O-Matic 100.

The number of records that could be played had gone from a couple of dozen records to 50 records, with both sides available for play. Until the introduction of the Select-O-Matic 100, the industry believed that 24 titles were all that were necessary for a selection of “pop” songs.

The other jukebox manufacturers quickly redeveloped their mechanisms to accommodate more records when it became obvious that the customers wanted a wider selection, and by 1956, 200 titles were available in a jukebox.

The expansion in capacity also meant that a wider variety of records could be available. Country and western and rhythm and blues could finally live in the same jukebox with Perry Como, Bing Crosby, Bill Haley and Elvis.

Unquestionably the biggest change to hit the jukebox industry came in 1948, when RCA introduced the 45.

Not only did they sound better than the 78s. but they were lighter, smaller, and the center hole was large and more suitable for automated operation.

In short, it was the perfect record for a jukebox. The 45 in the jukebox of the 1950s would become the focal point of the teenager and the first line source of rock and roll.

Until television forced radio to reinvent itself, radio was the mass medium, and with few exceptions had generally ignored blues, country, and other regional or “fringe” music. The jukebox filled this void.

In the 1950s, it was the jukebox where teenagers would find the latest in music. They were doing what Teresa Brewer suggested -  “put another nickel in…” - but they were selecting Chuck Berry, whose advice was to go “up to the corner and round the bend, right to the juke joint you go in. Feeling the music from head to toe, round and round, and round you go. Hail, hail, rock and roll! Deliver us from the days of old!”

Teresa didn’t know it, but Chuck was saying her days as a pop artist were numbered, as was the style of recordings she made.

These machines were more than music delivery systems, their external designs were trend setters in the art deco movement and an important aspect of their popularity. They offered the latest music at a time when most of the public could not afford to buy a record, much less their own playback system.

The jukebox was key to the popular spread of country, hillbilly, rhythm and blues, and of course the development of rock and roll music. For a generation, the jukebox at the local hang-out was the only place that some of the “hippest” and latest rock and roll could be heard.

Their significance has declined over the last few decades but in the 1940s through the early 1960s they were an important focus for the young. Rock and roll might have been beaten down by the establishment if it had not been for the existence of jukeboxes in every bar, hamburger drive-in, bowling alley and malt shop where young people congregated.

For some of those who were there, Buddy Holly and Bill Haley will never sound better then when they were first blasting from a jukebox after inserting a nickel in a Wurlitzer. For those who weren’t there, its hard to capture it all, since it wasn’t just the jukebox that held the sound, it was where it was happening in time and place when teenagers and rock and roll were being invented.

As a 1950s Wurlitzer ad stated, “For millions, the jukebox was ‘America’s favorite nickel’s worth of fun’.”

Currently residing in Australia, Tom Lubin is an internationally recognized music producer and engineer, and is a Lifetime Member of the National Academy of Recording Arts and Sciences (Grammy voting member). He also co-founded the San Francisco chapter of NARAS. An accomplished author, his latest book is “Getting Great Sounds: The Microphone Book,” available here.

More from Tom Lubin on PSW:
You Make Music You Say? So, You’re In The Fashion Industry

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Monday, April 30, 2012

Power Lines: Determining When “Isolated Grounding” Is Needed

Options and errors in AC wiring

If electrical wiring, from main breaker panel to outlet, consists of Romex and plastic J-boxes, an “isolated” or “technical” ground system is already in place. This is the case In most, but not all, residential
wiring.

However, when wiring consists of metallic conduit and J-boxes, as in most commercial buildings, an isolated safety-grounding scheme can sometimes reduce audio system noise. It is most applicable in situations where conduit may come in contact with building steel, water pipes, gas pipes, or other structures because they may be grounded at distant locations (perhaps even the building next door or across the street) and will inject noise current into the safety ground system.

Special insulated ground or “IG” outlets (distinguished by a triangle marking on their face and, most popularly, orange in color) are used, which intentionally insulate the green safety ground terminal from the outlet mounting yokes or saddles.

Therefore, safety grounding is not provided by the J-box and conduit, but by a separate insulated green wire which is routed back to the main electrical panel.

Of course, the J-box and conduit must also be properly grounded by other, usually existing, means. Code requires that this or any safety ground conductor be routed in the same conduit or cable as the line and neutral circuit conductors. Although not explicitly stated in Code, this practice prevents loop inductance from limiting fault current, assuring fast breaker response should a fault occur.

Most often, wiring is not “daisy-chained” to outlets on the same branch circuit, so noisy leakage current from one device couples less to others on the same branch circuit. However, inductive coupling from phase conductors to the ground conductor (the major source of ground voltage differences between outlets) is not reduced.

Technical grounding practices are covered by NEC Article 250-74 and its exceptions. An excellent reference for system grounding, with emphasis on equipment racks, is a white paper by Middle Atlantic Products (to which I was a contributor) available at www.middle atlantic.com/pdflPowerWhitePaper4_07.pdf.

A potential problem with isolated grounding is that unaware users may connect a signal cable between a piece of equipment powered by an isolated ground outlet to another piece of equipment powered by a non-isolated ground outlet. Noise now enters the isolated ground system via the signal interconnect.

This problem is very common in large computer networks and, like most computer noise problems, will likely be blamed on something else.

The resultant ground loop and circulating noise currents defeat the purpose of the isolated ground system.

THE NEUTRAL-GROUND SWAP

National Electrical Code recommends that premises wiring be sized such that regulation of the most distant outlet on a branch circuit is 5 percent (6 volts of drop for 120-volt circuits) or better under full load. This means that 3 volts are dropped across both line and neutral conductors.

On the other hand, safety ground wiring normally carries well under 100 rnA of cumulative equipment leakage currents. This could occur for a worst-case scenario of 20 pieces of equipment, each having a three-prong AC plug and 5 rnA leakage current (maximum allowed by UL). In this scenario, total voltage drop over the length of the safety ground conductor would be only about 20 mv.

But, as shown in the schematic directly below, a simple outlet wiring error that swaps the neutral and safety ground conductors allows load current to flow in the safety ground wiring.

When equipment load currents of 15 or 20 amps flow in the safety ground wiring, voltage drops as high as 3 volts can occur over its length (assuming the safety ground wiring is the same gauge as line and neutral).

Although the outlet is still functional and safe, this error can cause system ground noise to increase by a factor of 100 or about 40 dB!

Simple outlet testers cannot find the problem because they test only for voltages at the outlet.

Since neutral and safety ground are bonded together at the main breaker panel, they are indistinguishable to these testers. Even more sophisticated testers cannot reveal the error.

However, if nominal loads (say 100-watt light bulbs) are plugged into each of the outlets on the branch circuit, a clamp-on ammeter like the one shown here at left can quickly reveal and locate the mis-wiring by measuring current at points A, B, and C.

Referring to the schematic, abnormally high currents would be measured at locations A and B but not C.

Bill Whitlock has served as president and chief engineer of Jensen Transformers for 20 years. Read more articles by Bill about best AC and electrical practices here.

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Church Sound: Vocal Microphone Technique (Includes Video)

It's not to the vocalist's advantage to send a poor signal to the sound system

It’s always been amusing to watch the band set up. The guitarist brings his amp, a few pedals, and maybe a couple of guitars.

The bass player brings his instrument, and often his own amp. The drummer uses the church’s drum kit, but he brings his own sticks and takes the time to tune and position the drums to his liking.

But the vocalist just uses whatever mic is handed to them.

My experience has been that the choice of microphone for the vocalists, especially the lead vocalist, has a substantial effect on her sound in the house, her intelligibility, and even her confidence in front of a crowd.

Using “whatever they give me” would be like the the guitarist playing “whatever guitar they hand me,” whether it’s a Fender Squire or a Paul Reed Smith Custom 24 guitar, or the sound guy saying, “Yeah, whatever. Behringer, Midas, Yamaha, DiGiCo: they’re all the same.”

The point: if you’re a vocalist, find a mic that really lets your voice give its best in your facility. If you’re the sound guy, then give real thought to what mics sound best on which vocalist, particular the main vocalists. Try out some new ones if you need to, and teach your team that “This is John’s mic!” Or encourage John to buy his own vocal mic.

And of course, audio engineers love working with untrained vocalists, who sing away from the mic, lean into the mic for their loud notes, and cup the grill (sarc off). The reality is that a good sound system will clearly amplify whatever sound (good or bad) that the vocal mic picks up. It’s not to the vocalist’s advantage to send a poor signal to the sound system.

Audix created this video, and they make some excellent vocal microphones (and some amazing instrument mics), including some at modest prices. Of course, they use Audix mics in these brief clips. But the techniques are appropriate for any handheld vocal microphone.

Check out more from David McLain at the Church Soundguy blog.

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Friday, April 27, 2012

In The Studio: The Six Rules For Adding Effects To A Mix

Choosing the best effects for each track more efficiently

 
Having trouble figuring how to use your effects during mixing?

Here are a set of rules that can help you choose the best effects for each track more efficiently, courtesy of The Mixing Engineer’s Handbook.

Rule 1 - As A General Rule Of Thumb, Try To Picture The Performer In An Acoustic Space And Then Realistically Recreate That Space Around Them.

This method usually saves some time over simply experimenting with different effects presets until something excites you (although that method can work too). Also, the created acoustic space needn’t be a natural one. In fact, as long as it fits the music, the more creative the better. 

Rule 2 - Smaller Reverbs Or Short Delays Make Things Sound Bigger.

Reverbs with decays under a second (and usually much shorter than that) and delays under 100 milliseconds (again usually a lot shorter than that) tend to make the track sound bigger rather than push it back in the mix, especially if the reverb or delay is stereo.

Rule 3 - Long Delays, Reverb Predelays, Or Reverb Decay Push A Sound Farther Away If The Level Of The Effect Is Loud Enough.

As stated before, delays and predelays (see below) longer than 100 ms (although 250 is where it really kicks in) are distinctly heard and begin to push the sound away from the listener. The trick between something sounding big or just distant is the level of the effect. When the decay or delay is short and the level loud, the track sounds big. When the decay or delay is long and loud, the track just sounds far away.

Rule 4 -  If Delays Are Timed To The Tempo Of The Track, They Add Depth Without Being Noticeable.

Most engineers set the delay time to the tempo of the track (see below on how to do this). This makes the delay pulse with the music and adds a reverb type of environment to the sound. It also makes the delay seem to disappear as a discrete repeat but still adds a smoothing quality to the element.

If you want to easily find the right delay time to the track and you have an iPhone, grab my “Delay Genie” app from the iTunes App Store. It’s free and will making timing your effects to the track incredibly easy.

Rule 5 - If Delays Are Not Timed To The Tempo Of The Track, They Stick Out.

Sometimes you want to distinctly hear a delay and the best way to do that is to make sure that the delay is NOT exactly timed to the track. Start by first putting the delay in time with the track, then slowly alter the timing until the desired effect is achieved.

Rule 6 - Reverbs Sound Smoother When Timed To The Tempo Of The Track.

Reverbs are timed to the track by triggering them off of a snare hit and adjusting the decay parameter so that the decay just dies by the next snare hit. The idea is to make the decay “breathe” with the track. The best way to achieve this is to make everything as big as possible at the shortest setting first, then gradually get longer until it’s in time with the track.

Of course, the biggest part of adding effects to a mix is experience, but following these rules will provide a perfect place to start.

Bobby Owsinski is an author, producer, music industry veteran and technical consultant who has written numerous books covering all aspects of audio recording. For more information be sure to check out his website and blog.

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Church Sound: Three Signs Your Monitor Settings Are Bad

Some approaches for making it work

 
Evaluating your monitor settings is a crucial step during the sound check and during the church service. 

Consider these three signs your monitor mix is bad or had gone bad.

1. A musician isn’t playing in time or looks lost.

This will happen if you didn’t get the right monitor mix during the sound check. It can also happen if they can’t hear the monitor due to the addition of the singing congregation.

Pro-actively, ask the musicians, at the end of the sound check, if they are OK with the monitor levels. During the service, if this happens, increase the overall volume of the musician’s monitor. As you reach the right volume, you’ll see the lost look go away and the house mix improve.

2. The house mix doesn’t sound right.

This will happen if the monitor volumes are too loud.

You can check this during the last part of the sound check. Walk to the third row from the stage and listen. You should hear the house loudspeakers and possibly a little monitor/stage volume. If the monitor volume is too great, then you need to cut the monitor volumes.

3. Musicians are bouncing up and down.

That is to say, they are signaling they need more volume or “more me” in their mix. This isn’t so much a sign the mix is bad as it’s a sign the addition of the singing congregation has made their monitors hard to hear.

In the case where you have multiple musicians signaling for more, increase the volume of your master monitor aux sends. This keeps the monitor mix consistent while increasing the overall volume. If it’s just one musician, go for a small bump in the monitor volume followed by a slight bump in their channel in the monitor.

A Tip Regarding Monitors & Guitar Amps

Consider guitar amp placement if you are having monitor problems. For example, if a singer keeps complaining they are hearing too much of the guitar, check where the guitar amp is placed. They might be on-axis with the guitar amp and so the monitor mix has nothing to do with it. Consider placing the guitar amp in a different direction or placing it on the opposite side of the guitarist so no one else is on-axis with it.

The Take-Away

Check with musicians during and after the sound check regarding their monitor mixes and volume levels. When it comes time for the church service, expect to make a monitor volume increase. 

Also, know that monitor mixes might have to change so keep your eyes on the musicians during the service, especially during the first song.

Ready to learn and laugh? Chris Huff writes about the world of church audio at Behind The Mixer. He covers everything from audio fundamentals to dealing with musicians. He can even tell you the signs the sound guy is having a mental breakdown.

 

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Thursday, April 26, 2012

In The Studio: Going For That “Vintage” Sound

What the old-school folks did, and it worked well for them

 
Do you know the reason we rave over those “vintage” recordings?

Imperfection.

Forty years ago, they didn’t use tubes, transformers, and tape by choice. Nope, it was all that was available to them at the time.

The “sound” of so many of our favorite recordings from that era came from running the audio through gear that couldn’t faithfully and accurately reproduce the signal.

Each piece of gear added something to the sound — warmth, low end, smoothness, punchy-ness, even noise.

In short, it was near impossible to get a clean, accurate recording. The gear added to the sound.

Our recording heros aren’t heroes because of the tools they used. They’re heroes because of how they USED the tools.

They took this “imperfect gear,” with all its pros and cons, and they made great music with it.

I really believe we have a huge advantage today.

We have access to affordable equipment that will give us nice, clean, accurate recordings. AND we can also get the “vintage” gear sound if we want, too.

The key difference between now and then? We have a CHOICE.

Do those “vintage” albums sound great? Sure.

Does that mean we need to make our recordings today sound just like those? Heck no.

Here’s what I think.

I vote that we constantly push the envelope, and use the technology available to us RIGHT NOW to make great-sounding music, even if it doesn’t sound like the hallowed vintage recordings we hold so dear.

Don’t get stuck in the past. You don’t need a $3,000 tube compressor to get great recordings. Heck, you might even find that the big fancy comp doesn’t even give you the sound you really wanted anyway.

Use whatcha got. That’s what the old-school folks did, and it worked well for them.

For example, I’m still amazed at the different tones I can get with a simple, run-of-the-mill compressor plug-in.

You can too.

Joe Gilder is a Nashville-based engineer, musician, and producer who also provides training and advice at the Home Studio Corner.Note that Joe also offers highly effective training courses, including Understanding Compression and Understanding EQ.

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Microfiles: Electro-Voice 664, The Legendary “Buchanan Hammer”

A single-element cardioid, dynamic type mic was the first model to incorporate the company’s patented Variable-D design

My Baltimore-area high school theater was outfitted with the first quality PA system I ever worked with.

It had JBL horns and cabinets in a center cluster, powered by Crown amplifiers, with a 6-channel TAPCO mixer in the sound booth and Electro-Voice 664 microphones on stage.

Initially, to my finely tuned 10th grade ears, the system didn’t sound very good – the performers could barely be heard, and there was a lot of feedback.

It wasn’t long before I figured out that the real problem was operator error, not the system.

I eventually got the hang of running it correctly, including learning the importance of proper mic placement.

And what mics they were! I fell in love with the EV 664 – far better sounding than my own Lafayette mics for my garage band, and built like a tank. Later, I wasn’t surprised to find out the 664 had the nickname “Buchanan Hammer” due to its rugged design.

The story behind the nickname, as I know it, is that during his legendary microphone lectures, the late Lou Burroughs (one of the founders of EV) would beat a 664 against a 2 x 4, and/or use it to hammer nails into a board, and then plug it in and use it for the rest of his presentation. The Buchanan part of the moniker refers to the town in Michigan where the company was headquartered for decades.

Timeless Design
Introduced in the mid-1950s, the 664 sported a cool “Art Deco” design, with a sleek yet curvy chrome body that evoked the popular automobile tailfins of the period. Inside was some serious technology.

The single-element cardioid, dynamic type mic was the first model to incorporate the company’s patented Variable-D design (U.S. patent number 3115207, awarded in 1963) still found in several EV mics to this day, including the broadcast-favorite RE20 and the recently introduced RE320.

Variable-D (“Variable Distance”) uses three ports to cancel sound from the rear, while the side ports (slots located on the sides) are coupled to the back of the diaphragm and help cancel high-frequency sounds.

The hole on the top (located toward the front of the body’s “raised spine”) works the same as the side ports, but has a longer path and added filters to affect mid frequencies. The single hole at the rear of the spine has a longer path and more filtering to address low frequencies.

This all combines to give the mic good pattern control over a wide frequency range and a reduced proximity effect. An ad from 1961 states “The 664 does not BOOM when performer crowds microphone.”

The 664 was available in three finishes: satin chrome, non-reflecting gray (664A) and a gold finish (664G). I also own one that is branded Executone EXCC, and it has a brown body with a gold windscreen, as well as a chrome model that is branded DuKane 7A160.

More Versions

There were actually two designs of the 664. The earlier one had a more classic base, three ports on each side, and the old script logo on the switch-plate. The later version had a more modern rounded base section, a single large port per side, and a larger switch-plate cover that featured the round, red EV logo. Both utilized a 4-pin EV screw-on connector that was popular on many of their models.

The 664 shipped with an 18-foot cable that was not terminated at the console end.

There was yet another version called the 664A, but it was supercardioid, a lot smaller, and had more modern styling. It still utilized the Variable-D design, but had a long plastic port along its spine instead of separate rear port entrances.

The 664 holds a special place in my collection. It was the first true professional caliber microphone I worked with, and the single port model shown here is the first mic I ever purchased just to collect and not use onstage.

Great mics and great memories!

Electro-Voice 664 Specs (original 3-port model)

Transducer Type: Dynamic, non-metallic Acoustalloy diaphragm
Polar Pattern: Cardioid
Frequency Response: 40 Hz – 15 kHz
Sensitivity:  -55 dB at 150 ohms
Nominal Impedance: Switchable high impedance or 150 ohms
Size:  7 3/16 x 1 7/8 inches
Net Weight: 28 ounces
1961 Price: $49.98

Craig Leerman is senior contributing editor for Live Sound International and ProSoundWeb.

 

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Wednesday, April 25, 2012

Differences, Cause & Effect And Consequences Of Polarity And Phase

The terms "polarity" and "phase" are often used as if they mean the same thing. They do not.

Polarity and Phase - these terms are often used as if they mean the same thing. They are not.

POLARITY: In electricity this is a simple reversal of the plus and minus voltage. It doesn’t matter whether it is DC or AC voltage. For DC, Turn a battery around in a flashlight and you have inverted or, more commonly stated, reversed the polarity of the voltage going to the light bulb. For AC, interchange the two wires at the input terminals of a loudspeaker and you have reversed the polarity of the signal coming from that loudspeaker.

PHASE: In electricity this refers only to AC signals and there MUST be two signals. The signals MUST be of the same frequency and phase refers to their relationship in time. If both signals arrive at the same point at the same time they are in phase. If they arrive at different times they are out of phase. The only question is how much are they out of phase, or stated another way, what is the phase shift between them?

The important point to note in these definitions is that you can reverse the polarity of one signal and you can measure this change. You need two signals to measure a phase shift.

For convenience, the word “speaker” will be used in place of the more correct term “loudspeaker” in the rest of this article.

A picture is worth 1,000 words… but a few words of explanation can help.

The following figures show the differences and some consequences of polarity and phase. Figures 1 through 12 show graphs of sine wave signals. Actually it is a sine wave from one signal source split two ways. Except for figure 1, one of the splits is “processed” by reversing its polarity and/or by delaying it (phase shifting it) as described. To put this in the real world, imagine two speaker systems side-by-side, each reproducing one of the signal splits. (More precisely, the graphs show what you would see on an oscilloscope looking at the output of a mixing console with each split going to a separate input after one of the splits has been “processed”.)

The vertical scale in the graphs is in arbitrary units of -2 to +2 with lines at each 0.5 interval. If you like, consider this as -2 to +2 volts. Because phase shifts are measured in degrees, the horizontal scale in the graphs is labeled in degrees with a vertical line at each 90-degree point. One full cycle or period of a sine wave is 360 degrees.

Assume that the signals shown are 1 kHz sine waves, in which case each vertical line represents 1/4 millisecond of time. Sound travels in air about 3.4 inches (85 mm) in 1/4 millisecond so each vertical line also represents this distance. Note that in the graphs the signals all start 1/4 millisecond or more from the left so you can clearly see when each signal starts. (The importance of this will be seen in figure 9.) There is no signal along the flat line from -90 to 0 degrees.

Signals In Polarity, In Phase
Figure 1: This shows 3 periods or 3 cycles of two simple sine waves. Both are +/-1 volt high at their peaks = total of 2 volts. One is shown in blue the other in red.

Figure 2: This is what happens when the two are combined (= added together). This is exactly what would happen on a line exactly between the two side-by-side speakers. The two signal beings being in phase and in polarity add up so the peaks are now at the +/- 2 volt lines = 4 volts or twice the original signals. Acoustically this is an increase of 6 dB = 20 x log(1+1).

Signals Out of Polarity
Figure 3: This is like figure 1 but the second sine wave, shown in red, has been reversed in polarity. As you can see the + and - voltage points are exactly opposite from the first sine wave, shown in blue. This would be accomplished by reversing the +/- input connection on the speaker reproducing the red sine wave.

Figure 4: This is what happens when the two are combined. Each point of the two signals being in phase, but opposite polarity, adds up to zero. Acoustically this is an infinite decrease of output. Because you can’t take the log of 0 assume the difference is actually 0.0.01 volts (the dots = 58 more zeros). 20 x log of this number is -1200 dB. That should be pretty quiet. You can’t easily hear this with two speakers because of having two ears. But using a very carefully positioned microphone to measure this in a place with no sound reflections, you would find almost no signal.

Signals Ot of Phase

Figure 5: The second sine wave, shown in red, starts 1/4 millisecond later (90 degrees later) than the first one, shown in blue. Put another way, the second signal has been delayed by 1/4 millisecond.

Figure 6: This is what happens when the two are combined and it’s pretty interesting. First notice that the peaks are almost at the +/-1.5 volt lines. The value is actually +/-1.414 volts. This is a 3 dB increase. This would be like listening to two speakers but the one reproducing the red sine wave is 3.4 inches (85 mm) further away from you than the other. The first thing you hear is only from the speaker reproducing the blue sine wave. The black line starts when the sound from the second speaker is heard and this line is the combined signal of both speakers.

Suppose the speaker reproducing the red signal were only 2.25 inches (57 mm) further away. The signals would be shifted by only 60 degrees. The increase for the combined signal would be about 4.5 dB. So the amount of phase shift is important.

The second thing to notice is what happens at 1/4 millisecond or 90 degrees after the blue signal starts when the second signal “kicks” into the picture represented by the line turning black. There is a distinct change in the waveform.

The third thing to notice is that the entire waveform after the “glitch” is shifted in time compared to figure 7 about 45 degrees = average of 0 and 90 degrees.

Signals Out Of Phase And Polarity

Figure 7: The second sine wave, shown in red, is a combination of the sine wave in figures 3 and 5. The signal not only has its polarity reversed but it is shifted in phase by 90 degrees compared to the first signal, shown in blue. In this case the speaker reproducing the red sine wave has its +/- input connection reversed in polarity and is 3.4 inches (85 mm) further away from you than the one reproducing the blue sine wave.

Figure 8: This is what happens when the two signals are combined. The picture is similar to figure 6 with two important differences. First the “glitch” at the point where the second signal starts is different. This is the point where the line turns black. Second is that the entire waveform is shifted by 45 degrees again but this time to the left of the original signal.

The “Glitches”
The glitches in figures 6 and 8 give an indication of what happens during the onset of a signal. While the so-called steady state portion of the combined signal (shown by the black portion of the lines) looks the same except for the amplitude change, these glitches will affect the transient attack of sounds. This is not to say that either will sound horrible, but a phase shift between otherwise identical replicas of a sound WILL make a difference in the sound of the initial transient attacks, depending on the frequency and amount of phase shift.

This is exactly the kind of phenomena that can occur in the crossover region of a speaker. This is because the distance from each driver to the listener is usually different and the crossover itself shifts the phase of the signal between the drivers. Speaker designers are often faced with a choice between something like what you see in figures 6 and 8. Neither is “correct” so a designer can only choose the one that “listens” better. Just looking at these two, I would bet the waveform in figure 8 might sound better and the choice would be to reverse the polarity of one of the drivers. These crossover “glitches” occur only over a small range of frequencies where both drivers reproduce the sound. It is well accepted by designers that this kind of “improvement” is sonically more significant than the fact that frequencies above and below the crossover point may be out of polarity.

Signal Phase Shifted 180 Degrees
This is where many get into trouble in thinking that phase and polarity are the same thing, meaning that it is often assumed that a 180 degree phase shift and reversing the polarity are the same.

Figure 9: In this figure each sine wave lasts for only 2-1/2 cycles. The second sine wave, shown in red, is shifted in phase 180 degrees from the first shown in blue. This is what would happen if the speaker reproducing the red sine wave were about 6.8 inches (170 mm) further away from you than the one reproducing the blue sine wave. You can see that between the 180 and 900 degrees the signals LOOK like they are simply out of polarity but they are NOT. It is VERY important to note that if you could not see the beginning or the end of these signals you could not tell whether they were out of polarity or 180 degrees out of phase. Too often this is what causes confusion between a polarity reverse and a 180 degree phase shift.

Figure 10: This is the result of combing the two signals. Unlike figure 4 where the signals are simply out of polarity, and completely cancel, there are clearly two positive halves of a sine wave visible before and after the two signals cancel along the black line between 180 and 900 degrees. The first is from the blue sine wave in figure 9 that occurs before the start of the red sine wave. The second is from the red sine wave in figure 9 that continues after the blue sine wave stopped.

Signal Phase Shifted 180 Degrees And Reversed In Polarity

Figure 11: This is the same as figure 9 but the polarity of the red signal is reversed from figure 9.

Figure 12: This is the two signals in figure 11 combined. Between the 180 and 900 degrees, the signals add much like in figure 2. However there are significant differences in the overall 90 to 1080 degree signal. The first 1/2 sine wave of this signal is only from the blue sine wave from figure 11. The last 1/2 sine wave is only from the red sine wave in figure 11. You can clearly see that both of these 1/2 sine waves are only 1 volt at the peaks. This is a clear difference from figure 2 where all the peaks reach 2 volts.

The reason is that the two signals in figure 11, even though identical, are offset by 180 degrees. They add together only between 180 and 900 degrees when both are being heard. More importantly, during this time period DIFFERENT parts of the same signal have added together. For example you can see that between 180 and 360 degrees it is the second 1/2 of the blue signal’s first complete sine wave that adds to the first 1/2 of the red signal’s first complete sine wave.

Real Audio Signals
Sine waves are easy to look at to dramatically show the difference between polarity and phase. Armed with this knowledge you can look at figures 13 through 18 that show something like a real audio signal where the effects of polarity and phase are more difficult to see.

The signal shown in these figures was a generated by a mathematical algorithm that produces something close to a pink noise signal. Pink noise contains all frequencies with an equal amount of energy in each octave band. Real audio signals don’t look much different than pink noise (but one would hope they sound better!). The scales on these graphs are arbitrary. You can look at the vertical scales as +/-3 volts if you like. However, because of the way the signal was generated, there was no way to define absolute time or degrees along the horizontal scales. Suffice it to say that the phase-shifted signal used in these figures was shifted by one data point out of the 240 data points that make up the signal lines.

There is one important thing to understand about phase shift. The amount of time one signal is delayed from another will have different effects at different frequencies. Assume there is a 1 millisecond time difference between two identical signals. At 500 Hz the result will be as shown in figure 10 because at 500 Hz the 1 millisecond time difference is a phase shift of 180 degrees. The signals are offset by 1/2 a cycle. At 1 kHz the signals will be offset by 1 complete cycle. In other words you would hear one cycle from the first signal then both combine then you’d hear the one cycle from the second signal after the first stopped. This is similar to what is shown in figure 12 (which shows only 1/2 cycle) but is not the result of the same conditions that were used to make figure 12. At 250 Hz the effect would be as shown in figure 6 because a 1 millisecond time difference corresponds to a 90 degree phase shift at 250 Hz or an offset of 1/4 cycle. At lower frequencies the phase shift would be even less and the signals would tend to add as in figure 2, approaching but never quite reaching the 6 dB increase shown in that figure.

Contrary to phase, polarity affects all frequencies the same way. It makes the positive portions negative and the negative portions positive. Put another way, it simply flips the signal over the same way at all frequencies. With these things in mind, examine figures 12 through 18

Effects of Polarity and Phase On “Real” Audio Signals
Figure 13: This shows a pink noise signal generated as noted above.

Figure 14: This shows both the original signal in blue and what happens when an identical but phase shifted signal is added to it, as shown in red. The red signal is similar to the combined signal shown in figure 6. Note the increases in signal level and the changes in the waveform (many glitches). However you can also see the combined signal follows the original fairly closely.

Figure 15: This shows both the original signal in blue and what happens when the phase shifted signal is also reversed in polarity and combined with it, as shown in red. In this case there are huge differences between the original and combined signal.

Figure 16: To better understand what is going on, this figure shows an averaged or integrated version of the pink noise signal in figure 13. This is basically what would you would see if you graphed the readings from a typical SPL meter for the signal in figure 13.

Figure 17: This shows the averaged signal from figure 16, in blue, and the averaged combined signal from figure 14, in red. Note that there are primarily level differences (mostly increases). Otherwise the two lines look very similar.

Figure 18: This really shows what is going on in figure 15. The blue line is the averaged signal from figure 16. The red line is the averaged signal from figure 15. The red line shows that the out of polarity and phase-shifted signal approaches a straight line. Because you are looking at a broad frequency range, you are seeing a severe cancellation of the lower frequencies due to the polarity reversal. However, unlike the low frequencies, the upper frequencies do not completely cancel due to the phase shift. The red line contains primarily high frequency energy. In the blue signal the higher frequencies are the small “bumps”. These can be clearly seen in the red signal and most of them correspond to those in the blue signal.

Figure 18 is a prime example of what you would hear if you stand exactly between two speakers playing the same signal (i.e. mono) with one speaker out of polarity. The bass will disappear. But, there will always be a difference in distance between you and the speakers due to the spacing of your two ears and probably a slight overall difference in distance between you and each speaker. A difference in distance means a difference in the time arrival and thus there will be phase shifts between the sound from the two speakers. The amount of shift will vary with frequency. Because of the shorter wavelengths at high frequencies, the phase shifts allow most of the highs to be heard. They may be out of polarity but the effect is like what is shown in figure 8. Also, in a room you would also hear sound reflections from the floor, walls, and ceiling. You would only hear something like the red line in figure 18 outdoors away from any reflective surfaces or in an anechoic chamber.

The small distance between your ears and any small difference in distance from you to each speaker do not cause appreciable phase shifts at low frequencies. This is because of the considerably larger wavelengths. The difference in your distance from each speaker might be only 1 inch (25 mm). However, the wavelength of even a 1 kHz sound is roughly 1 foot (300 mm) and at 100 Hz roughly 10 feet (3 m). At the lower frequencies the polarity difference predominates because the phase shifts due to the difference in your distance from the speakers is very small compared to the wavelengths of the low frequencies. Thus the lower frequency signals, being nearly in phase but out of polarity, will cancel like in figure 4. The lower the frequency the less the phase shift between the two speakers and the greater the cancellation.

A Polarity / Phase Field Trip!!
(As with all physical exercise, check with your doctor first, who might not recommend you do this for some reason.)

Find two railroad tracks, lie across them, and wait.

Two trains, one on each track, come along. Both are right side up and both hit you at exactly the same time. The trains are in polarity and in phase.

The same thing happens again and both trains hit you at exactly the same time. However, this time one train is upside down.

That is a polarity reversal.

The third time both trains are right side up but one hits you first and the other hits you shortly after the first. That is a phase shift.

The last time the second train is upside down and hits you later than the first. That is both a polarity reversal and a phase shift.

Summary
So there you have it. Although this has only touched on a few areas concerning phase and polarity issues, it is hoped you better understand the difference between the two and a few of the effects of each. Remember that the audio frequency range covers wavelengths of over 30 feet (10 meters) at the lowest frequencies to less than an inch (under 25 mm) at the highest frequencies.

While a reversal of polarity will affect all frequencies identically, a difference in time arrival between two otherwise identical signals will have very different effects on the phase between them. The amount of phase shift will be different at different frequencies and this will depend on how much time difference there is between the arrival of the two signals.

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In The Studio: Tips For Mixing Rap Vocals, Part 3 - Compression

Compression is a powerful tool that many struggle to fully understand

 
Check out Mixing Rap Vocals Part 1 and Part 2 before reading this article.

Compression is a difficult subject because there is a lot you can do with it.

So let’s look at the main reasons to grab a compressor before getting into some of the more intricate uses.

Quick Macro-Dynamic Control

Macro dynamics refer to words and phrases. These are the clear dynamics you can hear as “this part is louder, that part is softer.” The most transparent way to get things sounding even is to actually automate the vocals manually.

But sometimes time doesn’t allow for this approach. So if you aren’t automating, a light ratio, slow attack, slow release, just catching the louder moments with the threshold is a good way to even things out.

Micro-Dynamic Control

What volume automation might not catch is the very quick dynamic changes – loose spikes at the fronts of words. These spikes aren’t heard so much as “volume” but more as an overall quality to the vocal.

The issue with these spikes is two fold – first, they eat away at your headroom pretty quickly– second, they will trigger any compressors you are trying to use for purposes besides micro-dynamic control.

It can be useful to dedicate a compression stage toward pulling back these vocal spikes. Generally a fast attack and release, and a light ratio does the job. The light ratio is to retain the articulation of the word and minimize frequency skewing.

The key is to set the threshold low enough to catch as much of the peak as possible while effecting the body of the signal as little as possible. I try to avoid using limiters for this purpose. I like the Empirical Labs Distressor for this (especially for controlling peaks while tracking), as well as digital style compressors such as the Logic or Pro Tools stock compressors or the Waves C1.

The attack setting is very important – it’s usually between a number of nano-seconds and two or three milliseconds in the digital world, and on the faster side of things for the analog world (totally varies unit to unit).

Getting A Vocal To Stay Audible Through A Mix

The power of compression is that you can make something louder while not actually raising the peak volume of the signal. This becomes extremely useful for making something cut through a dense mix or to come forward. This is probably where the majority of compression work for rap vocals come in.

Rap is generally an in-your-face, visceral style of music. The kick is physical, the snare is physical, subtlety isn’t really the overall goal. And the vocals are paramount. I’ve mixed a number of rap records where the vocals are lower in the mix, but never have I thought it was a good idea.

Generally I want the vocals to be equally as strong as the drums or stronger, and I want them as “forward” as possible. Compression is usually a part of that equation.

Optical Compressors

The smoothest way to get those vocals forward is through optical compression. The rounding quality of the attack and the unique shape of the knee in an optical compressor makes them ideal for vocal work.

Examples of optical compressors would be the CL1B, the LA2A, LA3A, your stock Logic compressor has an optical mode, RComp has an optical mode – and don’t quote me on this but RVox has an “optical” sound to it, as does the “smooth” setting on the UBK-1.

One of the advantages to opticals is that they tend to have easy access. Many have just one knob to control the degree of compression.

Attack And Release Time

Of course you’re not limited to simply optical compressors or fixed time settings.

Many other compressors work very well for rap vocals – in fact, any decent compressor can yield great results if set properly.

The key is setting the attack and release times appropriately. People will suggest milliseconds or time ratings for the best attack and release for vocals but the issue is that 300 ms on one compressor might give you the same results at 75ms on another.

So, I’d rather advise your compression technique based on the expected results. Your goal is to pull up as much of the “sustain” of the voice – the weight of it – while minimally affecting the articulation. Taking notes? – It’s about to get heavy.

When dealing with the articulation of the words, you’re primarily gauging your attack time. There’s generally a substantial range of attack speeds that work for vocals. What you don’t want to do is set the attack too short, or the shaping of consonants will be blurred. Nor do you want to set the attack too long, because you’ll allow the consonants to poke through too hard. So you want to find a middle ground.

A good way to experiment is to temporarily pull the threshold down a little farther than you normally would and find your attack setting that way, as the effect of the attack time will be more exposed with the lower threshold.

With the release time, my goal is to pull up as much of the body of the voice as possible. So I’m going to set the release on the faster side. I don’t want the voice to distort or become unnatural sounding, but I want as much body as possible before I get to that point.

In terms of both the compressor ratio, and the release time, I tend to be a little more aggressive with rap vocals than “softer” music. The “integrity” of the vocal sound is not really as important as the prominence of it. For a more relaxed, natural sound I might do a medium release and 3:1 or 4:1 ratio.

For a rap vocal I’m going for a pretty quick release, and I’m doing 4:1 up to 8:1. Rap isn’t really supposed to be “pretty,” so I don’t worry if the compression becomes a bit audible.

Thicker Vocals

Another great use for compression on vocals is to make the vocal sound thicker – particularly in rap. Rap is frequently recorded in home studios, even by big name artists. And home studios rarely produce the thick, full vocal sound that one can get at a professional facility.

So being able to thicken and give weight to a vocal is an extremely important skill. In order to do it right though, you need a little more than compression. You need an EQ to make sure the vocal is as even and smooth as possible. Then you need some “saturation.” Saturation is just a nice name for friendly distortion. Saturation moves and enhances the harmonics of the vocal. Over saturating will sound like crud, but just the right amount gives the impression of a richer sound.

So the formula is – get the vocals sounding clean, saturate to get the vocal sounding richer, and compress that signal. You have to tread carefully though as over EQ’ing, over saturating, or over compressing will make your vocals sound horrible. Unless you do all of that in parallel!

Parallel Compression

Parallel compression basically means making a copy of the signal, compressing the snot out of it, and then blending that parallel signal back in with the original signal.

The advantage here is that you can get really liberal with the effects, and just blend it in to where it doesn’t sound unnatural. This is great is you are trying to fill-out frequencies that weren’t really there in the original recording, because you can really saturate and compress the parallel signal and generate some very consistent dense harmonics.

Then just blend that in until just before it starts to sound too effected. One of the reasons I really like the UBK-1 for vocals is because it gives you a saturation stage followed by a compression stage and the ability to blend both in parallel.

Distortion Free Equalization

Lastly, compression can be used to tame frequencies without the artifacts from EQ. Often with vocals you’ll have moments where the vocalist changes their tone.

A common example is vocalists will often become more midrangy as they project because they tighten their neck and push more air through their nose. If you simply notch out some midrange – that might work, but it might also take away some of the energy of the overall performance, or some of the frequency information you need to make the vocal stand out.

A good alternative is to use a compressor with either an adjustable side chain, or an external side chain input. The side chain signal is what the compressor reacts to – so if you can EQ the side chain to target problematic frequency areas, the compressor will “intelligently” react to those tones and pull them down.

Conclusion

Compression is a powerful tool that many people struggle to fully understand, so try to get your hands on one and start experimenting. As always I’ll keep an eye on the comments in case there is anything that needs clearing up. I also encourage you to share your own compression tips here!

Matthew Weiss records, mixes, and masters music in the Philadelphia, New York, and Boston areas. Find out more about him here.

Be sure to visit the Pro Audio Files for more great recording content. To comment or ask questions about this article go here.

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Tuesday, April 17, 2012

In The Studio: The Pros And Cons Of Mixing On Headphones

Getting a reference point as to what sounds good or bad

 

Christine Lavin recently sent in the following question: “Where do headphones fit in the mix for you, or do you listen just to the speakers? I’ve been in studios where they have 3 sets of speakers—small medium and large—and the producer or mixer (if it’s someone else) listen on all three. But I’m a headphone person. Wonder what your opinion of that is?”

I’m a firm believer that you can get great mixes out of virtually any set of speakers in just about any room, and that includes headphones as well.

The trick is that you have to have enough listening time to get a reference point as to what sounds good or bad when you play your mix back elsewhere.

That’s why mixers began to take their own speakers wherever they went (or asked for NS-10s) in the first place.

It was something that they were familiar with, and since they were nearfields, the room didn’t come too much into play during the mix so they could be surer of the result.

Mixing on headphones does have 4 significant downsides though:

  1) You can’t wear them for as long as you need to (8, 10, 12 hours) before your head and ears get tired from the extra weight.

  2) You have a tendency to turn them up, which can lead to some quick ear fatigue, again limiting your ability to mix for long periods.

  3) Because most of the more expensive professional headphones really sound great, you get a false sense of what the mix is like (especially on the low end), and it causes you not to work as hard getting the frequency balance of the mix right.

  4) The vast majority of the audience won’t listen on phones after the mix is completed. Since a mixer is always aiming for a mix that sounds great on most speakers that the material is played on, you want to stay in that realm if possible, and even listen on some crappy speakers if possible as a check. Headphones just sound too good for that.

That said, headphones do have their place. They’re great for editing in that you can hear clicks, pops and inconsistencies that you may otherwise miss while listening on speakers, and they’re a great check for panning and checking reverb tails when mixing, but I wouldn’t personally use them for an entire mix.

But if you’re mixing in your bedroom and don’t want to wake the the kids, significant other or neighbors, then by all means, go for it. Just make sure that you listen to some other material that sounds great on speakers first so you have a reference point of what sounds good and what doesn’t.

Here’s a bonus question also from Christine: “Do you think it’s optimum to mix in a different studio from where you record, and/or bring in someone with fresh ears who hasn’t been there for the recording of all the tracks?”

The only time I think that it’s worth going to a different studio to mix is if you think that your mix is going to sound a whole lot better as a result, like in the case of mixing through a console rather than in the box.

Otherwise, I’d stay in the same studio that you tracked, if for no other reason than not having to worry about software and plugin compatibilities that always seem to crop up when you go to a different system.

As for getting a different mixer, if you’re stuck on the sound of the project for some reason, then it’s great to have a fresh set of ears (as in someone else’s) for the mix.

Also, if you can hire someone who’s a great mixer who you think can beat anything that you might come up with, then it may definitely worth it (“may” is the operative word here - you might not get what you expect). Otherwise, I think you’re a lot better off using the same team that tracked the song(s) from the beginning for the sake of vision and continuity.

Bobby Owsinski is an author, producer, music industry veteran and technical consultant who has written numerous books covering all aspects of audio recording. For more information be sure to check out his website and blog.

 

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Monday, April 16, 2012

Why Measured System Response Doesn’t Match What’s Heard

What you see isn't always what you get...

Many audio field technicians are now in possession of measurement systems that can be used to assist the listening process in equalizing sound reinforcement systems.

But, they’re often surprised to find that the measured system response correlates poorly with subjective impression of how the system sounds.

In other words, the system can sound good when it looks bad on the analyzer, and it can sound bad when it looks good on the analyzer.

As a result, some users have become frustrated and distrustful of analysis systems in general.

Let’s look at why the eye and ear do not always agree on what is best regarding the response of the sound system.

First, consider the most popular methods of measuring the response of the sound system.

By “response,” I am referring to the magnitude of the frequency response as displayed on a dB (vertical) vs. logarithmic (horizontal) scale. The goal of technical system equalization is to produce a “flat” horizontal line on this display.

WORKING IN REAL TIME

The real-time analyzer (RTA) is essentially a bank of meters, each driven with a 1/n-octave constant percentage bandwidth filter so that only the level of a limited range of frequencies is displayed by each meter.

The original RTAs used analog meters, but current versions use a vertical row of LEDs for each 1/n-octave band. One-third octave resolution is the most popular, and correlates well with the response of the human auditory system. 

The RTA input is fed from an omnidirectional test microphone located at a listener position. Omnis are used because they typically have a very flat, “benign” frequency response over most of their band pass.

RTAs can also be software-based, utilizing the sound card on a personal computer to provide the A/D conversion of the microphone output voltage. A mathematical algorithm (the FFT) is used to produce the previously described dB vs. frequency display.

These “digital” analyzers emulate their analog counterparts in how the information is displayed, but differ in that the filters and display is the product of a computer algorithm rather than analog filters. This type of RTA is more versatile, as the octave-fractions, colors, etc. are under software control.

Regardless of which type is used, the standard method-of-use is to drive the sound system with pink noise (equal energy per 1/n octave) and adjust the system equalizer for a “flat” magnitude response on the analyzer display.

RTAs are powerful tools when certain guidelines are followed, but indoors they can indicate a system response with poor correlation to what the listener is hearing. The major consideration is the placement of the measurement microphone.

If the mic is placed in the near field of the loudspeaker (typically less than 10 feet), the correlation with human hearing is pretty good. At this position, the direct energy from the loudspeaker dominates what is being observed on the analyzer and very little of the reflected energy from the room is included in the displayed response. Adjustment of the equalizer for a flat direct sound field on the analyzer produces a desirable result.

The down side to the near-field placement is that the measured response is very sensitive to small vertical movements of the microphone when the loudspeaker has offset vertical components (as most do). This sensitivity can be reduced if the microphone is moved to a greater distance from the loudspeaker (into the far field) since the path-length difference back to the individual components becomes more equal.

But, as the microphone is moved further away, the reflected energy from the room begins to dominate the displayed response.

GIVING EQUAL WEIGHT

Microphones have no “perceptual” abilities. They do not localize sound or discriminate early sound energy from late energy like humans do.

A listener at a distance remote from the loudspeaker will pay more attention to the direct field of the loudspeaker than sound that is building-up in the room.

A microphone gives equal weight to all energy without regard to where it is coming from.

A simple experiment to verify this is to stand at the microphone position and listen to the loudspeaker and then route the mic through a headphone amplifier and listen to it through headphones - not the same thing at all.

Low frequency sounds tend to linger in rooms longer than high frequency sounds, because most rooms have more high frequency absorption than low frequency absorption.

As such, the room becomes “bass heavy” when the total sound field is considered. This extra low frequency information will dominate what is observed on the RTA, and the knee-jerk reaction is to attempt to “flatten” the response by boosting the high frequency bands on the equalizer.

The result is a system with excessive high frequency output and a resultant “harsh” sound quality.

When RTAs are used in this manner, it is important to equalize to a “target curve” rather than for a flat frequency response. The popular “X” curve for theaters is flat to 2 kHz, where it starts rolling off the high frequency response at about 3 dB per octave. It is -10 dB at 10 kHz relative to 2 kHz.

This represents 1/10th power at 10 kHz relative to flat response. The one-third-octave analyzer and the target curve have served sound practitioners well for years, and remains a viable approach to system calibration.

RECENT METHODOLOGIES

Technology has yielded some new methods for acquiring the system response at a listener position. A complex comparison (both time and frequency information) of the input and output of a system is called the transfer function. It includes both the magnitude and phase response of the loudspeaker/room at the microphone position.

This has become a popular method of analysis, as it allows any input stimulus to be used to test the system, since the displayed response is just the difference between “what you put in” and “what you got out.”

Transfer function analysis has the added advantage of the ability to use a “time window” to exclude late arriving energy from consideration in the response. This can prevent the low-frequency build-up problem that plagues traditional real-time analysis. With proper implementation of a time window, the system response can be adjusted without the need for frequency weighting via a target curve.

A major difference between transfer function analysis and 1/n-octave real-time analysis is that the former requires the removal of the signal delay between the two signals being compared. The stimulus (the reference signal) always has a much shorter path back to analyzer input than the output of the measurement microphone. Sources of delay include the travel time through the air and the latency of digital processors.

Failure to properly synchronize the reference signal and the microphone’s signal will result in an erroneous display of the system’s response. The length of the time window must also be selected - in other words, “how much of the room decay do I want to include in the response?”

Unfortunately, there is not an optimum size for the entire spectrum. A short time window excludes much of the room decay at the expense of low-frequency resolution. A long time window improves frequency resolution at the expense of gathering too much of the room’s decay. A compromise is required.

The human auditory system perceives pitch on a proportional (logarithmic) frequency scale. This is one reason that we use constant-percentage bandwidth filters for tuning audio systems - the bandwidth grows with increasing frequency.

Frequency-dependent bandwidth suggests that the length of the windowing function used in transfer function analysis should be varied in the same manner - a decreasing length with increasing frequency.

This produces a somewhat “anechoic” response at high frequencies with increasing frequency resolution as frequency decreases.

The time window length is a function of frequency, with even the longest window (highest frequency resolution) excluding much of the late energy from the room.

Another caveat of this type of analysis is that much greater frequency detail is possible than with the typical 1/3-octave banded display. Phase interference effects from reflections or multiple drivers are clearly visible on the analyzer.

Such anomalies are almost always position-dependent, so careful “corrections” at one seating position will be inappropriate for another.

Both the loudspeaker and the measurement microphone should be carefully positioned to avoid the creation of very early high-level reflections.

SPECIAL EFFECTS?

The “floor bounce” effect is a common example of a very early reflection (typically within a few milliseconds of the first sound arrival) that produces a unique acoustic response for each listener seat for all but the lowest octaves of the spectrum. This is an example of “less is better” when measuring the response, as a 1/3-octave display lacks the resolution to observe the effect in detail and produces less of a temptation to “fix” it.

The floor bounce effect can be minimized by use of an appropriate frequency-dependent time window or by simply laying the measurement microphone on the floor, or on a board placed across the listener seats. The effect usually disappears with the presence of an audience, so we do not wish to consider it when tuning the sound system.

The use of variable-length time windows and the synchronous transfer function allow the system to be tuned in a manner similar to the near-field RTA method (flat response on a log frequency display), even at remote positions in the room. It is superior to the RTA method in that the effects of air absorption are readily apparent and can be compensated for via equalization. Near-field techniques do not include air absorption for the simple fact that the sound has not traveled very far before it is picked up by the microphone, so it hasn’t passed though enough air to be significantly attenuated.

By far, the biggest problem with tuning sound systems is failure on the part of the technician to recognize anomalies that cannot be corrected with equalization.

The equalizer is a “global” device, meaning that its response curve will be impressed on all of the sound radiated from the loudspeaker, regardless of the direction in which it is radiated.

Many, if not most, of the anomalies observed on the analyzer are unique to each listener position. The technician must learn to recognize and ignore such events. They include:

—Floor-bounce effect

—Interference between multiple drivers

—Reflections from objects near the mic or loudspeaker

Events that produce a more global effect, and can therefore be addressed with equalization include:

—Boundary-loading of loudspeakers

—Coupling between multiple low-frequency drivers

—The direct-field loudspeaker response

With training and experience, the system technician can implement methods that reveal system imperfections that are correctable, and hide those that are not - regardless of the analysis method used. Better yet, system designers can design systems with fewer “un-equalizable” problems.

BAD IS ALWAYS BAD

The old adage “An ounce of prevention…” could never be truer.

System equalization then becomes meaningful and fast, providing the “icing on the cake” of the performance of a sound system.

It makes a good loudspeaker sound better, and brings the system to its fullest potential given the acoustic environment into which it is placed.

A bad room is a bad room, regardless of how we process the electrical signal that drives the loudspeakers.

When used properly, the traditional 1/n-octave real-time analyzer is a useful tool outdoors at any distance. Indoors, the effects of reflected sound and non-frequency-uniform room absorption produce some problems for this method at measurement distances remote from the loudspeaker.

One solution is to utilize a weighting curve that reduces the target level of the high-frequency portion of the spectrum. Attempts to achieve a flat system response at remote listener positions without the use of a weighting curve can result in harsh-sounding systems and even component damage.

Transfer function analysis addresses some of the shortcomings of the 1/n-octave RTA, but it requires greater expertise on the part of the user. Failure to properly compensate for the time differential between the reference and measured signal can produce wildly erroneous results.

The time window length must also be selected by the user, and different lengths will produce different displayed responses. A frequency-dependent time window produces a display that correlates well with human perception.

The most important feature of either measurement method is a knowledgeable operator - one who understands the caveats of each approach along with the basic characteristics of the human auditory system.

None of the questions raised here have a single, correct answer. This means that experience, good judgment, and common sense rooted in Newtonian physics are still a part of the measurement process.

Sound is a relatively easy quantity to measure, but measurements that correlate with human perception are much more difficult. Analyzers driven by omni directional microphones do a poor job of emulating the human listener. At this point one could ask, “So why measure at all? Why not just listen?”

Next time, we’ll have a look at this provocative question.

Pat and Brenda Brown own and operate SynAudCon, the leading independent professional audio education source, with training sessions held around the world and online. For more info go to www.synaudcon.com.

 

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Friday, April 13, 2012

Church Sound: The Art (And Necessity) Of Compression

Helping to keep sources in place in the mix

A while back, I got the rare opportunity to work with the youth band at our church.

These guys have an incredible heart and passion to worship and have loads of raw talent which translates into a powerful time of worship.

When they lead, as a worshipper I feel free and emboldened to praise God the way He created me too.

When they lead, as a sound operator I have to work as hard and quick as ever to create a decent mix to help facilitate that worship.

More Compressors Please

As is the case with most youth bands and even many churches, they’re not using state of the art or high dollar gear for their services.

Now don’t get me wrong, they are not operating with the bare minimum. The system includes an Allen & Heath console, JBL loudspeakers and subwoofers, and solid system and signal processing. But what I longed for that night was individual compressors.

Keeping The Vocals On Top

Maybe this never happens to you, but in a mix including three vocals, an un-caged drum set, two electrics, acoustic, bass, and keyboard, I had a hard time keeping the vocals out on top to lead the group in worship while keeping the music strong.

The vocalists on the team are gifted in leading worship, but for a variety of reasons (key of the song, dynamic range, mic etiquette, etc) their volumes were all over the place that night and the second I took my finger off their faders I would either lose them or have way too much of them.

With seven stage monitors, acoustic drums, three amps and a very small stage, I was dreaming for a few compressors to help me layer the mix the way I wanted.

Most Useful Tool

The compressor is one of the sound man’s most useful tools - yet I’m always surprised how few people seem to understand and know how to effectively use this critical piece of gear. I would like to help a few more of you get comfortable using compressors.

What Is Compression Really?

The clearest definition of compression that I’ve ever seen is this: “Compression is the art of making louder parts of a composition appear softer, and conversely, the softer parts appear louder.”

That night, if I would have left the lead singer’s fader in one spot for the entire night his volume alone would have ranged anywhere from 85 dB to 120 dB. Alright that might be an exaggeration but he got loud.

When he was closer to the 85-95 dB volume he could barely be heard over the drums and guitars. Neither end of the spectrum is really acceptable in a good mix, so compression comes along and makes it possible to narrow down that volume range to make things more mixable.

Example: How A Compressor Works

Let’s say I have a 20 dB range between a vocalist’s quiet singing versus their loudest singing. With a compressor I can take that 20 dB range and make it as small as a 1 dB range, but since I don’t want to eliminate the artistic dynamic range that the singer is using to create the mood or feel of what they are singing, I can get that range down to a very manageable 5-8 dB that will make mixing significantly less complicated but still leave some of that dynamic in place.

So how do we get our compressor to do that? With some understanding of the compressor’s settings you can be on your way to a smoother sound and a less stressful time behind the board.

Control Elements

Threshold. In simple terms the threshold is the point where the compressor starts to do its thing.

Since there is a wide range of compressor and mixer brands I’m going to talk about these settings more generically as opposed to using the numbers on the knob. If the input meter on your console (let’s say negative infinity to +15 dB) matches that of your compressor, things will be a little clearer as the numbers will match.

You must first set the gain (or trim) of your channel on your mixer (on my regular console that is around +3, or typically where the green lights first turn to yellow or maybe the yellow light just starts to glow on the meter).

Now if your numbers match, and your vocal meter is showing signal between -5 and +10 dB, I’d start with my threshold set close to 0 dB. If your numbers don’t match, once your gain is set turn your threshold knob and find the area where the gain reduction knobs just come on.

Begin with your threshold there and if you find it’s not compressing frequently or soon enough you can lower the threshold from there to make it kick in sooner.

Ratio. This one is a relatively simple concept. The ratio simply says for every “X” dB the source goes up in volume, the compressor will only let the output go up “Y” dB.

For example, if you set a vocal mic with a 3:1 ratio, for every 3 dB the vocal increases coming into the board, the output will only increase 1 dB.

You can think of the ratio as setting the size potential of the source. If you want it to be able to go bigger, you can leave your ratio smaller. If you want it to stay a little smaller, or perhaps be more under control, you can set your ratio higher.

I tend to start with a ratio of 3:1 for most vocals and guitars, and often times I will go 4:1 or even 5:1 on drums or very dynamic guitars.

My preference is to start low and if you need more compression (less range) you can always increase the ratio. The reason it is my preference is simply this, I don’t want to take away any more control from the musicians than is absolutely necessary to make the mix work well.

If I start it at 5:1 when 3:1 will do and don’t adjust it down, I may be holding that source back. If I start low and it’s still too big, I can easily adjust my ratio up.

Attack. The attack is how quickly the compressor responds to the volume change. A slower attack will sound a little smoother, rounding out the sound of your source a little bit and in essence making it sound a little “fatter.”

A slower attack will generally be less noticeable which can be good for vocals and some thin or scratchy guitars. Setting your attack to a faster setting can be great for instruments such as drums or any other very aggressive instruments.

A faster attack will give an instrument more of an aggressive, pumping feel, and potentially bring out more of the high end edginess. The ultimate decider on where to set this is by listening. I tend to set vocals a little slower, guitars in the middle, and drums faster to start.

From there, if you need a little more aggressiveness or snap you can speed up the attack, and if it needs to be a little smoother or fuller you can slow it down. As in all things with sound, let what you hear guide your settings and adjust until you are happy.

Release. The release is the back side of the attack, and sets how quickly you want to release the compression once that loud burst is over. As with the attack, a slower release will sound smoother and less noticeable but could end up taking some of the aggressiveness out of aggressive instruments by compressing them when they don’t need to be. I again will tend to start a little slower for vocals, middle of the road for guitars, and faster for drums.

You’ll want to again experiment with where to set this by listening to the sound. If the source sounds like it is pumping a little bit, slow the release down to help even it out a bit. If it feels like you might be losing something on the next note/beat, you likely need to speed the release up a bit.

Again, let the sound of the source guide you to where it should be set. Listen and adjust until it sounds right to you.

Output. Most compressors have an output to help boost the volume of the end result, and here’s where I tend to see a lot of mistakes made.

Now that you’ve taken that 85 to 105 dB vocal and compressed it down to a manageable 85 to 93 dB, you may need to increase the output a little to get it over those guitars and drums. Instead of reaching for the gain or trim knobs (which would then bring more signal into the compressor and would change how you’ve set your compressor), if you add 5 dB of gain to your output you just took that 85-93 dB and made it 90-98 dB.

Especially Useful in Worship Environments

Compressors are a huge help to the sound man and used right they will help you get great sound out of your sources and give you the ability to get the mix where you want it. They’re especially useful in the worship environment where the voice of those leading the worship must always be present but not piercing, where more and more guitars are being used to lead the music but can’t overtake the vocals, and where many churches use acoustic drums.

Wrap Up

I truly believe that no one setting is right for any vocal or instrument. If you start with a lower basic setting and then adjust based off of what you are hearing, your compressors can give you a great edge to get your mix balanced and layered according to plan. Just remember, you don’t want to compress something more than you need to. If you’re having trouble keeping a source in it’s place in the mix the compressor is the tool to help you make that happen.

Duke DeJong has been involved in live production for over 15 years, has spent 10-plus years in full time ministry, and in 2011 began serving as the church relations director for CCI Solutions. You can find him online at dukedejong.com or on Twitter.

 
Check out more from David McLain at the Church Soundguy blog.

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In The Studio: A Boring (But Effective) Way To Avoid Embarrassing Mixes

It always pays off when you remember to do it

 
You know the feeling…

You pop in your latest mix to play it for a friend.

And as SOON as the mix starts playing, you want to crawl into a hole and disappear. “I swear this sounded better in my studio,” you tell yourself.

You start making excuses, trying to figure out why this mix you were so proud of 60 seconds ago is now embarrassing the crap out of you.

We’ve all been there.

Heck, I still go there occasionally. :>)

And sadly there’s no magic pill to prevent it from happening, but there IS one technique that works, although it’s kinda boring.

What is it?

Use a reference track while mixing.

See? I told you it was boring. But stick with me…

(If you’re unfamiliar with this, I’m referring to comparing your mix to a professional mix WHILE you’re mixing.)

This is probably the single greatest tool for snapping yourself out of the clouds of “I’m the best mix engineer to ever grace this planet” to “Oh, crap, this mix needs some serious work.”

Why? Because it IMMEDIATELY lets you compare your precious mix with a professional mix, one that you KNOW sounds good.

Pick a song that you love, one that has a similar sound to the one you’re working on, and make it a habit to constantly flip back and forth between your mix and the reference track.

This can save you hours of working in the wrong direction, and it can help you get better mixes.

All you have to do is make your mix sound like the reference mix. Easy, right? :>)

No, it’s definitely not. But without a reference mix, you’re simply mixing based on what you THINK sounds good at the time. The reference track gives you a direct line back to reality. It keeps you grounded.

There are a bunch of ways to pull this off. Here’s how I do it. I loop a song in iTunes, then run it to a stereo channel on my mixer. Then I run the output of my DAW to another stereo track.

Then I just use the mute buttons to instantly flip back and forth between the reference and the mix.

Like I said, it’s not exciting, and it’s something I tend to forget to do, but it ALWAYS pays off when I remember.

The problem? Making your mix sound like the reference track isn’t easy, and it’s dang near impossible if you don’t know how to use the tools you have available to you.

Tools like EQ.

Master the use of EQ, and you’ll be well on your way to mixes that, rather than embarrassing you, will make you proud.

Joe Gilder is a Nashville-based engineer, musician, and producer who also provides training and advice at the Home Studio Corner.Note that Joe also offers highly effective training courses, including Understanding Compression and Understanding EQ.

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Why Not Wye? When Combining Two Signals Into One Is Not A Good Idea

Anything that can be hooked-up wrong, will be. You-know-who said that, and she was right.

 

Wye-connectors (or “Y”-connectors, if you prefer) should never have been created.

Anything that can be hooked-up wrong, will be. You-know-who said that, and she was right.

A wye-connector used to split a signal into two lines is being used properly; a wye-connector used to mix two signals into one is being abused and may even damage the equipment involved.

Here is the rule: Outputs are low impedance and must only be connected to high impedance inputs—never, never tie two outputs directly together—never.

If you do, then each output tries to drive the very low impedance of the other, forcing both outputs into current-limit and possible damage. As a minimum, severe signal loss results.

“Monoing” Low End
One of the most common examples of tying two outputs together is in “monoing” the low end of multiway active crossover systems. This combined signal is then used to drive a subwoofer system.

Since low frequencies below about 100 Hz have such long wavelengths (several feet), it is very difficult to tell where they are coming from (like some of your friends). They are just there—everywhere.

Due to this phenomenon, a single subwoofer system is a popular cost-effective way to add low frequency energy to small systems.

So the question arises as how best to do the monoing, or summing, of the two signals? It is done very easily by tying the two low frequency outputs of your crossovers together using the resistive networks described below.

You do not do it with a wye-cord.

Summing Boxes
Figure 1 shows the required network for sources with unbalanced outputs. Two resistors tie each input together to the junction of a third resistor, which connects to signal common. This is routed to the single output jack.

The resistor values can vary about those shown over a wide range and not change things much. As designed, the input impedance is about 1k ohms and the line driving output impedance is around 250 ohms.

The output impedance is small enough that long lines may still be driven, even though this is a passive box. The input impedance is really quite low and requires 600 ohm line-driving capability from the crossover, but this should not create problems for modern active crossover units.

The rings are tied to each other, as are the sleeves; however, the rings and sleeves are not tied together. Floating the output in this manner makes the box compatible with either balanced or unbalanced systems.

It also makes the box ambidextrous: It is now compatible with either unbalanced (mono, 1-wire) or balanced (stereo, 2-wire) 1/4-inch cables.

Using mono cables shorts the ring to the sleeve and the box acts as a normal unbalanced system; while using stereo cables takes full advantage of the floating benefits.

Stereo-to-Mono Summing Box
Figure 2 shows a network for combining a stereo input to a mono output. The input and output are either a 1/4-inch TRS, or a mini 1/8-inch TRS jack. The comments regarding values for Figure 1 apply equally here.

Balanced Summing Boxes
Figures 3 and 4 show wiring and parts for creating a balanced summing box. The design is a natural extension of that appearing in Figure 1.

Here both the tip (pin 2, positive) and the ring (pin 3, negative) tie together through the resistive networks shown.

Use at least 1 percent matched resistors. Any mismatch between like-valued resistors degrades the common-mode rejection capability of the system.

Termites In The Woodpile
Life is wonderful and then you stub your toe. The corner of the dresser lurking in the night of this Note has to do with applications where you want to sum two outputs together and you want to continue to use each of these outputs separately.

In other words, if all you want to do is sum two outputs together and use only the summed results (the usual application), skip this section.

The problem arising from using all three outputs (the two original and the new summed output) is one of channel separation, or crosstalk. If the driving unit truly has zero output impedance, then channel separation is not degraded by using this summing box.

However, when dealing with real-world units you deal with finite output impedances (ranging from a low of 47 ohms to a high of 600 ohms).

Even a low output impedance of 47 ohms produces a startling channel separation spec of only 27 dB, i.e., the unwanted channel is only 27 dB below the desired signal. (Technical details: the unwanted channel, driving through the summing network, looks like 1011.3 ohms driving the 47 ohms output impedance of the desired channel, producing 27 dB of crosstalk.)

Now 27 dB isn’t as bad as first imagined. To put this into perspective, remember that even the best of the old phono cartridges had channel separation specs of about this same magnitude.

Therefore stereo separation is maintained at about the same level as a high-quality hi-fi home system of the 1970s.

For professional systems this may not be enough. If a trade-off is acceptable, things can be improved.

If you scale all the resistors up by a factor of 10, then channel separation improves from 27 dB to 46 dB.

As always though, this improvement is not free. The price is paid in reduced line driving capability.

The box now has high output impedance, which prevents driving long lines. Driving a maximum of 3000 pF capacitance is the realistic limit. This amounts to only 60 feet of 50 pF/foot cable, a reasonable figure.

So if your system can stand a limitation of driving less than 60 feet, scaling the resistors is an option for increased channel separation.

Presented with permission from Rane Corporation.

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Thursday, April 12, 2012

Don’t Overlook These Aspects Of The House System/Tech Package

The small things can be the very things that make for smooth running, enjoyable shows

As we left off last time, our hero was fighting for truth, justice… Oops - wrong story.

Our story here is to continue the discussion of creating a truly useful technical information package for a visiting acts and sound personnel, is a bit less glamorous.

But it need not be overly complicated, either.

Previously (here), we looked at the basic format of the tech package, and talked about the first five of 10 essential items it should offer. Now, let’s lay out the second five.

Snakes and cabling. The snake routing information sheet for the venue where I work, a theatre in a large city, looks like a parts list for a 747. I’m not sure when this system infrastructure was installed, but I can tell it took a good bit of thought and labor.

The snake line sheet in the tech package is essential for everyday operation and can be helpful to house crew and traveling crew alike. Be sure to include all the oddball lines, such as those to the spot booth, stage manager offices, and places unique to your particular venue, like the bar next door or the trap door operator position under the stage.

This sheet can do double duty by serving as a place to note malfunctioning or suspect lines as well as lines that have been commandeered as temporary replacements. Permanent lines can be noted, like those for a “stage god” mic or for emergency announcements.

Even the main sound system drive lines and data lines should be noted. This can save many a frustrating moment should something fail at an inconvenient moment (as if there is a good time for a failure of anything in a sound system).

I’ve heard it suggested that keeping crucial information like this in one’s head is a type of job insurance, the thinking being that the “keeper of secrets” will be the only one able to save the day. Let me dispel this myth here and now. If the working relationship is destined to end, the fact that the poor sap getting the job next will have a tough time is not job security.

Rather, having this information on record can never hurt, and really, can only help. It might even save that job one day, not to mention a show.

Console layout. The house console at our venue is used for every type of show imaginable. From serving as a relatively simple drive signal router for shows with nearly self-contained sound systems to having every single channel chock full of orchestra inputs for a ballet, the board does it all.

I’ve found it easy to more or less permanently assign some console input strips to a single function each, no matter what the show may be. This page in the info package not only lists the permanent channels, but also has blank spaces where an input list can be created for the empty channels.

The page should be correlated with the snake line sheet to eliminate inconsistency. Any problem channels or input strips can be noted to assure that maintenance is performed. In fact, I keep several blank copies in the folder so that input lists can be created on the spot, if needed.

Backstage paging. This seldom-praised portion of the house audio infrastructure is just as essential to many performances as the sexiest loudspeaker array in the house rig. If the actors or crew cannot hear the paging system telling them their cues and calls, chaos can ensue and further, it can rain down upon your poor head. Stage managers are not enemies to make if it can be avoided.

A list of every loudspeaker in the system, as well as its location and operating condition, serves dual purposes. Visiting engineers can quickly identify areas without coverage that may need to be supplemented and the house crew can do periodic checks of the system with a handy checklist.

Touring crew will often ask for a thorough check of program and paging in the dressing rooms, if they value their paychecks.

Actors have a habit of turning off those speakers equipped with volume pots and then throwing tantrums when a call to the stage is missed.

A list of the paging system that includes the details of the installation pathways can aid in troubleshooting.

I once had a paging system partially fail in between shows, but because only one “trunk” of this 70-volt system had failed, I was able to determine that some workmen installing a CCTV system had evidently drilled in just the wrong place in one wall.

I wish I could claim that my paging system list helped save the day. The truth is that this incident is the reason for the detailed chart I have today. Lesson learned.

System and microphone inventory. This is the most straightforward part of any tech package. A simple list of the microphones and any other audio equipment in the inventory, listed with columns for serial numbers, repair status information, and any other details deemed useful will suffice.

As with the backstage paging sheet, this page can be useful for taking inventory as well as for informational purposes. Providing it to visiting engineers along with a copy of the console input list will really speed up the process of creating an input list on those shows for which the house is providing production.

Product data. I’ve downloaded and printed product information from manufacturer web sites for some of the newer and lesser-known pieces of equipment in our sound system. The idea here is for anyone who is unfamiliar with a specific speaker or processor in the rig to be able to flip to the back of the folder and review details like power handling or frequency response of speakers.

These are the details that have no other place in the kind of “quick glance” overview found elsewhere in the package. On-line versions of a tech package could easily include a set of links to manufactures web pages for most of the items in an inventory.

Fellow house engineers out there, as well as you touring folks - I hope that this information is useful. It’s about communication, and the more we do it, the less we all goof up, and the better we all look (and most importantly, function in our jobs).

The day-to-day operation of a theatre or rock club is probably one of the least glamorous aspect of our industry, perhaps only second to last on the list, ahead of the fine art of amplifier dust screen cleaning.

But the small things a good house sound person and the crew do when no one is looking can be the very things that make for smooth running, enjoyable shows for all concerned.

Next time, the fine art of dust screen cleaning! (Just kidding.)

Mark E.P. Woods served as head of audio for a large performance theater, and now is the technical director theatre/convention center.

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