By Dave Hill

 There are many different ways to make a gain control circuit for an analog compressor

 You can use a; tube, FET transistor, diode bridge, VCA, optical (light dependent resistor) or PWM (pulse with modulator).  All methods have good and bad points about them.  The undesired part of this can range from parts that are no longer made, sonic artifacts, extreme complexity or difficulty to build, costly control requirements, or just very costly parts.   In designing a compressor with as little artifacts as possible, the gain control choices are limited.  PWM has been used in vintage compressors and also modern devices.  If one takes that idea and uses the latest technology it is possible the build a compressor with very little negative sonic artifacts.  It should be noted that all electronic devices will add some distortion to the audio path.  The ear hates some of these distortions and likes others.

 The basic idea is that audio is energy, in electrical form when you are inside a piece of gear.  In a compressor the gain control method needs to reduce this energy.  A VCA lets a percentage of the energy through, determined by a control voltage (it is a variable gain amplifier, or voltage controlled amplifier).  The problem is when the control voltage is changed there are undesired things that happen.  The control voltage can leak into the audio path and the distortions that the VCA generates when the control voltage changes.  The result is a dynamically changing distortion, plus control leakage adding up to an ugly sonic artifact.  The faster the VCA needs to work, the worse this can become.

The Crane Song STC-8 Compressor uses a PWM gain control circuit

 If you could operate a switch at a high enough speed you would be able to control the average energy on the output of the switch. This energy could also be fed into a simple circuit that would help make the average more accurate.  If the switch was on 50% of the time and off 50% of the time, there would be 50% of the energy at the output of the switch.  If it was on 10% of the time and off 90% of the time you would have 10% of the energy at the output of the switch.

 The modern PWM gain control circuit design has the advantage of being able to use current technology, very high speed parts.  They were not available 20 years ago or were very costly and power hungry.  Components that can switch between off an on in less than 1nS, that are suitable for build a switched gain control element are now available. 

 With the switching element selected, a circuit needs to be built that will turn the control voltage into a variable width on-off switching command.  This circuit is known as a pulse width modulator.  In a PWM the width of the pulse, or the on time to off time ratio changes with the applied control voltage.  With very careful circuit design and printed circuit layout it is possible to build a very accurate system with very low artifacts. The artifacts and be in the range -118 db with respect to the maximum signal level.

 There are other decisions in designing a compressor; how complex the side chain or detector circuit is and weather it is a forward feed or feedback design, complexity and circuit types can be interrelated.

 In a feedback design the output of the compressor drives the side chain and then the gain control element.  If the circuit is well designed the compressor will work very well an act like a self adapting or self correcting gain control device.  The circuits must be very fast, for a fast acting compressor, this is not easy to do.

 The forward feed design requires calculating the desired gain before gain is changed.  It requires a high degree of accuracy in this calculation and in the control voltage / gain reduction relationship, or law of the gain control element.

 If a compressor’s control circuit does not perform well, it will limit the performance of the compressor.  But with out a very good gain control circuit, it is not possible to build a very good compressor.  Pushing technology to its limits is a good thing.  Copying vintage designs does not advance the sonic world.

Mark Knopfler with his SCM25a

By Brad Lunde

What makes a studio monitor sound great? Does higher price always mean better sound?  What about the parts used in manufacturing?  Is it the company that made it or the design of the speaker?  Buying a great pair of speakers can be a difficult step when there are so many conflicting ideas about what constitutes better sound. So often people just go with a brand they think is supposed to be good, shove them in a room and hope for the best. Then, if they don’t sound right-well…then it’s just a mystery! 

When I send a set of monitors out for a demo, I KNOW that it will go well if the room sounds good and bad if the room has some problems.  The user never realizes that the room’s sound is why the speakers don’t seem right; they are convinced its the speakers fault and they only need different speakers!  Better sound always means a better room and a speaker system is the combination of the speaker + the room together.

ATC300ASLs at Front Stage

Professional recording studios spend a fortune on a proper design because rooms will never sound great unless you work at it.  For everyone stuck with an office space, a spare room at their house or make shift studio, there’s some work that has to be done. The room will impart a sound to everything inside it, from monitors and microphones to pianos and guitar amps.   A performance recorded in a bad room will never truly sound great, no matter how good the gear.  But, once you get the room sounding good, everything else seems to fall into place. More so than a lot of things, investing time and effort on the room itself will pay off in big way!

Sonodyne SM100Ak at Sphere

Some simple things to do are to start with the basics and listen carefully to your room.  Play back some full range audio and walk the perimeter. Stop in the corners, stand high and squat low.  Listen carefully how the room sounds differently in different spots. Knowing where the bad spots are can help you avoid placing any transducers (amps, speakers or microphones) there in the future.  Avoid hard parallel surfaces if at all possible, as your sound bounces between them like a ping pong ball.  Break that bounce up with acoustical panels covering one or both parallel walls.  Put a thick rug over a hard floor to stop the bounce between floor and ceiling.  If the bass is very boomy in spots and non existent in others, you probably need some bass traps, something you can build or order from an acoustic panel company. These will help to smooth out the low end response.  If your speakers are on the meter bridge, put them on stands above and behind the console – the console face is reflecting midrange back in your face and changing the way the speakers sound.  If your speakers are near side walls, put absorption panels at the first reflection points (the spot where the speaker sound first hits the wall) to improve the imaging and response of your speakers.  Also put absorption panels behind the speakers. You’d be amazed just how much acoustical energy exists back there.

One of the simplest solutions is to buy a complete room kit from a number of quality acoustics companies, such as  RPG, ATS or GIK.  They are a good investment and a step in the right direction. With a little TLC, you’ll be amazed how much better your monitors can sound!   

-Brad Lunde

RA is a Nonlinear TDM plug-in for the professional Audio Engineers that demand the most from their plug-ins. The plug-in can be thought of as an amplifier that is being over driven, but there is control over what part is being overdriven.

The controls in RA are divided into 4 groups:

-The first and most useful control is the LOW LEVEL control. This control brings up the low level content of the audio. It may also be thought of as a detail control. It functions like an amplifier being over driven, but works more on the low level part of the signal and less on the high level part. This control generates third harmonic distortion. The LOW LEVEL control can also have a multiplying effect on other plug-ins by increasing their effect. The LOW LEVEL control is useful on most any kind of signal and is a little like upward compression with out attack and release times.

-The second control group is the PEAK controls. They are like an adjustable soft clip, or an amplifier where you have control over how it saturates, but leaves the low level part of the signal alone. The distortion generated by the PEAK controls is third harmonic. The PEAK control rounds off the peaks of the wave shape and has a hardness value that can be changed. The HARDNESS control’s soft setting (0) limits the peak to approximately -3dbfs and the hard setting (100) limits peaks to approximately -10dbfs.
In a test some people did not hear 50% distortion if it is only second harmonic. But it does change the feel of things and can have a thickening effect. The TOP control rounds off the top of the wave shape and the BOTTOM control rounds off the bottom of the wave shape. When used together it is possible to have the even harmonic content cancel and have third harmonic distortion left.

-The final group has two controls are more like house keeping. The first control which is in front of the signal processing is a DRIVE control that has a plus and minus 6 db range. Being the distortion generator is level dependent, this will allow for a more or less aggressive operation of the process and the ability to better match the process to the track. The DRIVE control is also unity gain from the input to the output. However if you have a -1dbfs signal and increase the drive gain to +6 db you will clip and the clip light indicator will turn red.

-The last control of the plug-in is an output level trim which has a range of plus and minus 6 db for matching gain. Clipping the signal with the trim control will light the clip light indicator, but the coloring controls cannot clip. The main use of the LEVEL TRIM is for level matching, being the curve bending is very non linear it is not possible to have a fixed unity gain.
The graphic display shows how the signal is being distorted and is useful in the prediction of what the plug-in will do. The controls are interactive which means each one will affect the process in front of it. You could almost think of the curve display in a graphics program.

Test drive Dave’s latest invention, download a free 30 day demo at: http://davehilldesigns.com.

Stories have circulated for years on what was used to mix “so-and-so’s” hit record. Was it a Fairchild 670 on the two-mix? A Neve 33609? Or was it some cheap unheard of box that someone had in a closet? While these questions are never ending, there are some companies that are not only part of this lore, but are also still in business today and making versions of the same products they did 20+ years ago. Solid State Logic is one of these companies.

The now vintage SSL G Series consoles are considered (by some) to be some of the best sounding consoles SSL ever produced. For others that aren’t a fan of the SSL “sound”, there is still little argument about the impact and longevity these consoles have had on music production. One of things that a lot of users have loved about these consoles is the G series compressor in the master section of the console. It produced a sound that till recently was only obtainable if you owned an SSL console. But nowadays SSL and other companies have branched out and offered this compressor as a single dedicated piece of outboard gear. First it was Alan Smart-a former employee of SSL-and later several companies followed suit. These units have maintained a hefty price tag for several years. Even the G Series stereo compressor for the more recent SSL X-Rack series requires several thousand dollars for the price of admission.

Recently Chameleon Labs introduced the 7720 Stereo Compressor. It’s a single rack space unit, utilizing “THAT Corporation” (formerly DBX) VCA circuitry. Additionally, it has a high-pass filter built into the detection circuit. So…how does it sound? Below are two identical sections of a song. One mix is processed through the Chameleon 7720. The other is processed through the SSL XLogic G Series Compressor. Both are receiving between two to four dB of compression, with a ratio of 4, attack time of .3 and the release being set to “Auto”. The SSL has a street price of $3595 while the Chameleon has a street price of $629. Chameleon labs motto is “Value Conscious Audio”. See what you think!

SSL XLogic G Series
Chameleon Labs 7720

First off all, what exactly is “Boutique Audio”? Well, I’m glad you asked. (if you did) Boutique Audio is best described as smaller professional audio companies that, at their core, is driven by inventor(s) and/or their vision/products. Their staff can range anywhere from one or two people to 30+ employees. Some companies that fit this description handle everything from product design through manufacturing, sales and distribution. Others simply invent the products and outsource the rest of the day-to-day business activities to third parties. The later approach is more common but in different degrees. Now, the other part of this definition is cost. Any company designing quality products with expensive parts (welcome to the “pro” in pro audio) and manufacturing small batches of them at a time will be more expensive than bigger companies that are are manufacturing large quantities overseas. Most of us equate the rise and popularity of these companies around the same time ADAT machines came on the scene and then really kicking up with the advent of the inexpensive Pro Tools systems at the turn of the century. Compared to today’s standards this seems like an appropriate timeline, but some of the companies that meet this definition have been manufacturing product for 25 plus years! (like GML and Tube Tech to name a few)

So, back to the my title and the original question: What makes these companies successful? Also, why do thousands of people every year choose to buy $2000 microphone preamps instead of a $1000 16 channel mixers? Well, as I touched on before, quality is certainly a big factor. But, let’s be realistic: there are a lot of companies out there today trying to get your money. Actually, so much so it seems like we’re a little over saturated with some of these “flavor of the month” companies. So what’s the secret to quality and longevity? The answer is the people. If you look at the veteran companies that have been in business for many years you’ll see quality focused people that believe in their products and even more importantly, they believe in integrity. These are people that believe in taking care of their customers, always doing the right thing and not making business decisions that jeopardize anyone in any part of their business.

So the next time you get out your check book and are wondering if a piece of gear is right for you, do your homework! Once you get past whether or not you like what the product does, dig a little bit deeper. Who are you buying from (the manufacturer)? How long have they been in business? What’s their personal reputation (look at online forums, reach out to their customers, etc.)? Have they done right by others (who have they done business with in the past)? Does it look like they’re going to be in business a year from now? How many products do they have on the market? Here’s something to consider: If it takes five years to bring one product to market and they don’t have any new products on the horizon, you have to wonder if they going to be around to service and update your purchase two years from now!

TransAudio Group is proud of the companies it represents and was founded on the philosophy of working with good people (first) that make good products (second). If you ask any of the above questions of the companies that TAG represents, you won’t be disappointed with the answers you receive.

Much like vintage microphone preamps, the lore around vintage microphones is a deep and treacherous pit-a long plastic hallway where thieves and pimps run free and good men die like dogs (thank you Dr. Thompson).  But, in all seriousness, the epidemic of misinformation on this topic has gotten way out of hand.   Add the forums and the continued marketing of new products that come out of a catalog from China and it’s real hard to discern fact from fiction.   The truth is, those of us that have used a lot of vintage mics know that most of the time, three or four Neumann U-47s, AKG C12s or ELA M 251s in a mic locker are all going to sound different.  Those of us that haven’t worked with a lot of the real deal are forced to believe that whatever mic we purchased (that is supposed to sound like a mic that is no longer being manufactured) is the sound of a certain generation of technology.  So, when a manufacturer claims to be capturing the “sound” of a certain mic-which “sound” are they catching?  Admittedly, there are certain characteristics that a particular microphone possesses and most of the time, that is what is trying to be duplicated in the design process of these modern day microphones.   Take the highly sought after ELA M 251 for example…

I recently sat down with David Bock (an authority on vintage mics if there ever was one) and asked his opinion on the subject.   What are the core elements that make up a 251 and how does it compare to other microphones?  David: “A great vintage 251 is a unique sum of it’s parts and design that sets it well apart from other mics, even one with many similar parts like the AKG C12.  Though made by the same company and sharing the same capsule, tube, transformer, and psu, it’s the small details that set the 251 apart in it’s sound and design.  It has a larger headgrille and more protective mesh.  This coupled with a special variation capsule and self biased tube contribute to a sound that’s easily distinguished from it’s closest cousin, the C12.  This sound is often described as having a full low end and very sweet top end, which finds it to be flattering on many things.  It also benefits from slightly higher gain and lower effective noise floor, which also contributes to it’s great versatility.”

So that being said, let’s hear what it sounds like.   Below is a link to download a Pro Tools session (the audio files are consolidated for non Pro Tools users).  The purpose of this session is to compare the sonic differences between a current production Bock 251 and a vintage ELA M 251.  The Bock design is hand built in the USA using only the finest and most appropriate parts available to achieve the sound David described above.  The ELA M 251 is a vintage microphone with a street value of around $15,000-if you can find one.  Per usual, I encourage you to listen to this session in your own studio or listening environment and let your ears do the talking.  I think you’ll hear an immediate difference that will spark some questions of your own.

Bock Audio 251 session

There’s a lot of talk these days about audio summing amplifiers. Several companies are making them and for the most part they are being presented as a go between for engineers that want the sound of an analog console without the cost and/or maintenance of the real deal. With most people mixing in or from Pro Tools these days, whose mixer has a qualified 144dB of dynamic range, the question arises of why you want to do this? The answer: coloration and distortion. On the surface that statement sounds a lot worse than it is, but the reality is that transformers or tubes introduce distortion, which in turn deliver that warm and fuzzy coloration that sum (get it?) of us love and some of us can live without. Aside from coloration of sound, splitting out tracks or stems from your DAW to one of these 16 channel boxes (most of them available are only 16 channels) has some other benefits as well. These include but are not limited to inserting analog gear across certain parts or stems of a mix and more defined separation of parts in the stereo field.

Over the past several years I have had the opportunity to test drive just about every summing amplifier on the market. Some of them have been very impressive while others have not. When I first started doing this, the biggest challenge I ran into was how to effectively do an A/B comparison between some of these boxes. If you think about it, it’s almost like trying to compare a mix done on an SSL console to the same mix being done on a Neve. It’s not really possible-they’re never going to be the same. Different mix decisions are going to be made on different boards. You could certainly put the time in and match the vibe, but to do a true A/B comparison is not technically possible. So, the idea of taking a completed multitrack mix from Pro Tools and stemming it out across a summing amplifier is not going to work. It’s going to change the sound, levels, coloration and so on. If you’re mixing in Pro Tools, there is no coloration. If you take said mix and run it through anything that has, oh let’s say a “dark” sounding transformer, your mix has just become very dull from a sonic perspective. Okay-so what if we run a finished two mix through the box just to compare the color? We could do that and it would be very telling-but, it would not show us how the box handles separation of different elements in the stereo field. Interesting dilemma, isn’t it?

So here’s what I came up with: What if we take a rough mix of a song with no automation, no EQ, no compression, no FX, etc. (just the raw tracks) and did some comparisons with those? We know what the raw tracks sound like, so if we just get some decent levels and print a couple mixes for comparison, that should give us an idea of what a particular summing amp is doing, right? Let’s see..

Below are three download links for a section of a blues song (Right-Click and choose ”Save Target As…” to download). They are as follows:

1. A rough mix “bounced to disk” in Pro Tools HD

2. The same rough mix that has been outputted through the Digidesign 192 analog outputs and printed back into a stereo audio track (again through the analog inputs) in Pro Tools and

3. Again the same rough mix split out from the Digidesign 192 analog I/O and through the Tube Tech SSA2B Summing Amplifier and printed back into a stereo audio track (yet again through the analog inputs) in Pro Tools.

The split out was as follows: Drums (L&R 1/2), Bass Guitar (L&R, 3/4), Guitars (L&R, 5/6), Organ (L&R, 7/8), Horns (L&R, 9/10) and Vocal (L&R, 11/12).

Enjoy! I am personally very fond of the what the Tube Tech delivers and think you too will be presently surprised.

There always seems to be some confusion on the topic of word clock, what it is and why it’s relevant. Let’s see if we can clarify some of this. First, let’s take a stroll back to digital audio theory 101.

When recording in the digital domain, the A/D convertors of a digital audio workstation are taking snapshots of the incoming audio signal at a specified rate. These snapshots are called samples and hence the term sample rate. So, by way of example, if you have a session that’s recording at 48k, the incoming audio is being recorded (or acquired) at 48,000 samples per second. Makes sense so far, right? Now, in order for all these samples to play back at the same way in which they were recorded (which in this example is 48k) there has to be a common clock source as a point of reference.

Now, I know what some of you may be thinking: “If I don’t own an external clock, how does this process work on a system just using a computer and a firewire or usb interface?” Answer: All digital devices have their own internal word clock-everything from CD players to home computers using just iTunes. Otherwise the playback of the digital audio wouldn’t work properly. Now, this doesn’t always mean the fidelity of that clock is necessarily all that great. Remember: When you purchase a $400 audio interface that has microphone preamps and a bunch of other bells and whistles, you get what you pay for. Convenient? Absolutely. Audiophile? Far from it. But, what happens when there are multiple digital devices in the chain? Good question. Here’s an example we like to use:

Imagine a band with a drummer who can keep good time. The drummer sets the beat, everyone follows along, and they sound great. But suppose each band member decides to use their own metronome and listened to it rather than the drummer. Even if everyone set his or her metronome to the same tempo and tried to start it as soon as the drummer hits the first downbeat, reaction times would vary and they won’t all click at exactly the same time. Furthermore, the metronomes may not be calibrated the same way or be perfectly stable in the first place. So after a couple of minutes, the performances will all start to drift further and further apart. That would become one very sloppy band!

Like said band , all digital components need an accurate clock to keep the data stepping through at a constant rate. Otherwise this sloppiness called jitter occurs and the fidelity and accuracy of what you’re hearing is compromised.

Below are links to download some files that compare the differences between an internally clocked mix and the same mix clocked using the Drawmer M-Clock Lite. I encourage you to download the files (right click on the file name) and listen to the sonic differences. Drawmer makes the only line of digital clocks that follow the AES Grade 1 clock specification-the highest regarded specification known for superior clock performance. People ask me all the time what is a quick and easy way to improve the fidelity of their mixes. I always suggest using a master clock. It’s an easy addition and a sure fire way to make sure everyone in your digital audio workstation is playing on the same team.

Mix Using an Internal Clock

Mix Using the Drawmer M-Clock