The following is a Facebook exchange that I had with a former student outfitting his new studio.  He raises some great questions about what makes a cable compatible with audio.

JB:
if i were to put an audio snake through 1 1/4 conduit and i were to use cat5 as a temporary cheap(free) way to do this would it work for 16 channels?

The Faderport by Presonus

This week I purchased the Presonus Faderport for use with my DAW, Sonar Producer Edition 8.31.  I have been finding that I really like the immediate control and hand/ear coordination that a real fader provides because I have been working a lot on the API Vision at U. Mass Lowell.  I read a zillion comparisons between the Faderport and Frontier Design’s Alphatrack, a similar single fader automation encoding device. Here’s a very interesting video that compared the speed of the faders on both units.

As you can see from the video, the Faderport’s reaction time is significantly faster than the Alphatrack.  It should be noted that fades as fast as the ones in the video are pretty rare.  For something this fast, most of us would do a mute automation and not a super fast fader move.  The fader on the Faderport is very nice feeling.  It’s quite smooth and although it is a little noisy, it is clearly the highlight of the unit.

The pan knob frankly sucks.  It is a detented pot, so it clicks as you turn it.  There is no specific center detent and in Sonar after you move the pan control the closest you can get to center is +/- 1%.  You also need to rotate the knob completely several times before you get a hard pan left or right.  The pan control is actually more cumbersome than doing fade automation with a mouse.

The instructions for install are really poor.  For individual DAWs you can’t use the included CD-ROM for the installation.  Rather you need to go to the PreSonus website and download a specific driver/plugin for your DAW, but they don’t tell you this in the instructions.  The Sonar driver doesn’t include the feature of being able to program the single user-assigned button with a task and the PROJ button (which is supposed to open the track/edit view in Sonar) doesn’t work at all.  You can open the Mix window and the Transport control, but not the Edit window which is the most commonly used window in Sonar.

It also isn’t clear if you can assign the fader or the pan control to anything other than volume or pan, so you are REALLY limited as to what you can control and automate from the device.  All in all I found the Faderport pretty disappointing and I plan on returning it and getting an Alphatrack.

Here’s the video overview of the PreSonus:

Daking Audio Gear: Mic Pre IV and 3 FET II Compressors

I currently have three of these units in my studio right now and I have had a chance to really put them through their paces.

First, I should say that the sound quality on these units is pristine. There is very little coloration of the sound even when using heavy compression. Many compressors seem to roll off high end when they attenuate heavily, but this is not the case with the FET II. The FET II uses Jensen transformers both in and out of the unit and the pc board is extremely clean and well designed. The FET is in a socket so if it were ever to go bad, it is easy to replace.

The FET II excels at transparent compression and is easily used on bus or program material where lesser compressors really start to sound yucky. The attack times vary between 250 micro seconds to 64 milliseconds and it’s fast enough to be used effectively as a brickwall limiter if desired. The release characteristics are I think what really set the compressor apart though. You have some standard settings of .5 – 1.5 seconds, but also some really nice dual time constant releases designed to mimic some of the nicest compressors in history. The idea behind dual time constant release is this: the compressor releases a little fast at the beginning and then slows down. This effectively eliminates the “pumping and breathing” sounds associated with more abrupt release times.

I have also been able to get some really nice vocal distortion (think Flood’s production techniques) out of it by using the fastest attack and release times and a very high ratio (20:1). Then I drive a very hot signal (over +20) and get a very pretty sounding harmonic distortion very appropriate for alternative rock vocals like NIN, PJ Harvey or Smashing Pumpkins.

I recommend using only XLR cables in and out of the unit, you can use a 1/4″ input but it boosts the signal 14 dB to make up for the -10/+4 difference in operating levels between consumer and pro gear. Another odd thing is the power supply (external, but not a wall wart) uses a DB25 connector which looks pretty weird, but works perfectly well. Just make sure your intern doesn’t try to run the power supply into the DB25 input on an audio interface or multitrack….Bad intern! Bad intern!

You can link two units together to work in stereo with a 1/4″ guitar cable. The sidechaining connection uses DC summing to tell the linked unit when to compress and does not send audio. The FET III does audio summing, but it’s in stereo and is geared more towards working in stereo anyway.

All of the knobs on the unit are switches so you can set two or more compressors exactly the same way and repeat your settings later on. The knobs are really heavy and feel like you’re really working with pro gear.

All in all this is a great compressor with excellent transparent compression that doesn’t color the sounds you are working with. You can use it to chase the waveform to create harmonic distortion with the fastest attack settings to add a little crunch to vocals, bass or drums.

I can’t recommend it more highly.

The following video is about my recording philosophy and the gear that I use at Indecent Music.  I record, mix, and master out of Indecent Music.  I also provide audio engineering training and private lessons so that song writers can learn to be more effective at making their own demos.

Their is difference between someone who makes beats – meaning composing and performing (or programming) original instrumental music, someone who is really a producer, and a recording engineer that specializes in hip hop tracking and production.

The fastest way to learn to beat making is to make beats with whatever you have available. I have worked with a couple of heads who were complete geniuses with the Playstation software from MTV. Their music was simply amazing. Software that is highly under-rated is FL Studio or FruityLoops. The step sequencer is the easiest way to make music quickly. Read the manual! Watch videos online.  Start working with as many other beat makers that you can find on the net, in your home town. For me, competition made me write stuff that was much better than working by myself in a vacuum. The three big instruments to learn would be keys, drums, and bass. You did not need to work in a studio to do this kind of work. You need a computer, a decent audio interface (Not an M-Box), and a couple of nice monitors. If money is a factor, don’t get a Mac. You get a lot more computer in the PC world and there’s tons of software available.

A real producer puts the whole show together. They hire everyone, often write songs with the artists, choose the studio to work in, find live musicians to fill out the sound. Sometimes that means doing everything yourself. A lot of the time the producer FUNDS the project and gets the biggest share of the profit (if any).  A producer is a big picture person usually with an excellent understanding of the psychology of creative people, motivation, fear, competition and excellence. This is something that comes with lots of experience, a strong musical background, charisma and usually fame or money.

An engineer deals with the tiniest details of tracking and mixing. Moving a mic a half inch, rotating a mic off axis, how to attenuate the peaks of the kick to get it to sound bigger, without making it wimpy. Attack and Release time minutia for compressing drums, bass and vocals. How the sound stage can be used to the best advantage, how to either avoid masking or use it to create new timbres. You need to learn this either in a studio as an apprentice, in a good audio school that has great facilities (I teach at New England Institute of Art in Boston and at U. Mass Lowell both have great facilities) and then leverage that into getting good internships.

Sometimes there are people who really are all three. Sometimes you will find yourself in one role or the other depending on who you’re working with.

The best job to get to learn audio engineering is working for live sound companies as a grunt. You will carry the bass bins, mic stands and a 43 foot console. But you will get to watch the FOH and monitor guys throw down. Live is good because it forces you to learn to do things quickly and it puts you around dozens of musicians every weekend. Not wanting to be embarassed is a very powerful way to learn.  You are always on stage being watched from the time you load in, to the time you strike the stage.

(posted to GearSlutz 7-4-09)

One of the hardest audio processors to understand is the compressor.  Even after several years of using compressors many of my students and readers still have lots of questions about how to dial in the sound that they are trying to get.  Compressors are in the Dynamics Processors family which also includes limiters, expanders, gates and noise reduction.  Dynamics processors work in the Amplitude Domain.  Compressors work on the amplitude of an audio signal, which is basically the loudness of the signal.  Look at a waveform view of an audio signal:

Graph of a Sine Wave with Amplitude and Frequency

The vertical axis shows Amplitude, which in analog (electrical)  audio refers to the amount of voltage in an analog signal. When the wave is above the center line, then the voltage is positive and when the wave is below the line the voltage is negative. Audio (in the electrical analog sense)  is AC or Alternating Current which means the voltage goes from positive to negative and then back again. The further away from the center line, the higher the voltage and the louder the wave will sound.

The Dynamics of music is generally thought to be the differences between the loud parts of music and the quiet parts of the music.  The dynamics of audio includes all of the differences in amplitude along the waveform.  In most pop music, for instance, the loudest parts of the music are the snare drum hits, followed by the lead vocal, then the background music. Notice in the following image the red dots above the waveform.  They are marking the locations of the snare and kick drum hits in the music.

The red dots mark the locations of the snare and kick drum hits.

Notice that there is audio in between the loud hits as well, but that it just has a lower amplitude. Compressors and all dynamics-based effects work on the amplitude of the audio, to adjust and change the differences in voltage.  The loudest level in digital audio is 0 dB Full Scale or (0 dB FS) which means that anything above that level will be distorted or simply just an error.  We can’t change the loudest possible level, but we can change everything that is below that level.

What a compressor does:

A compressor attenuates (decreases amplitude) audio that is above a threshold by a ratio.  The attack time is how quickly the compressor starts to attenuate the signal after the threshold is exceeded and the release time is how quickly the compressor stops attenuating the signal when the audio drops below the threshold level.

Probably the most common use of a compressor is to make an audio signal sound louder without peaking out the signal and causing clipping and distortion.  In a nutshell, the loudest parts of the audio signal (the peaks) are made a little bit quieter so that all of the signal can be boosted by the amount that the peaks were attenuated.

I am a PC guy.  I think that it is the best platform that gives the most options for hardware and software.  I think Mac’s a great, but they’re way more expensive and they have always felt like toys to me.  They are the Nerf brand of computer.  Unfortunately most pro studios have Mac’s and I find that I need to use my external disks both in other people’s studios and in my own.  The only decent format that works in both is FAT32, but on both platforms FAT32 is NOT the file system of choice.

Windows machines really prefer NTFS, the NT File System which has many fewer limitations.  Mac has their own file system as well.  An important problem with the FAT32 file system is that the maximum size of a file is 4 GB. Windows won’t let you format a hard drive with FAT32 if the drive is a big modern drive.  In fact Windows XP will not format a drive bigger than 32 GB with FAT32.  This is a good example of a Windows-Style Suck-a-doodle-doo.  You need to format to Fat32 with a Mac, Linux or use a third-party tool to format on a Windows machine.  Windows will read and write to a larger FAT32 drive, but won’t allow you to create one.

For Windows, the easiest tool to download is Acronis’ True Image Home.  (http://www.acronis.com/homecomputing/download/trueimage/) They offer a free 15 day trial that will allow you to format large disks as FAT32.  Just go through the process for “Adding a drive…”

Acronis is a great back up software tool as well.  It allows you to create images of your system disk and incremental or differential backups as well.  I find that it better than Norton’s Ghost, but have found that it doesn’t handle hard disk failure on the destination drive very well.  Their support offerings are pretty good, but not fast.

As many of you already know, I am now teaching at two colleges: University of Massachusetts Lowell and the New England Institute of Art.  At both schools I teach in the Audio Production departments, but at UML, it is called SRT or Sound Recording Technology. I can recommend both of the text books.  They have different perspectives and both are well established texts in the field.

The first book that is required reading for UML’s  class 78.305 “Survey of Music Technology” is Experiencing Music Technology by David Williams and Peter Webster.  The book is quite expensive in stores, but is a little cheaper at Amazon as usual. A new edition of the book has just become available to update the content with internet technologies, contro surfaces and other innovations from the last 10 years.

Experiencing Music Technology

The second book is also expensive unfortunately. Audio in Mediais in its 8th edition and is one of the most updated books on the subject. This text covers everything from acoustics to post-production. It’s fantastic overview of music technology from mics and loudspeakers to control surfaces and signal processors.

Audio in Media

So first things first: credit where credit is due.  I learned how to do these modifications from Billm’s Blues Junior Modification pages at http://home.comcast.net/~machrone/bluesjunior.htm

His parts (which I bought from him) and his instructions were excellent and really worth reading.  It was a cool little project and took a couple of hours to do. I heartily recommend getting the parts and directions from him because they were really helpful.

The following is a photo of the back of the Blues Junior on my bench with the screws removed from the back plate.  I didn’t remove the screws on the sides on on top of the amp.  Those are for the chassis inside the speaker enclosure.

I like to use a muffin tin to keep all of the parts that I take off of a project. It’s REALLY easy to get small parts out of the muffin holes even if you don’t have any finger nails, like me.  Some other people I know like egg cartons, but I had a bad experience with a soldering iron and Styrofoam that doesn’t need to be repeated.

The following photo is what the PCB looks like before you start voiding your warranty!

Now I have unscrewed the PCB from the chassis and I have wiggled the shafts of the pots out of their holes.  I also removed the daughter card for the foot pedal and the speaker out.

The blue resistor in the very center of the picture replaces the stock resistor and helps to cool down the amp bias.  It’s below the three horizontal caps and next to the 3 vertical gray caps. This helps to remove the mushiness (lack of definition and transient response), especially in the lower frequencies.

You can see the two leads from the resistor poking through the back of the PCB. Incidentally the PCB is incredibly cheap and uses very little copper. You have to be VERY careful to not ruin the copper pads by using too much heat. The pads will peel up and roll themselves into a tiny ball.

If you look in the upper left corner in the following picture you can see the big blue cap that I piggy-backed onto the gray caps that was the stock cap.  By doubling the capacitance of the power filter cap, you create a much bigger store of power which helps the amp deal with very fast transients and low frequencies.  By store the power in the cap, the amp can respond much better to the attack of picking and popping.

So that’s it for the photos.  The sound was really improved.  The amp responded much more quickly to changes in dynamics and the Low B note of my baritone guitar came through easily.  Before the modification, the amp struggled to produce the low E from a regular guitar with much authority.  I also replaced the speaker with an Eminence Swamp Thang speaker which is supposed to have extended low frequency response.  Speaker specifications really suck in general and usually don’t represent what timbre humans perceive.  This speaker has roughly the same specs as the speaker I replaced, but it sounds completely different.

All in all, I am really pleased with the changes!