AF’s Weblog

June 14, 2012

Top 10 Things That Can Never Be Taught Often Enough In Audio

Filed under: Live Sound — Tags: , , , , — audiofanzine @ 8:48 am

10. Musicians feel most comfortable and play best when they hear what they need to hear on the stage. Of course, the experienced monitor guys and recording guys already know this. But it’s something for those less experienced to think about. No, it’s not about how much power you have or what kind of monitor wedges. It’s about psychology.

And I think it’s true that if you become good at monitors and understand how to please musicians, you are 90 percent there towards becoming a good mix engineer.

Sure, the last 10 percent might be the “magic” but you can’t make magic without the basics.

9.  Sound travels at 1,130 feet per second, at sea level, at 68 degrees F and 4 percent relative humidity. This is important because if you understand how sound propagates, you’ll automatically know more about microphone placement, setting delay towers, and things like delaying the mains to the backline. And you should also know that the speed changes with temperature, humidity, and altitude. (If you don’t, it’s a good idea to look it up.)

8.  The Inverse Square Law. You know, the thing about a doubling of the distance from the source means that the acoustic power is cut by 1/4, right? This applies all over the place, from mic technique to loudspeaker arrays. It relates to how much power you will need from the power amplifiers.

For instance, if you normally cover an audience at 20 to 60 feet from your stacks, but for the next gig, the audience will be 40 to 100 feet away, how much more power will you need to maintain about the same acoustic power? About four times as much! Maybe think about delay stacks (see #9).

Let’s see some more pointers…

2. Grounding. Let’s not mince words here: this is a subject you need to understand. If you have more than one path to ground in your audio system, and the resistance to ground is different between them, you will have problems with hum and buzz.

Related to this is how you terminate your connections, especially if any parts of the system go back and forth from balanced to unbalanced terminations.

It’s also a good idea to learn the sonic signatures of different kinds of hum and buzz to therefore speed up your troubleshooting when the time comes. This is because some types of buzz are not related to grounding problems, but instead may be power supply issues, for instance.

1.  Gain structure, baby. This is the main one, the real deal. The thing that, if you can’t learn, or don’t understand or have forgotten, will get you into more trouble than anything else. There will be more noise and/or more distortion in the system unless you get this right. And there will be less gain before feedback, too.

So here’s the deal: every input and every device has an optimum range of levels it wants to see or wants to work with. If you’re feeding something a signal that is too low, you have to make this up somewhere, and therefore you’ll be bringing up the noise more than it should be. And that noise will be in your signal from then on.

Oh, sure, there are noise reduction devices you can use, but why do that when proper gain structure will take care of it for you? And really, we should use the least processing possible to get the job done because things sound better that way.

Alternatively, if you an input or a mix bus is fed too much signal, headroom will run out, which means you’re adding distortion. And this, also, cannot be removed later. Artistically adding distortion via plug-ins, hot-rodded guitar amps or certain outboard gear can be cool. Adding it by slamming your inputs or your mix bus is not cool.

For instance, if a wireless microphone output can be set at line level, but you set it to mic level and connect it to a mic input on your mixer, you will have more noise than if you connect the line output to the line input. Why? Because essentially you’re padding down the output then boosting it back up again with a high-gain mic preamp.

Sure, sometimes you might want to put the signal through a transformer or other “good” distortion device—just be aware that from a gain structure point of view, this is not ideal.

OK, that’s the list. If you’ve already mastered these things, great! You’re probably doing better mixes, with more gain before feedback, better coverage and happier musicians than those who don’t. But please don’t rest on your laurels – get out there and learn as much as you can.

Those of us going to your shows will know it when we hear it!

To read the full detailed article see:  Top 10 Things That Can Never be Taught Often Enough in Audio


September 30, 2009

Understanding Reverb

When we hear sounds in the “real world,” they are in an acoustic space. For example, suppose you are playing acoustic guitar in your living room. You hear not only the guitar’s sound, but because the guitar generates sound waves, they bounce off walls, the ceiling, and the floor. Some of these sound waves return to your ears, which due to their travel through the air, will be somewhat delayed compared to the direct sound of the guitar.

This resulting sound from all these reflections is extremely complex and called reverberation. As the sound waves bounce off objects, they lose energy and their level and tone changes. If a sound wave hits a pillow or curtain, it will be absorbed more than if it hits a hard surface. High frequencies tend to be absorbed more easily than lower frequencies, so the longer a sound wave travels around, the “duller” its sound. This is called damping. As another example, a concert hall filled with people will sound different than if the hall is empty, because the people (and their clothing) will absorb sound.

Reverberation is important because it gives a sense of space. For live recordings, there are often two or more mics set up to pick up the room sound, which can be mixed in with the instrument sounds. In recording studios, some have “live” rooms that allow lots of reflections, while others have “dead” rooms which have been acoustically treated to reduce reflections to a minimum – or “live/dead” rooms which may have sound absorbing materials at one end, and hard surfaces at the other. Drummers often prefer to record in large, live rooms so there are lots of natural reflections; vocalists frequently record in dead rooms, like vocal booths, then add artificial reverb during mixdown to create a sense of acoustic space.

Whether generated naturally or artificially, reverb has become an essential part of today’s recordings. This article covers artificial reverb – what it offers, and how it works. A companion article covers tips and tricks on how to make the best use of reverb.

Now let’s take a sneak peak into the nitty gritty of reverb…


Advanced Parameters II

High and low frequency attenuation. These parameters restrict the frequencies going into the reverb. If your reverb sounds metallic, try reducing the highs starting at 4 – 8kHz. Note that many of the great-sounding plate reverbs didn’t have much response above 5 kHz, so don’t worry if your reverb doesn’t provide a high frequency brilliance – it’s not crucial.

Reducing low frequencies going into reverb reduces muddiness; try attenuating from 100 – 200Hz on down.

Early reflections diffusion (sometimes just called diffusion). Increasing diffusion pushes the early reflections closer together, which thickens the sound. Reducing diffusion produces a sound that tends more toward individual echoes than a wash of sound. For vocals or sustained keyboard sounds (organ, synth), reduced diffusion can give a beautiful reverberant effect that doesn’t overpower the source sound. On the other hand, percussive instruments like drums work better with more diffusion, so there’s a smooth, even decay instead of what can sound like marbles bouncing on a steel plate (at least with inexpensive reverbs). You’ll hear the difference in the following two audio examples.

Maximum DiffusionNo Diffusion

The reverb tail itself may have a separate diffusion control (the same general guidelines apply about setting this), or both diffusion parameters may be combined into a single control.

Early reflections predelay. It takes a few milliseconds before sounds hit the room surfaces and start to produce reflections. This parameter, usually variable from 0 to around 100ms, simulates this effect. Increase the parameter’s duration to give the feeling of a bigger space; for example, if you’ve dialed in a large room size, you’ll probably want to add a reasonable amount of pre-delay as well.

Reverb density. Lower densities give more space between the reverb’s first reflection and subsequent reflections. Higher densities place these closer together. Generally, I prefer higher densities on percussive content, and lower densities for vocals and sustained sounds.

Early reflections level. This sets the early reflections level compared to the overall reverb decay; balance them so that the early reflections are neither obvious, discrete echoes, nor masked by the decay. Lowering the early reflections level also places the listener further back in the hall, and more toward the middle.

High frequency decay and low frequency decay. Some reverbs have separate decay times for high and low frequencies. These frequencies may be fixed, or there may be an additional crossover parameter that sets the dividing line between low and high frequencies.

These controls have a huge effect on the overall reverb character. Increasing the low frequency decay creates a bigger, more “massive” sound. Increasing high frequency decay gives a more “ethereal” type of effect. With few exceptions this is not the way sound works in nature, but it can sound very good on vocals as it adds more reverb to sibilants and fricatives, while minimizing reverb on plosives and lower vocal ranges. This avoids a “muddy” reverberation effect that doesn’t compete with the vocals.


Now that we know how reverb works, we can think about how to apply it to our music – but that requires its own article! So, see the article “Applying Reverb” for more information.

To read the full detailed article see:  Understanding Reverb

June 12, 2009

Antelope Audio – Trinity Clock

Antelope Audio presents their new Trinity, a multi channel, all format audio & video master clock that uses 64 bit DSP.

To see more exclusive video demos visit Audiofanzine Videos.

April 13, 2009

Audio Encoding: What Lies Ahead?

Introduction to Audio Encoding

In today’s high definition world, we want the best quality in every film, picture, and sound we encounter. Blu-Ray DVD’s seem poised to take over the market, pushing the inferior DVD’s down the same path as the VHS tape. In much the same way, we see digital audio pushing for the same quality. Although the popularity of the iPod and mp3s has given rise to an age of over-compressed, low-quality music, we also see a rise in vinyl sales, as well as developments such as Sony’s Super Audio CD (SACD). This implements a relatively new process to encode audio, paving the way for a massive change in the quality of music we listen to, if it is accepted.

Analog vs. Digital

In order to understand audio encoding, the difference between analog and digital must be understood. Something that is analog is an uninterrupted, pure, natural sound. The human voice, a guitar, and a vinyl record are all examples of analog sound. When a vinyl record is cut, a needle senses the vibrations from an audio source and cuts it exactly into the vinyl. This is why vinyl could be said to have the highest sound quality, and is still popular today, despite many alternatives and developments. An analog signal can encompass all frequencies, even the inaudible. This is why a live orchestra sounds more “full” than a recording, even of the highest quality. Audiophiles will argue that the energy of the inaudible frequencies adds to the quality of sound, even though they cannot be perceived by the ear.


A digital signal is a replication of an audio signal by a number of ones and zeros. It is the same way a picture on a computer screen is replicated by thousands of intensity values represented in binary code. The music on an iPod, a compact disc, and an mp3 are examples of digital replication. Even many modern musical instruments have implemented digital sound, from digital keyboards to electronic drums to guitar pedals. Digital sounds can be made with a programmable chip, rather than a circuit, and is much more reliable, inexpensive, and easy to mass produce as a result. However, there are drawbacks to digital sound that sacrifice the sound quality, and these drawbacks are being constantly developed and upgraded to replicate an analog signal more precisely.

Now let’s take a closer look…



Thus, 1-Bit modulation has been implemented in many new “high definition” audio devices, and developers continue to use this process to expand into multi-bit modulators and other hybrid converters. It has become accepted among audiophiles, and is slowly taking the place of PCM. Is one better than the other? It is still debatable. However, 1-Bit modulation allows for simpler circuitry and much better noise shaping in lower frequency bands. The SNR is much better than PCM, except in the higher frequency range, where much of the noise is inaudible anyway. The design is simpler, using more digital implementation than PCM, and as programming advances, digital functions like noise shaping will be enhanced.

Once 1-Bit modulation starts to become affordable, consumers will begin to realize the poor quality of mp3’s. Steps will be made to expand 1-Bit audio into the portable market, and “high definition” audio will become the norm. Until then, only the select few who have heard the differences will know how much better sound quality can be, and will only strive to educate the rest.

To read the full detailed article see Audio Encoding

January 5, 2009

Test: Zoom ZFX Stack Package Review

Tube or not tube
Zoom Stack Package: The Test

Though the concept may not be original, Zoom has come out with an interesting interface that provides guitarists and bassists with an intuitive software/hardware setup, but with a twist: an integrated vacuum tube. In recent years there have been quite a few interface choices for guitarists and bassists, especially from line 6, but Zoom’s take on this combines amp modeling software with a USB audio interface featuring a Hi-Z input with a vacuum tube. Let’s take a closer look…

Vue générale

The S2t interface comes in the form of a vintage looking mini amp-head (222mm x145mm x82.5mm). The whole thing seems well made and robust (1.1kg). The knobs neither feel nor look cheap, and have just the right tightness for precision tweaking, though they are a little crowded, making it difficult to not touch an adjacent knob while turning one of them. All the input and output jacks also look and feel solid. Basically it’s a device that seems built to last, especially when compared to its competitors.


The ZFX package includes the S2t interface and the ZFX plugin. Installation on Windows XP proved to be fairly easy and hassle free. This interface apparently also installs easily on Mac OS X and Linux systems without any drivers needing to be installed*. Be aware, though, that the included ZFX software will not work on those systems, only Windows. This is why Zoom included Guitar Rig 3 LE in the package, and not because they thought their software wasn’t good enough, as some might assume. Zoom also kindly included Steinberg’s Cubase LE4 for those in need of a DAW.

Vue générale

I wanted to see if the interface and software would work on a less powerful PC so I installed it on my old laptop (Windows XP SP2, 512 MB Ram). Apart from a few graphic issues (my card is no doubt a little too old) the interface and software worked perfectly once I correctly configured my audio settings in my DAW (Sonar). I was able to get very low latency (1ms) which surprised me from this USB interface.

There are two pdfs (startup & manual) as well as two very light printed startup guides included. The problem with these guides is that they focus mainly on installation and using the ZFX software (150 pages!). There’s next to nothing on the physical interface (the S2t) itself, nor on advanced settings. I had an issue with direct monitoring latency which was too long (around 15-20ms), for which I couldn’t find any information. I ended up testing all basic and advanced settings, finally finding the solution when I changed the buffers in the advanced section. It seemed pretty obvious after the fact, but there was no mention of buffers and their effect in any of the guides I read.

Let’s take a closer look…


Again, what’s interesting in this device is it’s versatility or modular aspect. You can, if you want, experiment with other tubes inside the S2t, and the ZFX software also offers a lot of flexibility in terms of combinations and editing. And though the ZFX plugin/standalone couldn’t be considered sonically superior to competitors like Guitar Rig or Amplitude, it isn’t quite inferior to them either, and offers some unique features. Of course the best thing would be for you to try it out yourself, especially if you want to test it against Guitar Rig. You might just find that you like it better. But even if you find the ZFX software not to your liking, the S2t interface still has much to offer. At around $190 (about the same price as the Line 6 UX2) you get a robust and relatively stable guitar/bass oriented interface with low latency and pretty good sound quality. The Hi-Z feature is an interesting plus, though nothing revolutionary. The integrated tube will not convince everyone, but the fact that it’s USB powered should make it more appealing to Home Studio enthusiasts on the go.

Solidity: the interface, its knobs, and connecters
Nice look
Hi-Z Concept – integrated vacuum tube
Flexibilty and modular aspect of both interface and ZFX software
USB powered

ZFX GUI is a resource Hog
Manuals hardly deal with the S2t
ZFX plugin only works for Windows-based PCs

Read the full Zoom Stack Package review article.

December 15, 2008

Test: BIAS Peak Pro XT6 Review

Though there are many audio editors for PC, the Mac world looks mainly to DSP Quattro, soundBlade, Audacity, WaveBurner or Peak (a non-exhaustive list), whose features range from “very basic” to “very sophisticated”. No wonder then that this software update of Peak Pro XT 6 (the extended version) by BIAS has been eagerly awaited.

Of the three versions, LE, Pro, and Pro XT 6, it’s the latter that was installed and tested. With Pro XT 6, BIAS has consolidated its editor, SoundSoap 2 and SoundSoap Pro cleaning software, version 1.2 of its Master Perfection Suite (which explains the price difference between the Pro version at $599 MSRP and the Pro XT at $1199 MSRP), and the Peak Production Pack. This pack includes a library of sounds (a little over 1 GB) by Sound Ideas, Shortwave, Serafine Sound FX, PowerFX, and Hollywood Edge, plus a $100 Broadjam coupon, a limited version of ONE sample player, the SFX Machine LT multi-effects (light version), 32 VST MDA plugins, Reveal LE and SoundSoap LE (identical to the full versions, but that only work within Peak), JackOSX (0.74), Soundflower (1.2.1) and WireTapPro (1.3.4). Take a look at the comparisons between the three versions here.

The software has to be authorized via the internet by entering the serial numbers into the BIAS Authorization Manager. Once that is done, it will be immediately validated. You can authorize two computers, and the BIAS Key is no longer necessary (you regain a USB port …). At startup, there’s a surprise: the GUI has been redesigned. From the dominance of gray in version 5 we go to a dark gray-black look (by default, which be can changed), and buttons now feature a simulated backlighting which highlights an icon when the mouse pointer is rolled over it. The windows are “magnetized” with a default position that can be recalled.

Let’s take a closer look…


ddp 2

BIAS delivers an editor that’s very complete, plus a sound and software suite that’s just as comprehensive, which explains its price. It’s true that we’ve become used to getting more for our money with the likes of Apple’s Logic Suite for example, but we must not forget that this is far from the norm … Plus you have to take into account the loops and effects (excellent quality), the various included plugins which perfectly fulfill their role (starting with native Peak plugins), EQ with different Freq versions, Sqweez compressor, multiband compressor Sqweez-5, and Reveal, all of which are very well designed. There’s not enough space here to talk about them, but they deserve their own test.

With perfect stability, Peak provides worry-free processing of batch files. There are practically no bugs, but note, however, the sensitivity of the Preview feature, which must be handled with care (better not rush things). In fact, I have only encountered a single problem with a batch processing and saving files as .WAV from a batch of files in AIFF in which some files contained accents and special characters (which didn’t pose a problem with Peak 5). Contact with BIAS failed to reproduce the problem on their side …

Another point that can be a problem but which, BIAS says, would be corrected in a future update: when you open files, that weren’t created by Peak, for the first time, the software changes the modified date, even if there is no editing. This may cause problems in managing backups with Time Machine, Super Duper or other CCC …

In short, Peak is still unavoidable, even essential to any Mac users treating many music files in any format, or wishing to prepare playlists regularly, regardless of their destinations. If the XT version seems too expensive, the Pro version offers a comprehensive editor, without the Master Perfection Suite and SoundSoap Pro.

Plein de fenêtres

Comprehensive audio editor
Many DSP functions
Ideal for sound design
Many audio formats
Cache in RAM
Native editing in most compressed formats
Envelopes (volume, plugins, etc..)
Playlist management
Import/export to SMIDI hardware samplers
Quality of sample rate conversion
Quality of dithering
Vbox 3
VST and AU support
Virtual instrument support
Convolution reverb included
Comprehensive Pdf manual
Almost bug-free
Comprehensive XT Suite (Master Perfection Suite, SoundSoap Pro)
Doesn’t use a USB dongle

Changes the modified date of files as soon as their opened
Sample rate conversion very slow on Mac PPC
Pitch shift and time stretch algorithms (despite the progress made …)
Ram cache for 16-bit files only
Sensitive preview feature
Problems with batch conversion containing certain characters

Read the full review of BIAS Peak Pro XT6 Review here.

December 9, 2008

Test: M-Audio ProFire 2626 review

With no fewer than 26 inputs, 26 outputs, 8 integrated Octane microphone preamps, and ProTools M-Powered compatibility, the latest interface from M-Audio aims to find its niche in the category of intermediate-level FireWire audio interfaces. Should the competition be worried?

trois quart

M-Audio has been releasing quality products at very attractive prices (even aggressive prices) for some time now: microphones, MIDI controllers, sound cards and other Home Studio accessories. Bought in 2004 by Avid, which also owns Digidesign, M-Audio now offers sound cards that are Protools M-Powered compatible.

This Protools version allows Home studio owners to use this “legendary” software and create sessions that are compatible with TDM versions, something which had already been possible with Digidesign cards (like MBox, DIGI002, DIGI003) but which has now become more affordable thanks to M-Audio.

The product we’ll be reviewing is the M-Audio Profire 2626 digital audio interface, which has the following specs:

  • 26 x 26 simultaneous analog/digital I/O
  • Up to 24-bit/192kHz
  • 8 mic/line preamps using Octane technology including 2 instrument inputs on the front panel
  • Two ¼” TRS headphone outputs, and a user-assignable master volume knob
  • An onboard DSP mixer that allows routing of internal signals without taking up processor resources
  • Standalone operation (functions as eight-channel mic pre/eight-channel A/D-D/A converter)
  • JetPLL technology – jitter elimination (unwanted variation of one or more characteristics of a periodic signal)
  • Wordclock I/O

Basically, it’s got a lot of nice features which, for an average price of USD 899.95 MSRP (around $699 average street price), could be a very good alternative for people who want to mix with an analog or digital console. There are enough inputs & outputs to put down your tracks without a problem.

What’s in the box?…


M-Audio has come up with a very good product at an interesting price with their Profire 2626. Except for some minor installation issues and control panel display bugs, its internal routing, quality preamps and converters, and numerous inputs/outputs (almost boggles the mind considering the price) make this interface a must-have for people who want to record numerous tracks or who want to try venturing outside of their favorite sequencer to use a console (and thereby use external effects). The icing on the cake is that it works as a standalone A/D-D/A! So with an average street price of $699 it definitely deserves a value for the money award.

Number of Inputs/Outputs
Quality preamps and converters
Internal Routing
2 headphone outputs
Has Wordclock
Compatible with Protools M-Powered

Knob push/pull quality
Phantom common to inputs 1 to 4 and 5 to 8
Installation slightly arduous
Display bugs in M-Audio control panel

Read the full M-Audio ProFire 2626 review here.

August 26, 2008

Sound Techniques : basics of acoustics

Filed under: Instructional articles — Tags: , , , , — audiofanzine @ 3:49 pm

With stopwatch in hand, our perception of time seems straightforward. But in everyday life we’re not always watching the clock, and everyone knows that the passage of time is relative. It differs from one person to another and especially from one activity to another: An hour spent watching a great movie doesn’t feel as long as an hour in traffic.

Read the article about basics of acoustics on Audiofanzine.

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