AF’s Weblog

February 28, 2011

Fender Blacktop Series Review

Instead of launching the umpteenth reissue of a catalog instrument, Fender decided to innovate by mounting high-output passive humbuckers on a new series called Black Top. This new product range includes a Telecaster, a Jaguar and a Stratocaster equipped with the same pickup combination based on two humbuckers. The Jazzmaster gets a more original pickup combination with one humbucker (Hot Vintage Alnico Bridge Humbucking Pickup) and one P-90 in the neck position.

The Sonic Boom!

Originally, the humbucker pickup was invented by Gibson’s engineers to suppress unwanted noise by electrically and magnetically linking two single-coil pickups in series and out of phase. From the “practical” standpoint, guitar players know the properties of such pickups: a powerful, round and warm sound. As a consequence, humbucker pickups are the best solution for distortion sounds. Fender has a strong personality due to its single-coil pickups that provide a crystal-clear sound (they can be heard on many legendary rock albums). However, they could never really take the lead in the humbucker market — controlled by Gibson since the 1950’s.

Design

 

Fender Blacktop Series

The series is entirely produced in Fender’s factory in Mexico. All bodies are made out of alder with bolt-on maple necks with 9.5″ fingerboards and 22 medium-jumbo frets (except for the Jazzmaster). By standardizing the design and finish Fender can actually lower the price to a MSRP of $450! Most models are available with two different fretboards: maple or rosewood. The latter gives a warmer, rounder and more precise tone. Considering the price, we guess that the bodies are not made out of premium-quality wood but rather out of two or three glued pieces of wood. Just being realistic: with such prices, you cannot expect to get the same resonance as from a massive-wood, one-piece body. All guitars have a perfect skin: a polyurethane varnish with a faultless glossy finish. The neck finish is the same as on the Classic Reissue Series. It is very thick and protects the wood perfectly, providing excellent grip and optimal playing comfort while allowing to quickly access every point of the neck. The truss rod adjustment is accessible on the top of the neck, which is a modern and very convenient feature. The nickel/chrome hardware and the tone and volume knobs on all models recall the look of Fender amps. We must admit that this is a very original idea but it won’t be everybody’s taste.

 

You’re In the Army Now!

Let’s take a look at all new recruits of the Black Top Series.

 

Fender Blacktop Series

The look and sound of the Stratocaster is pretty well accomplished. The combination of the Candy Apple Red finish and the three-ply Mind Green pickguard looks wonderful and make the guitar a real eye-catcher. Two other finishes are available: Sonic Blue or Black with rosewood or maple fingerboard. The guitar has two Hot Vintage Alnico Humbucking pickups with chrome covers, a volume control, a tone control, a vintage-style tremolo, and a five-way toggle switch.

Position 1: full bridge pickup The sound is powerful and rich. It’s perfect for aggressive but precise rhythm parts.

Position 2: inside coils of the the two humbuckers. The response is hollow in the mid frequencies, the sound is lusty but not too wide.

Position 3: bridge and neck pickups in series. The low-frequency band is softened so that the mid range seems to be boosted, resulting in a flat and massive character.

Position 4: outer neck pickup. The most interesting sound among the five available. You get that unique Stratocaster sound without the sharpness.

Position 5: full neck pickup This setting produces too many lows, which results in a very heavy timbre. The tone is too heavy for rhythm parts but interesting for lead guitar.

Now let’s have a closer listen…

Conclusion

 

With the Black Top Series, Fender offers a very wide range of sound variations and finishes. The four different guitars use surprising pickup combinations! And their playability is almost perfect! The jewel of the family is the Jazzmaster, which is a really nice guitar aesthetically speaking but also provides a spicy expressive sound. The price in stores ($450) is very appealing and will surely attract guitar players who want a Fender without going broke.

Advantages:

  • Finish quality
  • Originality
  • Unbeatable value for money!
  • The Jazzmaster is especially appealing
  • Versatility of the Stratocaster

Drawbacks:

  • No gig bag
  • The Telecaster is a bit disappointing
  • Jaguar without a tremolo bridge

To read the full detailed article of the series with all sound samples see:  Fender Blacktop Series Review

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February 25, 2011

Much Ado About Dithering

Filed under: Mastering — Tags: , , , , — audiofanzine @ 10:31 am

It’s a dirty job to go from high-res audio to 44/16, but someone’s got to do it.

The ultimate form of digital audio used to have a 16-bit word length and 44.1 kHz sampling rate. Early systems even did their internal processing at 16/44.1, which was a problem — every time you did an operation (such as change levels, or apply EQ), the result was always rounded off to 16 bits. If you did enough operations, these roundoff errors would accumulate, creating a sort of “fuzziness” in the sound.

The next step forward was increasing the internal resolution of digital audio systems. If a mathematical operation created an “overflow” result that required more than 16 bits, no problem: 24, 32, 64, and even 128-bit internal processing became commonplace. As long as the audio stayed within the system, running out of resolution wasn’t a problem.

 

Nowadays, your hard disk recorder most likely records and plays back at 24, 32, or 64 bits, and the rest of your gear (digital mixer, digital synth, etc.) probably has fairly high internal resolution as well. But currently, although there are some high-resolution audio formats, your mix usually ends up in the world’s most popular delivery medium: a 16-bit, 44.1kHz CD.

What happens to those “extra” bits? Before the advent of dithering, they were simply discarded (just imagine how those poor bits felt, especially after being called the “least significant bits” all their lives). This meant that, for example, decay tails below the 16-bit limit just stopped abruptly. Maybe you’ve heard a “buzzing” sort of sound at the end of a fade out or reverb tail; that’s the sound of extra bits being ruthlessly “downsized.”

Dithering to the Rescue

Dithering is a concept that, in its most basic form, adds noise to the very lower-level signals, thus using the data in those least significant bits to influence the sound of the more significant bits. It’s almost as if, even though the least significant bits are gone, their spirit lives on in the sound of the recording.

Cutting off bits is called truncation, and some proponents of dithering believe that dithering somehow sidesteps the truncation process. But that’s a misconception. Dithered or not, when a 24-bit signal ends up on a 16-bit CD, eight bits are truncated and never heard from again. Nonetheless, there’s a difference between flat-out truncation and truncation with dithering.

Now let’s take a closer look at dithering…

Dithering Rules

The First Law of dithering is don’t dither a signal more than once. Dithering should happen only when converting a high bit-rate source format to its final, 16-bit, mixed-for-CD format (and in the years to come, we’ll probably be dithering our 32 or 64-bit internal processing systems down to 24 bits for whatever high-resolution format finally takes off).

For example, if you are given an already dithered 16-bit file to edit on a high-resolution waveform editor, that 16-bit file already contains dithered data, and the higher-resolution editor should preserve it. When it’s time to mix the edited version back down to 16 bits, simply transfer over the existing file without dithering.

Another possible problem occurs if you give a mastering or duplication facility two dithered 16-bit files that are meant to be crossfaded. Crossfading the dithered sections could lead to artifacts; you’re better off crossfading the two, then dithering the combination.

Also, check any programs you use to see if dithering is enabled by default, or enabled accidentally and saved as a preference. In general, you want to leave dithering off, and enable it only as needed.

Or consider Cubase SX, which has an Apogee-designed UV22 plug-in. Suppose you add this to the final output, then suppose ou add another plug-in, like the Waves L1-Ultramaximizer+. This also includes dithering, which defaults to being enabled when inserted. So, check carefully to make sure you’re not “doubling up” on dithering, and disable dithering in one or the other.

Dithering dans Cubase

If you insert dithering in Cubase SX, it defaults to being enabled. So if you use this, make sure that any other master effects plug-ins you add do not have dithering enabled (in this screen shot, the WAVES dithering has been turned off). Or, disable Cubase’s dithering section and use the other plug-in’s dithering instead.

The best way to experience the benefits of dithering is to crank up some really low-level audio and compare different dithering and noise-shaping algorithms. If your music has any natural dynamics in it, proper dithering can indeed give a sweeter, smoother sound free of digital quantization distortion when you downsize to 16 bits.

To read the full detailed article see:  All About Dithering

February 24, 2011

How to Create Wide Open Mixes

Filed under: Mastering, Mixing reviews — Tags: , , , , , — audiofanzine @ 11:15 am

Here are some secrets behind getting those wide, spacious, pro-sounding mixes that translate well over any system.

We know them when we hear them: wide, spacious mixes that sound larger than life and higher than fi. A great mix translates well over different systems, and lets you hear each instrument clearly and distinctly. Yet judging by a lot of project studio demos that pass across my desk, achieving the perfect mix is not easy…in fact, it’s very hard. So, here are some tips on how to get that wide open sound whenever you mix.

The Gear: Keep it Clean

Eliminate as many active stages as possible between source and recorder. Many times, devices set to “bypass” may not be adding any effect but are still in the signal path, which can add some slight degradation. How many times do line level signals go through preamps due to lazy engineering? If possible, send sounds directly into the recorder—bypass the mixer altogether. For mic signals, use an ultra-high quality outboard preamp and patch that directly into the recorder rather than use a mixer with its onboard preamps.

Although you may not hear much of a difference when monitoring a single instrument if you go directly into the recorder, with multiple tracks the cumulative effect of stripping the signal path to its essentials can make a significant difference in the sound’s clarity.

But what if you’re after a funky, dirty sound? Just remember that if you record with the highest possible fidelity, you can always mess with the signal later on during mixdown.

The Arrangement

Before you even think about turning any knobs, scrutinize the arrangement. Solo project arrangements are particularly prone to “clutter” because as you lay down the early tracks, there’s a tendency to overplay to fill up all that empty space. As the arrangement progresses, there’s not a lot of space left for overdubs.

Here are some suggestions when tracking:

  • Once the arrangement is fleshed out, go back and recut tracks that you cut earlier on. Try to play these tracks as sparsely as possible to leave room for the overdubs you’ve added. Like many others, I write in the studio, and often the song will have a slightly tentative feel because it wasn’t totally solid prior to recording it. Recutting a few judicious tracks always seems to both simplify and improve the music.
  • Try building a song around the vocalist or other lead instrument instead of completing the rhythm section and then laying down the vocals. I often find it better to record simple “placemarkers” for the drums, bass, and rhythm guitar (or piano, or whatever), then immediately get to work cutting the best possible vocal. When you re-record the rhythm section for real, you’ll be a lot more sensitive to the vocal nuances.
  • As Sun Ra once said, “Space is the place.” The less music you play, the more weight each note has, and the more spaciousness this creates in the overall sound.

Now let’s take a closer look…

Mastering

Mastering is the Supreme Court of audio—if you can’t get a ruling in your favor there, you have nowhere else to go. A pro mastering engineer can often turn muddy, tubby-sounding recordings into something much clearer and defined. Just don’t expect miracles, because no one can squeeze blood from a stone. But a good mastering job might be just the thing to take your mix to the next level, or at least turn a marginal mix into a solid one.

The main point of this article is that there is no button you can click on that says “press here for wide open mixes.” A good mix is the cumulative result of taking lots of little steps, such as the ones detailed above, until they add up to something that really works. Paying attention to detail does indeed help.

To read the full detailed article see:  How to Create Wide Open Mixes

 

February 21, 2011

Fender Rumble 150 Review

Filed under: Amps, Bass — Tags: , , , , , , , , , — audiofanzine @ 12:11 pm

I got a bit nostalgic when I wrote this review. I was 16 when my first amp was waiting for me under the Christmas tree. Not a 15-watt amp but a big one. To play in a band, do live gigs and be cheered by a wild crowd. In short, I could stop playing by myself in my room. The amp was almost more bulky than the Christmas tree: it was a second-hand Fender BXR 300, a huge combo with casters. I played my first live gig with it — this 15″ amp brings back lots of memories.

Sixteen years later, I’m reviewing the Rumble 150. This descendant of the BXR is conceived for bass players looking for a first amp to play in a band — like I did at that time. Let me wipe a tear… And now, let’s get on with the review!

Workhorse

Fender Rumble 150

64 lbs, impressing size (13.4″ x 22.8″ x 23.6″) and 150 effective watts. A big 15″ woofer, a tweeter for high frequencies (it was missing on my BXR) and enough volume to provide a big and deep sound. The manufacturer kept the front port for the bass-reflex (it’s the third generation), and removed the bright LEDs and the carpet covering. The latter is replaced with a black textured vinyl covering. Carpet or vinyl? It’s all a matter of taste. Personally, I don’t like to dust nor vacuum clean. On the other side, Tolex is easily marked. It’s an aesthetic or practical choice.

The front side is sleek and simple, which is a good thing: just a black protection grill and a black panel with white silkscreen. As for controls and connections, everything is on the front panel. Nothing on the rear panel except the power connector. The connections are quite comprehensive: instrument input (with active/passive switch), effect loop, RCA aux input (for connection to a PC, MP3 player…), phones output, footswitch connector (for overdrive control), and XLR line output.

Fender Rumble 150

The amp also provides numerous settings: gain control, overdrive section (with gain, balance and bypass), two shape switches (punch and scoop), four-band EQ, and an on/off switch for the tweeter.

As mentioned in the headline, the Rumble 150 is equipped with four rugged casters. It also has two recessed side handles with springs, which are a bit too thin for my taste. However, they do their job. I love casters! What would be of our backs it it weren’t for them? Considering the price, it’s not surprising that the amp and preamp stages use solid-state technology. A big fan on the rear panel ensures cooling, and it also makes a bit of noise. This noise is not deafening but it is clearly audible when you aren’t playing.

So, what’s new? To make it short, the Rumble 150 has more output power than its predecessor. Indeed, the whole product range got more watts, except for the Rumble 15. I guess nobody is going to complain for getting 50 watts more, plus overdrive. The manufacturer doesn’t offer the 2×10″ alternative, which is a good choice considering that users of this amp want to play loud and heavy rather than gently.

The product is made in China. The overall manufacturing and finish quality is good. Ok, now let’s plug a bass guitar! For this review, I used my American PB deluxe 5, a Boss RC-20 (my faithful sampler on stage) and a Zoom H2.

As well as a pair of cables — never forget the essentials!

Now let’s take a closer look…

Conclusion

My impression at the end of the review is that the combo provides a wide sound range, in spite of its rather sophisticated sound character. The sound samples show that the user can shape the sound easily to find the tone he wants. A very good point if this is your first amp. If you are a beginner, you probably don’t really know what kind of music you’ll end up playing.

It’s always important to stay open and have the possibility to become an all-round bass player and have fun with any music genre. Considering its price, the Rumble 150 is an interesting product for musicians who start playing in a band and want enough output power for that. The quality and value for money are good. Give it a try and compare it with competitor products.

Advantages:

  • Value for money
  • Actual output power
  • Easy setting
  • Versatility
  • Casters

Drawbacks:

  • Tweeter distorts at loud volumes
  • Neutral sound character, especially when the tweeter is off
  • I’m too old for such gifts under the Christmas tree… I want my youth back!

To read the full detailed article see:  Fender Rumble 150 Review

February 18, 2011

Tonehammer Pianos Review

The launch of the Montclarion Hall Piano gives us the opportunity to present to you the full range of Tonehammer pianos, characterized by the same original approach of all the other instruments by the manufacturer.

Granny, This One’s For You…

Tonehammer Old Granny Piano

Chronologically, this was the first piano presented by Tonehammer. The manufacturer decided to sample an old brandless upright piano (the booklet says the piano was 60 or 70 years old) which had had no maintenance in years and was in a rather poor condition. Granny is really detuned and has no strings for the high notes. In fact, you can clearly hear noises when you hit and release the keys.

 

Tonehammer Old Granny Piano

In spite of being out of tune, the instrument is appealing. The manufacturer provides several presets, available in Untuned and Tuned versions. The piano is available in soft and bright versions. It includes a tone setting controlled by the wheel. You also get several programs with different impulses like tunnel, studio, stairwell, alley, garage, and subway, as well as Kotankt’s convolution reverb. You can change the settings using the edit functions. Since the impulses are provided in a separate folder, you can also use them with other Kontakt instruments — a very nice detail. You’ll also find three (sound design) presets whose aim isn’t authenticity.

Let’s hear some sound samples then…

Conclusion

 

Tonehammer has a reputation for going off the beaten path with its sampling products by offering rare and home-made instruments or using special recording situations for ensembles (see the Epic series). However, it also succeeds in offering more classic choir and piano samples while keeping a special approach to them. Among the special products, the Bowed and Plucked libraries are very original and even though you can find similar libraries out there, none of them reaches this sound quality. Besides their excellent audio quality, both include numerous extra features like an arpeggiator, sound design programs, impulses, etc.

 

Among the more traditional sounds, Emotional is a unique product because no other virtual piano currently provides this particular quality and roundness. This is one of my favorite pianos. Montclarion offers special acoustics with very interesting multimodes if you want to create particular ambiances. As for the cons, we noticed some slight phase problems, especially with Emotional and Montclarion. To solve the problem just narrow the stereo image a little bit (the changes are so slight that they won’t alter the sound). Also notice that you’ll need a powerful computer system with enough RAM and/or fast hard drives.

 

You can consider this series as several single instruments, but also as a comprehensive bundle offering almost everything in terms of piano sound (perhaps missing only a prepared piano) with an impeccable quality (Montclarion at 24 bits, no audible loop points on releases except in FX or Drone programs, no tuning or layer problems, etc.), which can complete (or not) the more “traditional” instruments other manufacturers have to offer. In any case, the full bundle costs only $389 — a very appealing price, considering its quality.

Advantages:

  • Sound quality
  • Programming quality
  • Originality
  • No audible loop points except on special effects
  • Numerous traditional programs
  • Numerous effect and drone programs
  • Bowed Legato
  • Plucked Dulcimer
  • Uberpeggiator
  • High-quality original impulses
  • Originality of the FX impulses
  • Interface and settings of Bowed and Plucked
  • Price

Drawbacks:

  • Locked Kontakt format of some libraries
  • No access to Kontakt editors in Emotional
  • No external access to the Bowed and Plucked impulses
  • Watch out for phase problems with certain programs
  • Powerful computer and enough RAM required

To read the full detailed article with sound samples see:  Tonehammer Pianos Review

February 17, 2011

Extreme Drum Processing: Exploring the Art of Filthy Signal Mutation

Filed under: Drums/Percussion, Mixing reviews, Plugin, Software — Tags: , , , , , , , — audiofanzine @ 8:43 am

I like music with a distinctly electronic edge, but also want a human “feel.” Trying to resolve these seemingly contradictory ideals has led to some fun experimentation, but one of the more recent “happy accidents” was finding out what happens when you apply heavy signal processing to multitracked drums played by a human drummer. I ended up with a sound that slid into electronic tracks as easily as a debit card slides into an ATM machine, yet with a totally human feel.

This came about because Discrete Drums, who make rock-oriented sample libraries of multitracked drums (tracks are kick, snare, stereo toms, stereo room mic tracks, and stereo room ambience), received requests for a more extreme library for hip-hop/dance music. I had already started using their CDs for this purpose, and when I played some examples of loops I had done, they asked whether I’d like to do a remixed sample CD with stereo loops. Thus, the “Turbulent Filth Monsters” project was born, which eventually became a sample library (originally distributed by M-Audio, and now by Sonoma Wire Works).

Although I used the Discrete Drums sample library CDs and computer-based plug-ins, the following techniques also apply to hardware processors used in conjunction with drum machines that have individual outs, or multitracked drums recorded on a multitrack recorder (or sample CD tracks bounced over to a multitrack). Try some of these techniques, and you’ll create drum sounds that are as unique as a fingerprint – even if they came from a sample CD.

Effects Automation and Real Time Control

Editing parameters in real time lets you “play” an effect along with the beat. This is a good thing. However, it’s unlikely that you’ll be able to vary several parameters at once while mixing the track down to a loop, so you’ll want to record these changes as automation.

Hardware signal processors can often accept MIDI controllers for automation. If so, you can sync a sequencer up to whatever is playing the tracks. Then, deploy a MIDI control surface (like the Mackie Control, Novation Nocturn, etc.) to record control data into the sequencer. Once in the sequencer, edit the controller data if needed.

If the processor cannot accept control signals, then you’ll need to make these changes in real time. If you can do this as you mix, fine. Otherwise, bounce the processed signal to another track so it contains the changes you want.

Software plug-ins for DAWs are a whole other matter, as there are several possible automation scenarios:

  • Use a MIDI control surface to alter parameters, while recording the data to a MIDI track (hopefully this will drive the effect on playback)
  • Twiddle the plug-in’s virtual knobs in real time, and record those changes within the host program
  • Use non-real time automation envelopes
  • Record data that takes the form of envelopes, which you can then edit
  • Use no automation at all. In this case, you can send the output through a mixer and bounce it to another track while varying the parameter. This can require a little after-the-fact trimming to compensate for latency (i.e., delay caused by going through the mixer then returning back into the computer) issues.

For example, with VST Automation (Fig. 1), a plug-in will have Read and Write Automation buttons.

Ohm Force Predatohm & VST automation

Fig. 1: Click on the Write Automation button with a VST plug-in, and when you play or record, tweaking controls will write automation into your project.

If you click on the Write Automation button, any changes you make to automatable parameters will be written into your project. This happens regardless of whether the DAW is in record or playback mode.

Now let’s take a closer look at some other plug-ins…

So What’s the Payoff?

Drum loops played by a superb human drummer, with all those wonderful little timing nuances that are the reason drum machines have not taken over the world, will give your tracks a “feel” that you just can’t get with drum machines. But if you add on really creative processing, the sounds will be so electronified that they’ll fit in perfectly with more radical instruments synths, highly processed vocals, and technoid guitar effects.

So, get creative – you’ll have a good time doing it, and your recordings won’t sound like million others. What good are all these great new toys if you don’t exploit them?

To read the full detailed article see:  Extreme Drum Processing

February 11, 2011

Capturing Guitar Amps in the Wild: Multi-Channel Micing for Live Sound

There are almost as many ways to capture guitar amplifier sound with a microphone as there are for a piano. And as with piano (and kick and snare drum, for that matter) single-mic approaches can’t always provide the best solution for guitar amps – we must also explore multiple-mic approaches.

A Vox AC30 with a Shure KSM32

(above) and an Orange 4 x 12

with an Audio- Technica AT4050.

 


About four decades ago, at the “dawn” of modern live sound reinforcement, there was the Shure SM58 for vocals and the SM57 for instruments.  This eventually included mic’ing guitar amps, because as the PA got bigger than the backline, there was a danger that the guitars wouldn’t be heard over the vocals (causing the sound guy’s credibility to be doubted by the guitar player’s girlfriend behind his back).  In the golden days of rock, tuning the PA consisted of saying “check, one-two” into an SM58 and manipulating the faders on a Klark Teknik DN30 graphic EQ until the voice sounded as natural as possible.  Because the SM57 and SM58 have nearly identical response, this led to natural sounding instruments as well.

 

Over the years, sound systems have become increasingly full-range and high-fidelity, with modern systems exhibiting smoother, more even response.  At the same time, today’s large-diaphragm condensers have become more rugged and sturdy than their tube-based ancestors, and have made their way out of the studio and onto the stage.  “Big Mick” Hughes, Metallica’s engineer for a quarter century, is credited with putting Audio-Technica AT4050 studio condensers on stage and introducing their use in stereo pairs on guitar rigs.

One popular approach is to deploy a pair of matched studio-quality large diaphragm condensers, each on a separate cabinet of a stereo guitar rig, that also act as a pair of stereo “ears” for in-ear monitors (IEM). They also provide redundancy to the PA, and can be panned or doubled as needed.

Desired Response

Dual Shure SM57s – one for each

speaker cone – on this 65amps

Monterey 2×12 combo.

Most guitar amps don’t achieve their proper “sound” until the onset of clipping, producing that warm, yummy crunch, but yielding high-decibel sound pressure.  Strategies include using a “power soak” to draw some of the power off, going with lower-powered guitar amps, or remotely locating the amp or just its cabinet and isolating it from the performance stage.

 

Dynamic mics produce a contoured response, with warmth in the lows due to proximity effect, and often, a highmid presence.  Besides the Shure SM57, perennial dynamic mic choices for guitar cabinets include the Electro-Voice RE20, Sennheiser MD421 and MD409 (replaced by the 421 II and e609), AKG D 112, joined by a relatively new contender, the Audix i5.

Condenser mics offer extended highs and lows while providing a flatter frequency response.  The Neumann U87 is the gold standard for large diaphragm condenser mics, rarely seen outside of studios. It’s heritage also includes the TL103.  The AKG C 414, in all its variations, has been crossing over to the stage for many years, popular in particular for drum overhead and grand pianos.  Audio-Technica’s AT4050 is the largeformat condenser that first broke into live sound specifically for guitar cabinets, followed closely by the Shure KSM32.

Ribbon mics, with a bi-directional figure-of-eight pattern, have a transparent sound that allows the amplifier’s character to be clearly heard with a natural roll-off in the highs.  They re-entered recording studios several years ago when manufacturers began making them more rugged to withstand normal handling.  The Royer R-121 was the first modern ribbon to find widespread acceptance, and two years ago the company released a ruggedized “live” version with a thicker ribbon.  Recently, the new Shure KSM313 ribbon has earned its place on national tours, as has the new A-T AT4081 ribbon mic.

Now let’s take a closer look at other solutions…

The Direct Route

A Radial JDX DI can capture the

warmthof tube guitar amps while

addingresponse that emulates a

guitar speaker.

In the world of live hard rock or heavy metal, it’s common to find amplifier DIs which take their signal from after the guitar amp and in parallel with a speaker cabinet.  The original is the Hughes & Kettner Redbox, and Radial Engineering makes a modern JDX “amplifier DI” that’s active and employs Class A discrete electronics. These devices capture the warmth of tube guitar amps, while adding response that emulates a guitar speaker.

 

Redbox DIs eliminate inconsistencies from mic selection and placement, accidental misplacement of the mic and speed up changeovers on multi-band concerts by requiring only a re-patch of an XLR – no mic to move.  They employ electronics to emulate the response of a guitar cabinet’s speaker cone, rolling off the highs like a real speaker.  They’re specially equipped to take the higher voltage of a guitar amp’s output, but the big warning is they don’t act as a speaker load and must be used with a cabinet, or the amp will fry them.  When used in combination with a single microphone, the results can provide a wide range of creative options, and their relative distances are only determined by the one mic’s position.

 

This is a personal favorite for in-ear monitor mixes learned from Meredith Brooks, with the DI

The desired mic position can be clearly

marked on the cabinet’s grill using

gaffe tape. (That’s a Royer R-121L

ribbon mic, by the way.)

panned away from the rest of the band and the mic towards the band, but it’s a stereo effect and works best with a stereo IEM mix with both ears in.

The distance from the speaker cabinet is considered important in most studio recording applications, but in live sound, the inverse square law dictates that placing the mic right against the grill cloth reduces bleed from adjacent sound sources.  That said, when guitar amps are placed next to each other, use of gobos can increase their isolation from each other.

 

With modern in-ear monitoring, guitar players no longer need their cabinets on-stage with them, so it’s common for the guitar tech to set them up off stage (hopefully on the opposite side of the stage from the monitor console).  This gives the guitar tech full access to the amps during the show, and keeps them from muddying up the sound in the venue.

Today’s live sound systems provide opportunities to easily make multi-track recordings that allow engineers to compare various approaches to many sound reinforcement applications by swapping different combinations of inputs and auditioning them in the PA, without having to annoy the band to play the song over and over.  It also allows the engineer to demonstrate mic choices to a guitar player while he’s standing at the console and listening instead of playing.  Do this, and it leads to better communication and collaboration, and you may even become friends for life.

To read the full detailed article see:   Capturing Guitar Amps in the Wild

February 9, 2011

DC Offset: The Case of the Missing Headroom

Filed under: Mastering — Tags: , , — audiofanzine @ 3:11 pm

It was a dark and stormy night. I was rudely awakened at 3 AM by the ringing of a phone, pounding my brain like a jackhammer that spent way too much time chowing down at Starbucks. The voice on the other end was Pinky the engineer, and he sounded as panicked as a banana slug in a salt mine. “Anderton, some headroom’s missing. Vanished. I can’t master one track as hot as the others on the Kiss of Death CD. Checked out the usual suspects, but they’re all clean. You gotta help.”

Like an escort service at a Las Vegas trade show, my brain went into overdrive. Pinky knew his stuff…how to gain-stage, when not to compress, how to master. If headroom was stolen right out from under his nose, it had to be someone stealthy. Someone you didn’t notice unless you had your waveform Y-axis magnification up. Someone like…DC Offset.

 

Okay, so despite my best efforts to add a little interest, DC offset isn’t a particularly sexy topic. But it can be the culprit behind problems such as lowered headroom, mastering oddities, pops and clicks, effects that don’t process properly, and other gremlins.

DC Offset in the Analog Era

We’ll jump into the DC offset story during the 70s, when op amps became popular. These analog integrated circuits pack a tremendous amount of gain in a small, inexpensive package with (typically) two inputs and one output. Theoretically, in its quiescent state (no input signal), the ins and out are at exactly 0.00000 volts. But due to imperfections within the op amp itself, sometimes there can be several millivolts of DC present at one of the inputs.

Normally this wouldn’t matter, but if the op amp is providing a gain of 1000 (60dB), a typical 5 mV input offset signal would get amplified up to 5000mV (5 volts). If the offset appeared at the inverting (out of phase) input, then the output would have a DC offset of –5.0 volts. A 5mV offset at the non-inverting input would cause a +5.0 DC offset.

There are two main reasons why this is a problem.

  • Reduced dynamic range and headroom. An op amp’s power supply isbipolar (i.e., there are positive and negative supply voltages with respect to ground). Suppose the op amp’s maximum undistorted voltage swing is ±15V. If the output is already sitting at, say, +5V, the maximum voltage swing is now +10/-20V. However, as most audio signals are usually symmetrical around ground and you don’t want either side to clip, the maximum voltage swing is really down to ±10V—a 33% loss of available headroom.
  • Problems with DC-coupled circuits. In a DC-coupled circuit (sometimes preferred by audiophiles due to superior low frequency response), any DC gets passed along to the next stage. Suppose the op amp mentioned earlier with a +5V output offset now feeds a DC-coupled circuit with a gain of 5. That +5V offset becomes a +25V offset—definitely not acceptable!

Now let’s take a closer look at some other cases…

Digital Solutions

There are three main ways to solve DC offset problems with software-based digital audio editing programs.

  • Most pro-level digital audio editing software includes a DC offset correction function, generally found under a “processing” menu along with functions like change gain, reverse, flip phase, etc. This function analyzes the signal, and adds or subtracts the required amount of correction to make sure that 0 really is 0. Many sequencing programs also include DC offset correction as part of a set of editing options (Fig. 3).
  • Apply a steep high-pass filter that cuts off everything below 20Hz or so. (Even with a comparatively gentle 12dB/octave filter, a signal at 0.5Hz will still be down more than 60dB). In practice, it’s not a bad idea anyway to nuke the subsonic part of the spectrum, as some processing can interact with a signal to produce modulation in the below 20Hz zone. Your speakers can’t reproduce signals this low and they just use up bandwidth, so nuke ’em.
  • Select a 2—10 millisecond or so region at the beginning and end of the file or segment with the offset, and apply a fadein and fadeout. This will create an envelope that starts and ends at 0, respectively. It won’t get rid of the DC offset component within the file (so you still have the restricted headroom problem), but at least you won’t hear a pop at transitions.

Case Closed

Granted, DC offset usually isn’t a killer problem, like a hard disk crash. In fact, usually there’s not enough to worry about. But every now and then, DC offset will rear its ugly head in a way that you do notice. And now, you know what to do about it.

To read the full detailed article please see:  DC Offset

February 7, 2011

SWR HeadLite Amplifier Head & Amplite Amplifier Review

On today’s menu, to get rid of the cold and warm up, we have a light but nourishing pair of SWR class-D heads (an amp/preamp combo and a power amplifier).

This would be a dream come true on any restaurant’s menu. And since French gastronomy (Yes, I is Vrench!) is now part of the UNESCO world cultural heritage, I would like to talk about food. But what do 7 lbs of potatoes and a 800-watt amplifier system have in common? First of all, the weight! And also the fact that both fit inside the vegetable compartment of a small fridge. Class-D amplifiers are more common nowadays. Most manufacturers have developed their own models, and now SWR serves us a new interpretation.

Small is Sweet…

SWR HeadLite and AmpLite

And it even fits in my gig bag’s pocket. The main advantages of a switching amplifier are its extremely compact size and very light weight, despite a high output power. Just imagine riding to the recording studio with 400 watts on your bike. And if that’s not enough, imagine the same with 800 watts! Plus a tube preamp, semi-parametric EQ, compressor and enhancer. The whole universe of SWR has been miniaturized an fitted into a very convenient and compact housing.

Headlite: 1.8″ x 8.5″ x 9.8″ for 3.7 lb. / Amplite: 1.8″ x 8.5″ x 9.8″ for 3 lb. Incredible! But before testing these products, let me make a brief summary of the brand as well as of class-D technology.

Garage Brand

In the beginning of the 80’s, clean sound was trendy. It had to be less raw and more sophisticated than the past decade. All radio stations played New Wave synth music, Michael Jackson was the King of pop and soul music changed disco for funk. An engineer at Accoustic Control Corporation, the brand of choice of Larry Graham, Jaco Pastorius and John Paul Jones (to name just a few!), decided to radically change the bass amplification market based on the fact that many famous studio musicians wanted more sound clarity and neutrality.

SWR HeadLite and AmpLite

Steeve W. Rabe started his small revolution in a garage where he, together with some associates, tried out many preamp/EQ/amp combinations until he satisfied the pro bass players in Los Angeles. A handful of them tested the prototypes during different recording sessions.

This long and arduous work would lead to the brand’s first amplifier head in 1984. Called PB-200 (which became later the famous SM-400), this amp head already offered all the features that made the young company a success: a tube preamp, a stereo amp, a semi-parametric EQ, a DI out (a groundbreaking feature for a bass amplifier), an aural exciter, and a compressor.

Following the success of their amp heads in recording studios, SWR started to manufacture speaker cabinets to set a foot in the live amplification market. The first Golliath speaker cabinet was introduced in 1986 and combined four 10″ speakers (a new concept introduced by Trace Elliot) with a tweeter. The success was immediate in the professional amplification market.

In 1997, Steeve W Rabe sold his company to form Raven Labs. The new owners would sell the brand again to FMIC (Fender Musical Instrument Corporation) in 2003. Today, the products of the brand are manufactured mainly in Corona and Ensenada, California, together with other Fender products.

Now let’s take a closer look…

Marcus’ Favorite Tool

I had to return the products just before the trade show in Paris (France) so that Marcus Miller could use them for demos. I’m moved by the fact that I could use the same gear as Michel and Marcus (yes, since we all use the same amp, we call ourselves by our first names): it is almost as if I had intruded into the privacy of these two bass guitar gods…

It’s true, I’m boasting a bit but this conclusion is mine and I want it to be brilliant and positive. SWR offers an affordable class-D system considering its quality. It requires a bit of adaptation to manage all possibilities and to get on with the lack of visual scales around the controls, but I’ll bet you anything that they will add them to future versions. To be seriously considered — for fun or business.

Advantages:

  • Size
  • Weight
  • Sound shaping possibilities
  • Connections
  • Output power
  • Sold with bag (not included with the products reviewed)

Drawbacks:

  • No scale around the controls
  • Only one speaker out on the HeadLite

To read the full detailed review with sound samples see:  SWR Headlite Amp

February 4, 2011

Dynamics Processing Meets Rock Guitar: How to Compress a Guitar or Bass

Dynamics processing with studio-oriented processors? Been there, done that. But have you re-visited it lately in a guitar context? Dynamics control for vocals or program material is very different compared to guitar. Much of this is because there are many ways to use dynamics processing for guitar (or bass). So, let’s take a look at the different ways to apply dynamics, with examples of suggested settings.

For an introduction to compression, check out the article “Compressors Demystified.” If you’re already up to speed, let’s give a few basics on how to set up studio processors with guitar (however, note that these same basic techniques work with plug-in software compressors as well as hardware).

The Interface Space

“Stomp box” dynamics processors, while designed specifically for guitar, are more limited than rack-mount studio hardware – but the latter have issue levels with guitar. Interfacing involves one of four approaches:

Use the instrument input. If the processor has an “instrument” input, you’re golden. Plug the guitar directly into the processor, then run it into the mixer, amp modeler, guitar amp (assuming you can adjust the output level to avoid total overload), or whatever. Look for an instrument input impedance above 100kilohms, and preferably above 220kilohms, to avoid dulling high frequencies and reducing level. But too high an impedance (in the 5-10Megohm range) reaches a point of diminishing returns, because now the input may be too sensitive and prone to noise pickup. A 1Megohm impedance is a good compromise setting.

Use a preamp or suitable direct box. Adding a preamp or direct box (assuming it has an appropriately high input impedance) before the processor will preserve the guitar signal’s fidelity and allow for best level matching. If you’re driving a guitar amp, you may be able to use the dynamics processor’s output control to add some extra overdrive, but don’t go overboard (or do, if you like really nasty sounds!).

Insert into your guitar amp’s effects loop. If you want to record with your guitar amp but are using a line-level processor, patch it into the guitar amp’s effects loop. The loop should be able to provide line levels for the send (goes into the processor’s input) and return (comes from the processor’s output).

If you’re using a hardware mixer, insert the dynamics processor into your mixer’s channel inserts. This will also match levels properly, although you’ll still have to figure out how to interface the guitar with the mixer. The choices are the same as above: If the mixer has an instrument input, great. If not, use a preamp, direct box, etc. between the guitar and mixer.

Now let’s take a closer look how to really do it…

Double Your Pleasure

Patching two compressors in series, with both set for small amounts of compression, can give a significant amount of compression but sound less obvious than using a single compressor to give the same amount of compression. The first stage essentially “pre-conditions” the signal so that the second compressor doesn’t have to work so hard.

 

If you have a stereo compressor that can be set to dual mono operation, you can patch the two individual compression channels in series. With plug-ins, you can just insert two in series in a track. The drawback is that unlike standard compression, where you have to adjust only one set of controls, an ˆ la carte approach requires adjusting both sets of compressor controls. While this might seem like a disadvantage, most of the time you’ll set them to similar settings anyway.

Window Shopping

To get an idea of what’s out there in compressor-land, visit a few retailers and manufacturers and you’ll see the choices are huge, ranging from under a hundred dollars to thousands (and thousands!) of dollars. But realistically, for the type of application we’re describing here, you don’t need anything too fancy – it’s not like you’re using the compressor to re-master vintage recordings for audiophile releases. Besides, these days technology is at a level where even fairly inexpensive devices can deliver excellent results.

 

In any event, all the above tips are just guidelines. Experiment with your dynamics processor, and you may find yet another way to exploit these perhaps unglamorous, but extremely useful, devices.

To read the full detailed article see:  How to Compress a Guitar or Bass

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